Merge "vorbisdec: support 192Khz sample rate for vorbis" into qt-aml-media-dev
am: f54b3d7732
Change-Id: I67fc51548fac57586e9841678731516d79c4f69d
diff --git a/apex/ld.config.txt b/apex/ld.config.txt
index af8ec06..f56e1b5 100644
--- a/apex/ld.config.txt
+++ b/apex/ld.config.txt
@@ -44,7 +44,7 @@
namespace.platform.asan.search.paths += /apex/com.android.runtime/${LIB}
# /system/lib/libc.so, etc are symlinks to /apex/com.android.lib/lib/bionic/libc.so, etc.
-# Add /apex/... pat to the permitted paths because linker uses realpath(3)
+# Add /apex/... path to the permitted paths because linker uses realpath(3)
# to check the accessibility of the lib. We could add this to search.paths
# instead but that makes the resolution of bionic libs be dependent on
# the order of /system/lib and /apex/... in search.paths. If /apex/...
@@ -131,3 +131,9 @@
# Add a link for libz.so which is llndk on devices where VNDK is not enforced.
namespace.sphal.link.platform.shared_libs += libz.so
+
+# With VNDK APEX, /system/${LIB}/vndk-sp${VNDK_VER} is a symlink to the following.
+# Add /apex/... path to the permitted paths because linker uses realpath(3)
+# to check the accessibility of the lib.
+namespace.sphal.permitted.paths += /apex/com.android.vndk.${VNDK_APEX_VER}/${LIB}
+namespace.sphal.asan.permitted.paths += /apex/com.android.vndk.${VNDK_APEX_VER}/${LIB}
diff --git a/apex/manifest.json b/apex/manifest.json
index b11187d..3011ee8 100644
--- a/apex/manifest.json
+++ b/apex/manifest.json
@@ -1,4 +1,4 @@
{
"name": "com.android.media",
- "version": 299900000
+ "version": 290000000
}
diff --git a/apex/manifest_codec.json b/apex/manifest_codec.json
index 09c436d..83a5178 100644
--- a/apex/manifest_codec.json
+++ b/apex/manifest_codec.json
@@ -1,4 +1,4 @@
{
"name": "com.android.media.swcodec",
- "version": 299900000
+ "version": 290000000
}
diff --git a/camera/Android.bp b/camera/Android.bp
index 2800595..b288bcf 100644
--- a/camera/Android.bp
+++ b/camera/Android.bp
@@ -86,6 +86,7 @@
"aidl/android/hardware/camera2/ICameraDeviceCallbacks.aidl",
"aidl/android/hardware/camera2/ICameraDeviceUser.aidl",
],
+ path: "aidl",
}
// Extra AIDL files that are used by framework.jar but not libcamera_client
@@ -96,4 +97,5 @@
"aidl/android/hardware/ICamera.aidl",
"aidl/android/hardware/ICameraClient.aidl",
],
+ path: "aidl",
}
diff --git a/camera/cameraserver/Android.bp b/camera/cameraserver/Android.bp
index ecaba3a..320c499 100644
--- a/camera/cameraserver/Android.bp
+++ b/camera/cameraserver/Android.bp
@@ -17,6 +17,10 @@
srcs: ["main_cameraserver.cpp"],
+ header_libs: [
+ "libmedia_headers",
+ ],
+
shared_libs: [
"libcameraservice",
"liblog",
@@ -25,7 +29,6 @@
"libgui",
"libbinder",
"libhidlbase",
- "libhidltransport",
"android.hardware.camera.common@1.0",
"android.hardware.camera.provider@2.4",
"android.hardware.camera.provider@2.5",
diff --git a/camera/ndk/Android.bp b/camera/ndk/Android.bp
index a2ee65d..d8220eb 100644
--- a/camera/ndk/Android.bp
+++ b/camera/ndk/Android.bp
@@ -107,7 +107,6 @@
],
shared_libs: [
- "libhwbinder",
"libfmq",
"libhidlbase",
"libhardware",
@@ -143,7 +142,6 @@
vendor: true,
srcs: ["ndk_vendor/tests/AImageReaderVendorTest.cpp"],
shared_libs: [
- "libhwbinder",
"libcamera2ndk_vendor",
"libcamera_metadata",
"libmediandk",
diff --git a/camera/ndk/impl/ACameraDevice.cpp b/camera/ndk/impl/ACameraDevice.cpp
index d24cb81..46a8dae 100644
--- a/camera/ndk/impl/ACameraDevice.cpp
+++ b/camera/ndk/impl/ACameraDevice.cpp
@@ -29,7 +29,7 @@
#include "ACameraCaptureSession.inc"
ACameraDevice::~ACameraDevice() {
- mDevice->stopLooper();
+ mDevice->stopLooperAndDisconnect();
}
namespace android {
@@ -112,19 +112,7 @@
}
}
-// Device close implementaiton
-CameraDevice::~CameraDevice() {
- sp<ACameraCaptureSession> session = mCurrentSession.promote();
- {
- Mutex::Autolock _l(mDeviceLock);
- if (!isClosed()) {
- disconnectLocked(session);
- }
- LOG_ALWAYS_FATAL_IF(mCbLooper != nullptr,
- "CameraDevice looper should've been stopped before ~CameraDevice");
- mCurrentSession = nullptr;
- }
-}
+CameraDevice::~CameraDevice() { }
void
CameraDevice::postSessionMsgAndCleanup(sp<AMessage>& msg) {
@@ -892,8 +880,14 @@
return;
}
-void CameraDevice::stopLooper() {
+void CameraDevice::stopLooperAndDisconnect() {
Mutex::Autolock _l(mDeviceLock);
+ sp<ACameraCaptureSession> session = mCurrentSession.promote();
+ if (!isClosed()) {
+ disconnectLocked(session);
+ }
+ mCurrentSession = nullptr;
+
if (mCbLooper != nullptr) {
mCbLooper->unregisterHandler(mHandler->id());
mCbLooper->stop();
diff --git a/camera/ndk/impl/ACameraDevice.h b/camera/ndk/impl/ACameraDevice.h
index 7a35bf0..6c2ceb3 100644
--- a/camera/ndk/impl/ACameraDevice.h
+++ b/camera/ndk/impl/ACameraDevice.h
@@ -40,6 +40,7 @@
#include <camera/NdkCameraManager.h>
#include <camera/NdkCameraCaptureSession.h>
+
#include "ACameraMetadata.h"
namespace android {
@@ -110,7 +111,7 @@
inline ACameraDevice* getWrapper() const { return mWrapper; };
// Stop the looper thread and unregister the handler
- void stopLooper();
+ void stopLooperAndDisconnect();
private:
friend ACameraCaptureSession;
diff --git a/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp b/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp
index 35c8355..e511a3f 100644
--- a/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp
+++ b/camera/ndk/ndk_vendor/impl/ACameraDevice.cpp
@@ -45,7 +45,7 @@
using namespace android;
ACameraDevice::~ACameraDevice() {
- mDevice->stopLooper();
+ mDevice->stopLooperAndDisconnect();
}
namespace android {
@@ -125,19 +125,7 @@
}
}
-// Device close implementaiton
-CameraDevice::~CameraDevice() {
- sp<ACameraCaptureSession> session = mCurrentSession.promote();
- {
- Mutex::Autolock _l(mDeviceLock);
- if (!isClosed()) {
- disconnectLocked(session);
- }
- mCurrentSession = nullptr;
- LOG_ALWAYS_FATAL_IF(mCbLooper != nullptr,
- "CameraDevice looper should've been stopped before ~CameraDevice");
- }
-}
+CameraDevice::~CameraDevice() { }
void
CameraDevice::postSessionMsgAndCleanup(sp<AMessage>& msg) {
@@ -1388,6 +1376,7 @@
// before cbh goes out of scope and causing we call the session
// destructor while holding device lock
cbh.mSession.clear();
+
postSessionMsgAndCleanup(msg);
}
@@ -1400,8 +1389,13 @@
}
}
-void CameraDevice::stopLooper() {
+void CameraDevice::stopLooperAndDisconnect() {
Mutex::Autolock _l(mDeviceLock);
+ sp<ACameraCaptureSession> session = mCurrentSession.promote();
+ if (!isClosed()) {
+ disconnectLocked(session);
+ }
+ mCurrentSession = nullptr;
if (mCbLooper != nullptr) {
mCbLooper->unregisterHandler(mHandler->id());
mCbLooper->stop();
diff --git a/camera/ndk/ndk_vendor/impl/ACameraDevice.h b/camera/ndk/ndk_vendor/impl/ACameraDevice.h
index 9e034c4..7fc699e 100644
--- a/camera/ndk/ndk_vendor/impl/ACameraDevice.h
+++ b/camera/ndk/ndk_vendor/impl/ACameraDevice.h
@@ -36,6 +36,7 @@
#include <camera/NdkCameraManager.h>
#include <camera/NdkCameraCaptureSession.h>
+
#include "ACameraMetadata.h"
#include "utils.h"
@@ -134,7 +135,7 @@
inline ACameraDevice* getWrapper() const { return mWrapper; };
// Stop the looper thread and unregister the handler
- void stopLooper();
+ void stopLooperAndDisconnect();
private:
friend ACameraCaptureSession;
diff --git a/cmds/screenrecord/Android.bp b/cmds/screenrecord/Android.bp
index 86476cd..6bdbab1 100644
--- a/cmds/screenrecord/Android.bp
+++ b/cmds/screenrecord/Android.bp
@@ -24,6 +24,10 @@
"Program.cpp",
],
+ header_libs: [
+ "libmediadrm_headers",
+ ],
+
shared_libs: [
"libstagefright",
"libmedia",
diff --git a/cmds/screenrecord/screenrecord.cpp b/cmds/screenrecord/screenrecord.cpp
index 7aa655f..f2a71b3 100644
--- a/cmds/screenrecord/screenrecord.cpp
+++ b/cmds/screenrecord/screenrecord.cpp
@@ -52,7 +52,7 @@
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaMuxer.h>
#include <media/stagefright/PersistentSurface.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/MediaCodecBuffer.h>
#include "screenrecord.h"
@@ -368,6 +368,7 @@
int64_t startWhenNsec = systemTime(CLOCK_MONOTONIC);
int64_t endWhenNsec = startWhenNsec + seconds_to_nanoseconds(gTimeLimitSec);
DisplayInfo mainDpyInfo;
+ bool firstFrame = true;
assert((rawFp == NULL && muxer != NULL) || (rawFp != NULL && muxer == NULL));
@@ -384,6 +385,11 @@
int64_t ptsUsec;
uint32_t flags;
+ if (firstFrame) {
+ ATRACE_NAME("first_frame");
+ firstFrame = false;
+ }
+
if (systemTime(CLOCK_MONOTONIC) > endWhenNsec) {
if (gVerbose) {
printf("Time limit reached\n");
diff --git a/cmds/stagefright/Android.mk b/cmds/stagefright/Android.mk
index 6eb2e9f..defc94f 100644
--- a/cmds/stagefright/Android.mk
+++ b/cmds/stagefright/Android.mk
@@ -3,21 +3,21 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
+ AudioPlayer.cpp \
stagefright.cpp \
jpeg.cpp \
SineSource.cpp
LOCAL_SHARED_LIBRARIES := \
- libstagefright libmedia libmedia_omx libutils libbinder \
+ libstagefright libmedia libmedia_codeclist libutils libbinder \
libstagefright_foundation libjpeg libui libgui libcutils liblog \
- libhidlbase \
+ libhidlbase libdatasource libaudioclient \
android.hardware.media.omx@1.0 \
LOCAL_C_INCLUDES:= \
frameworks/av/media/libstagefright \
frameworks/av/media/libstagefright/include \
frameworks/native/include/media/openmax \
- external/jpeg \
LOCAL_CFLAGS += -Wno-multichar -Werror -Wall
@@ -32,14 +32,16 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
+ AudioPlayer.cpp \
SineSource.cpp \
record.cpp
LOCAL_SHARED_LIBRARIES := \
libstagefright libmedia liblog libutils libbinder \
- libstagefright_foundation
+ libstagefright_foundation libdatasource libaudioclient
LOCAL_C_INCLUDES:= \
+ frameworks/av/camera/include \
frameworks/av/media/libstagefright \
frameworks/native/include/media/openmax \
frameworks/native/include/media/hardware
@@ -57,12 +59,12 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- SineSource.cpp \
+ AudioPlayer.cpp \
recordvideo.cpp
LOCAL_SHARED_LIBRARIES := \
libstagefright libmedia liblog libutils libbinder \
- libstagefright_foundation
+ libstagefright_foundation libaudioclient
LOCAL_C_INCLUDES:= \
frameworks/av/media/libstagefright \
@@ -83,12 +85,13 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
+ AudioPlayer.cpp \
SineSource.cpp \
audioloop.cpp
LOCAL_SHARED_LIBRARIES := \
libstagefright libmedia liblog libutils libbinder \
- libstagefright_foundation
+ libstagefright_foundation libaudioclient
LOCAL_C_INCLUDES:= \
frameworks/av/media/libstagefright \
@@ -111,7 +114,7 @@
LOCAL_SHARED_LIBRARIES := \
libstagefright liblog libutils libbinder libui libgui \
- libstagefright_foundation libmedia libcutils
+ libstagefright_foundation libmedia libcutils libdatasource
LOCAL_C_INCLUDES:= \
frameworks/av/media/libstagefright \
@@ -133,6 +136,9 @@
codec.cpp \
SimplePlayer.cpp \
+LOCAL_HEADER_LIBRARIES := \
+ libmediadrm_headers \
+
LOCAL_SHARED_LIBRARIES := \
libstagefright liblog libutils libbinder libstagefright_foundation \
libmedia libmedia_omx libaudioclient libui libgui libcutils
@@ -154,22 +160,23 @@
include $(CLEAR_VARS)
LOCAL_SRC_FILES:= \
- filters/argbtorgba.rs \
- filters/nightvision.rs \
- filters/saturation.rs \
+ filters/argbtorgba.rscript \
+ filters/nightvision.rscript \
+ filters/saturation.rscript \
mediafilter.cpp \
+LOCAL_HEADER_LIBRARIES := \
+ libmediadrm_headers \
+
LOCAL_SHARED_LIBRARIES := \
libstagefright \
liblog \
libutils \
libbinder \
libstagefright_foundation \
- libmedia \
libmedia_omx \
libui \
libgui \
- libcutils \
libRScpp \
LOCAL_C_INCLUDES:= \
diff --git a/media/libstagefright/AudioPlayer.cpp b/cmds/stagefright/AudioPlayer.cpp
similarity index 99%
rename from media/libstagefright/AudioPlayer.cpp
rename to cmds/stagefright/AudioPlayer.cpp
index 199b57b..208713d 100644
--- a/media/libstagefright/AudioPlayer.cpp
+++ b/cmds/stagefright/AudioPlayer.cpp
@@ -28,12 +28,13 @@
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALookup.h>
#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MetaData.h>
#include <media/stagefright/Utils.h>
+#include "AudioPlayer.h"
+
namespace android {
AudioPlayer::AudioPlayer(
diff --git a/media/libstagefright/include/media/stagefright/AudioPlayer.h b/cmds/stagefright/AudioPlayer.h
similarity index 100%
rename from media/libstagefright/include/media/stagefright/AudioPlayer.h
rename to cmds/stagefright/AudioPlayer.h
diff --git a/cmds/stagefright/SimplePlayer.cpp b/cmds/stagefright/SimplePlayer.cpp
index afb7db3..f4b8164 100644
--- a/cmds/stagefright/SimplePlayer.cpp
+++ b/cmds/stagefright/SimplePlayer.cpp
@@ -23,7 +23,7 @@
#include <gui/Surface.h>
#include <media/AudioTrack.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/IMediaHTTPService.h>
#include <media/MediaCodecBuffer.h>
#include <media/stagefright/foundation/ABuffer.h>
diff --git a/cmds/stagefright/audioloop.cpp b/cmds/stagefright/audioloop.cpp
index d4f2e8d..bd274d8 100644
--- a/cmds/stagefright/audioloop.cpp
+++ b/cmds/stagefright/audioloop.cpp
@@ -29,11 +29,11 @@
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/AMRWriter.h>
-#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/AudioSource.h>
#include <media/stagefright/MediaCodecSource.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/SimpleDecodingSource.h>
+#include "AudioPlayer.h"
#include "SineSource.h"
using namespace android;
diff --git a/cmds/stagefright/codec.cpp b/cmds/stagefright/codec.cpp
index e5a4337..f2d1c29 100644
--- a/cmds/stagefright/codec.cpp
+++ b/cmds/stagefright/codec.cpp
@@ -23,7 +23,7 @@
#include <binder/IServiceManager.h>
#include <binder/ProcessState.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayerService.h>
#include <media/MediaCodecBuffer.h>
diff --git a/cmds/stagefright/filters/argbtorgba.rs b/cmds/stagefright/filters/argbtorgba.rscript
similarity index 100%
rename from cmds/stagefright/filters/argbtorgba.rs
rename to cmds/stagefright/filters/argbtorgba.rscript
diff --git a/cmds/stagefright/filters/nightvision.rs b/cmds/stagefright/filters/nightvision.rscript
similarity index 100%
rename from cmds/stagefright/filters/nightvision.rs
rename to cmds/stagefright/filters/nightvision.rscript
diff --git a/cmds/stagefright/filters/saturation.rs b/cmds/stagefright/filters/saturation.rscript
similarity index 100%
rename from cmds/stagefright/filters/saturation.rs
rename to cmds/stagefright/filters/saturation.rscript
diff --git a/cmds/stagefright/mediafilter.cpp b/cmds/stagefright/mediafilter.cpp
index 2cf6955..66302b0 100644
--- a/cmds/stagefright/mediafilter.cpp
+++ b/cmds/stagefright/mediafilter.cpp
@@ -24,9 +24,9 @@
#include <gui/ISurfaceComposer.h>
#include <gui/SurfaceComposerClient.h>
#include <gui/Surface.h>
-#include <media/ICrypto.h>
#include <media/IMediaHTTPService.h>
#include <media/MediaCodecBuffer.h>
+#include <mediadrm/ICrypto.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
diff --git a/cmds/stagefright/record.cpp b/cmds/stagefright/record.cpp
index 95a16f3..37091c4 100644
--- a/cmds/stagefright/record.cpp
+++ b/cmds/stagefright/record.cpp
@@ -17,12 +17,11 @@
#include "SineSource.h"
#include <binder/ProcessState.h>
+#include <datasource/FileSource.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/CameraSource.h>
-#include <media/stagefright/FileSource.h>
#include <media/stagefright/MediaBufferGroup.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaCodecSource.h>
@@ -33,6 +32,8 @@
#include <media/stagefright/SimpleDecodingSource.h>
#include <media/MediaPlayerInterface.h>
+#include "AudioPlayer.h"
+
using namespace android;
static const int32_t kAudioBitRate = 12200;
diff --git a/cmds/stagefright/recordvideo.cpp b/cmds/stagefright/recordvideo.cpp
index a63b9b9..01a178e 100644
--- a/cmds/stagefright/recordvideo.cpp
+++ b/cmds/stagefright/recordvideo.cpp
@@ -14,8 +14,6 @@
* limitations under the License.
*/
-#include "SineSource.h"
-
#include <inttypes.h>
#include <sys/types.h>
#include <sys/stat.h>
@@ -25,8 +23,8 @@
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/AudioPlayer.h>
#include <media/stagefright/MediaBufferGroup.h>
+#include <media/stagefright/MediaBuffer.h>
#include <media/stagefright/MediaCodecSource.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MetaData.h>
diff --git a/cmds/stagefright/stagefright.cpp b/cmds/stagefright/stagefright.cpp
index bf36be0..02ade94 100644
--- a/cmds/stagefright/stagefright.cpp
+++ b/cmds/stagefright/stagefright.cpp
@@ -31,18 +31,15 @@
#include <binder/IServiceManager.h>
#include <binder/ProcessState.h>
+#include <datasource/DataSourceFactory.h>
#include <media/DataSource.h>
#include <media/MediaSource.h>
-#include <media/ICrypto.h>
#include <media/IMediaHTTPService.h>
#include <media/IMediaPlayerService.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/foundation/AUtils.h>
-#include "include/NuCachedSource2.h"
-#include <media/stagefright/AudioPlayer.h>
-#include <media/stagefright/DataSourceFactory.h>
#include <media/stagefright/JPEGSource.h>
#include <media/stagefright/InterfaceUtils.h>
#include <media/stagefright/MediaCodec.h>
@@ -69,6 +66,8 @@
#include <android/hardware/media/omx/1.0/IOmx.h>
+#include "AudioPlayer.h"
+
using namespace android;
static long gNumRepetitions;
@@ -305,7 +304,7 @@
seekTimeUs = -1;
if (shouldSeek) {
- seekTimeUs = (rand() * (float)durationUs) / RAND_MAX;
+ seekTimeUs = (rand() * (float)durationUs) / (float)RAND_MAX;
options.setSeekTo(seekTimeUs);
printf("seeking to %" PRId64 " us (%.2f secs)\n",
@@ -1086,7 +1085,7 @@
const char *filename = argv[k];
sp<DataSource> dataSource =
- DataSourceFactory::CreateFromURI(NULL /* httpService */, filename);
+ DataSourceFactory::getInstance()->CreateFromURI(NULL /* httpService */, filename);
if (strncasecmp(filename, "sine:", 5) && dataSource == NULL) {
fprintf(stderr, "Unable to create data source.\n");
diff --git a/cmds/stagefright/stream.cpp b/cmds/stagefright/stream.cpp
index 35bdbc0..22e2ef3 100644
--- a/cmds/stagefright/stream.cpp
+++ b/cmds/stagefright/stream.cpp
@@ -21,6 +21,7 @@
#include <binder/ProcessState.h>
#include <cutils/properties.h> // for property_get
+#include <datasource/DataSourceFactory.h>
#include <media/DataSource.h>
#include <media/IMediaHTTPService.h>
#include <media/IStreamSource.h>
@@ -28,7 +29,6 @@
#include <media/MediaSource.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/AMessage.h>
-#include <media/stagefright/DataSourceFactory.h>
#include <media/stagefright/InterfaceUtils.h>
#include <media/stagefright/MPEG2TSWriter.h>
#include <media/stagefright/MediaExtractor.h>
@@ -164,7 +164,7 @@
: mCurrentBufferIndex(-1),
mCurrentBufferOffset(0) {
sp<DataSource> dataSource =
- DataSourceFactory::CreateFromURI(NULL /* httpService */, filename);
+ DataSourceFactory::getInstance()->CreateFromURI(NULL /* httpService */, filename);
CHECK(dataSource != NULL);
diff --git a/drm/libmediadrm/Android.bp b/drm/libmediadrm/Android.bp
index d6db1d4..84f2f6d 100644
--- a/drm/libmediadrm/Android.bp
+++ b/drm/libmediadrm/Android.bp
@@ -2,9 +2,16 @@
// libmediadrm
//
-// TODO: change it back to cc_library_shared when MediaPlayer2 switches to
-// using NdkMediaDrm, instead of MediaDrm.java.
-cc_library {
+cc_library_headers {
+ name: "libmediadrm_headers",
+
+ export_include_dirs: [
+ "interface"
+ ],
+
+}
+
+cc_library_shared {
name: "libmediadrm",
srcs: [
@@ -19,6 +26,19 @@
"CryptoHal.cpp",
],
+ local_include_dirs: [
+ "include",
+ "interface"
+ ],
+
+ export_include_dirs: [
+ "include"
+ ],
+
+ header_libs: [
+ "libmedia_headers",
+ ],
+
shared_libs: [
"libbinder",
"libcutils",
@@ -34,7 +54,6 @@
"android.hardware.drm@1.2",
"libhidlallocatorutils",
"libhidlbase",
- "libhidltransport",
],
cflags: [
@@ -52,10 +71,17 @@
"protos/metrics.proto",
],
+ local_include_dirs: [
+ "include"
+ ],
+
proto: {
export_proto_headers: true,
type: "lite",
},
+ header_libs: [
+ "libmedia_headers",
+ ],
shared_libs: [
"android.hardware.drm@1.0",
"android.hardware.drm@1.1",
@@ -83,10 +109,17 @@
"protos/metrics.proto",
],
+ local_include_dirs: [
+ "include"
+ ],
+
proto: {
export_proto_headers: true,
type: "full",
},
+ header_libs: [
+ "libmedia_headers",
+ ],
shared_libs: [
"android.hardware.drm@1.0",
"android.hardware.drm@1.1",
diff --git a/drm/libmediadrm/DrmHal.cpp b/drm/libmediadrm/DrmHal.cpp
index 919f4ee..e79fd4b 100644
--- a/drm/libmediadrm/DrmHal.cpp
+++ b/drm/libmediadrm/DrmHal.cpp
@@ -895,9 +895,8 @@
status_t DrmHal::provideKeyResponse(Vector<uint8_t> const &sessionId,
Vector<uint8_t> const &response, Vector<uint8_t> &keySetId) {
Mutex::Autolock autoLock(mLock);
- EventTimer<status_t> keyResponseTimer(&mMetrics.mProvideKeyResponseTimeUs);
-
INIT_CHECK();
+ EventTimer<status_t> keyResponseTimer(&mMetrics.mProvideKeyResponseTimeUs);
DrmSessionManager::Instance()->useSession(sessionId);
diff --git a/media/libmedia/include/media/CryptoHal.h b/drm/libmediadrm/include/mediadrm/CryptoHal.h
similarity index 100%
rename from media/libmedia/include/media/CryptoHal.h
rename to drm/libmediadrm/include/mediadrm/CryptoHal.h
diff --git a/media/libmedia/include/media/DrmHal.h b/drm/libmediadrm/include/mediadrm/DrmHal.h
similarity index 100%
rename from media/libmedia/include/media/DrmHal.h
rename to drm/libmediadrm/include/mediadrm/DrmHal.h
diff --git a/media/libmedia/include/media/DrmMetrics.h b/drm/libmediadrm/include/mediadrm/DrmMetrics.h
similarity index 100%
rename from media/libmedia/include/media/DrmMetrics.h
rename to drm/libmediadrm/include/mediadrm/DrmMetrics.h
diff --git a/media/libmedia/include/media/DrmPluginPath.h b/drm/libmediadrm/include/mediadrm/DrmPluginPath.h
similarity index 100%
rename from media/libmedia/include/media/DrmPluginPath.h
rename to drm/libmediadrm/include/mediadrm/DrmPluginPath.h
diff --git a/media/libmedia/include/media/DrmSessionClientInterface.h b/drm/libmediadrm/include/mediadrm/DrmSessionClientInterface.h
similarity index 100%
rename from media/libmedia/include/media/DrmSessionClientInterface.h
rename to drm/libmediadrm/include/mediadrm/DrmSessionClientInterface.h
diff --git a/media/libmedia/include/media/DrmSessionManager.h b/drm/libmediadrm/include/mediadrm/DrmSessionManager.h
similarity index 100%
rename from media/libmedia/include/media/DrmSessionManager.h
rename to drm/libmediadrm/include/mediadrm/DrmSessionManager.h
diff --git a/media/libmedia/include/media/IDrm.h b/drm/libmediadrm/include/mediadrm/IDrm.h
similarity index 100%
rename from media/libmedia/include/media/IDrm.h
rename to drm/libmediadrm/include/mediadrm/IDrm.h
diff --git a/media/libmedia/include/media/IDrmClient.h b/drm/libmediadrm/include/mediadrm/IDrmClient.h
similarity index 100%
rename from media/libmedia/include/media/IDrmClient.h
rename to drm/libmediadrm/include/mediadrm/IDrmClient.h
diff --git a/media/libmedia/include/media/IMediaDrmService.h b/drm/libmediadrm/include/mediadrm/IMediaDrmService.h
similarity index 100%
rename from media/libmedia/include/media/IMediaDrmService.h
rename to drm/libmediadrm/include/mediadrm/IMediaDrmService.h
diff --git a/media/libmedia/include/media/SharedLibrary.h b/drm/libmediadrm/include/mediadrm/SharedLibrary.h
similarity index 100%
rename from media/libmedia/include/media/SharedLibrary.h
rename to drm/libmediadrm/include/mediadrm/SharedLibrary.h
diff --git a/media/libmedia/include/media/ICrypto.h b/drm/libmediadrm/interface/mediadrm/ICrypto.h
similarity index 100%
rename from media/libmedia/include/media/ICrypto.h
rename to drm/libmediadrm/interface/mediadrm/ICrypto.h
diff --git a/drm/libmediadrm/tests/Android.bp b/drm/libmediadrm/tests/Android.bp
index 9e0115e..2e39943 100644
--- a/drm/libmediadrm/tests/Android.bp
+++ b/drm/libmediadrm/tests/Android.bp
@@ -3,8 +3,8 @@
cc_test {
name: "CounterMetric_test",
srcs: ["CounterMetric_test.cpp"],
+ header_libs: ["libmedia_headers"],
shared_libs: ["libmediadrm"],
- include_dirs: ["frameworks/av/include/media"],
cflags: [
"-Werror",
"-Wall",
@@ -14,6 +14,9 @@
cc_test {
name: "DrmMetrics_test",
srcs: ["DrmMetrics_test.cpp"],
+ header_libs: [
+ "libmedia_headers"
+ ],
shared_libs: [
"android.hardware.drm@1.0",
"android.hardware.drm@1.1",
@@ -28,7 +31,7 @@
],
static_libs: ["libgmock"],
include_dirs: [
- "frameworks/av/include/media",
+ "frameworks/av/drm/libmediadrm/include",
],
cflags: [
// Suppress unused parameter and no error options. These cause problems
@@ -40,12 +43,14 @@
cc_test {
name: "EventMetric_test",
srcs: ["EventMetric_test.cpp"],
+ header_libs: [
+ "libmedia_headers"
+ ],
shared_libs: [
"liblog",
"libmediadrm",
"libutils",
],
- include_dirs: ["frameworks/av/include/media"],
cflags: [
"-Werror",
"-Wall",
diff --git a/drm/libmediadrm/tests/CounterMetric_test.cpp b/drm/libmediadrm/tests/CounterMetric_test.cpp
index 6bca0da..c2becb4 100644
--- a/drm/libmediadrm/tests/CounterMetric_test.cpp
+++ b/drm/libmediadrm/tests/CounterMetric_test.cpp
@@ -16,7 +16,7 @@
#include <gtest/gtest.h>
-#include "CounterMetric.h"
+#include <media/CounterMetric.h>
namespace android {
diff --git a/drm/libmediadrm/tests/EventMetric_test.cpp b/drm/libmediadrm/tests/EventMetric_test.cpp
index eb6c4f6..b3c3f62 100644
--- a/drm/libmediadrm/tests/EventMetric_test.cpp
+++ b/drm/libmediadrm/tests/EventMetric_test.cpp
@@ -16,7 +16,7 @@
#include <gtest/gtest.h>
-#include "EventMetric.h"
+#include <media/EventMetric.h>
namespace android {
diff --git a/drm/mediacas/plugins/clearkey/Android.bp b/drm/mediacas/plugins/clearkey/Android.bp
new file mode 100644
index 0000000..0113cb8
--- /dev/null
+++ b/drm/mediacas/plugins/clearkey/Android.bp
@@ -0,0 +1,55 @@
+//
+// Copyright (C) 2017 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+// http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+
+cc_library_shared {
+ name: "libclearkeycasplugin",
+
+ srcs: [
+ "ClearKeyCasPlugin.cpp",
+ "ClearKeyFetcher.cpp",
+ "ClearKeyLicenseFetcher.cpp",
+ "ClearKeySessionLibrary.cpp",
+ "ecm.cpp",
+ "ecm_generator.cpp",
+ "JsonAssetLoader.cpp",
+ "protos/license_protos.proto",
+ ],
+
+ proprietary: true,
+ relative_install_path: "mediacas",
+
+ shared_libs: [
+ "libutils",
+ "liblog",
+ "libcrypto",
+ "libstagefright_foundation",
+ "libprotobuf-cpp-lite",
+ ],
+
+ header_libs: ["media_plugin_headers"],
+
+ static_libs: ["libjsmn"],
+
+ proto: {
+ type: "full",
+ export_proto_headers: true,
+ },
+
+ include_dirs: [
+ "frameworks/av/include",
+ "frameworks/native/include/media",
+ ],
+}
diff --git a/drm/mediacas/plugins/clearkey/Android.mk b/drm/mediacas/plugins/clearkey/Android.mk
deleted file mode 100644
index 4b139a8..0000000
--- a/drm/mediacas/plugins/clearkey/Android.mk
+++ /dev/null
@@ -1,71 +0,0 @@
-#
-# Copyright (C) 2017 The Android Open Source Project
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-# http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-#
-LOCAL_PATH:= $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= \
- ClearKeyCasPlugin.cpp \
- ClearKeyFetcher.cpp \
- ClearKeyLicenseFetcher.cpp \
- ClearKeySessionLibrary.cpp \
- ecm.cpp \
- ecm_generator.cpp \
- JsonAssetLoader.cpp \
- protos/license_protos.proto \
-
-LOCAL_MODULE := libclearkeycasplugin
-
-LOCAL_PROPRIETARY_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := mediacas
-
-LOCAL_SHARED_LIBRARIES := \
- libutils \
- liblog \
- libcrypto \
- libstagefright_foundation \
- libprotobuf-cpp-lite \
-
-LOCAL_HEADER_LIBRARIES := \
- media_plugin_headers
-
-LOCAL_STATIC_LIBRARIES := \
- libjsmn \
-
-LOCAL_MODULE_CLASS := SHARED_LIBRARIES
-
-LOCAL_PROTOC_OPTIMIZE_TYPE := full
-
-define proto_includes
-$(call local-generated-sources-dir)/proto/$(LOCAL_PATH)
-endef
-
-LOCAL_C_INCLUDES += \
- external/jsmn \
- frameworks/av/include \
- frameworks/native/include/media \
- $(call proto_includes)
-
-LOCAL_EXPORT_C_INCLUDE_DIRS := \
- $(call proto_includes)
-
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_SHARED_LIBRARY)
-
-#########################################################################
-# Build unit tests
-
-include $(LOCAL_PATH)/tests/Android.mk
diff --git a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp
index bf35224..af7c367 100644
--- a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp
+++ b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.cpp
@@ -97,7 +97,8 @@
///////////////////////////////////////////////////////////////////////////////
ClearKeyCasPlugin::ClearKeyCasPlugin(
void *appData, CasPluginCallback callback)
- : mCallback(callback), mCallbackExt(NULL), mAppData(appData) {
+ : mCallback(callback), mCallbackExt(NULL), mStatusCallback(NULL),
+ mAppData(appData) {
ALOGV("CTOR");
}
@@ -112,6 +113,13 @@
ClearKeySessionLibrary::get()->destroyPlugin(this);
}
+status_t ClearKeyCasPlugin::setStatusCallback(
+ CasPluginStatusCallback callback) {
+ ALOGV("setStatusCallback");
+ mStatusCallback = callback;
+ return OK;
+}
+
status_t ClearKeyCasPlugin::setPrivateData(const CasData &/*data*/) {
ALOGV("setPrivateData");
@@ -135,6 +143,19 @@
return ClearKeySessionLibrary::get()->addSession(this, sessionId);
}
+status_t ClearKeyCasPlugin::openSession(uint32_t intent, uint32_t mode,
+ CasSessionId* sessionId) {
+ ALOGV("openSession with intent=%d, mode=%d", intent, mode);
+ // Echo the received information to the callback.
+ // Clear key plugin doesn't use any event, echo'ing for testing only.
+ if (mStatusCallback != NULL) {
+ mStatusCallback((void*)mAppData, intent, mode);
+ }
+
+ // Clear key plugin doesn't use intent and mode.
+ return ClearKeySessionLibrary::get()->addSession(this, sessionId);
+}
+
status_t ClearKeyCasPlugin::closeSession(const CasSessionId &sessionId) {
ALOGV("closeSession: sessionId=%s", sessionIdToString(sessionId).string());
std::shared_ptr<ClearKeyCasSession> session =
diff --git a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h
index f48d5b1..c6938e6 100644
--- a/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h
+++ b/drm/mediacas/plugins/clearkey/ClearKeyCasPlugin.h
@@ -71,11 +71,17 @@
ClearKeyCasPlugin(void *appData, CasPluginCallbackExt callback);
virtual ~ClearKeyCasPlugin();
+ virtual status_t setStatusCallback(
+ CasPluginStatusCallback callback) override;
+
virtual status_t setPrivateData(
const CasData &data) override;
virtual status_t openSession(CasSessionId *sessionId) override;
+ virtual status_t openSession(uint32_t intent, uint32_t mode,
+ CasSessionId *sessionId) override;
+
virtual status_t closeSession(
const CasSessionId &sessionId) override;
@@ -105,6 +111,7 @@
std::unique_ptr<KeyFetcher> mKeyFetcher;
CasPluginCallback mCallback;
CasPluginCallbackExt mCallbackExt;
+ CasPluginStatusCallback mStatusCallback;
void* mAppData;
};
diff --git a/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp b/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp
index eaa3390..cb69f91 100644
--- a/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp
+++ b/drm/mediacas/plugins/clearkey/ClearKeyFetcher.cpp
@@ -89,7 +89,7 @@
// asset_id change. If it sends an EcmContainer with 2 Ecms with different
// asset_ids (old and new) then it might be best to prefetch the Emm.
if ((asset_.id() != 0) && (*asset_id != asset_.id())) {
- ALOGW("Asset_id change from %llu to %" PRIu64, asset_.id(), *asset_id);
+ ALOGW("Asset_id change from %" PRIu64 " to %" PRIu64, asset_.id(), *asset_id);
asset_.Clear();
}
diff --git a/drm/mediacas/plugins/clearkey/ecm.cpp b/drm/mediacas/plugins/clearkey/ecm.cpp
index 9fde13a..b3b5218 100644
--- a/drm/mediacas/plugins/clearkey/ecm.cpp
+++ b/drm/mediacas/plugins/clearkey/ecm.cpp
@@ -17,6 +17,8 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "ecm"
+#include <inttypes.h>
+
#include "ecm.h"
#include "ecm_generator.h"
#include "protos/license_protos.pb.h"
@@ -76,7 +78,7 @@
return status;
}
if (asset.id() != asset_from_emm.id()) {
- ALOGE("Asset_id from Emm (%llu) does not match asset_id from Ecm (%llu).",
+ ALOGE("Asset_id from Emm (%" PRIu64 ") does not match asset_id from Ecm (%" PRIu64 ").",
asset_from_emm.id(), asset.id());
return CLEARKEY_STATUS_INVALID_PARAMETER;
}
diff --git a/drm/mediacas/plugins/clearkey/tests/Android.bp b/drm/mediacas/plugins/clearkey/tests/Android.bp
new file mode 100644
index 0000000..575863c
--- /dev/null
+++ b/drm/mediacas/plugins/clearkey/tests/Android.bp
@@ -0,0 +1,45 @@
+//
+// Copyright (C) 2017 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+// http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+
+cc_test {
+ name: "ClearKeyFetcherTest",
+
+ srcs: ["ClearKeyFetcherTest.cpp"],
+
+ vendor: true,
+
+ // LOCAL_LDFLAGS is needed here for the test to use the plugin, because
+ // the plugin is not in standard library search path. Without this .so
+ // loading fails at run-time (linking is okay).
+ ldflags: [
+ "-Wl,--rpath,${ORIGIN}/../../../system/vendor/lib/mediacas",
+ "-Wl,--enable-new-dtags",
+ ],
+
+ shared_libs: [
+ "libutils",
+ "libclearkeycasplugin",
+ "libstagefright_foundation",
+ "libprotobuf-cpp-lite",
+ "liblog",
+ ],
+
+ include_dirs: [
+ "frameworks/av/drm/mediacas/plugins/clearkey",
+ "frameworks/av/include",
+ "frameworks/native/include/media",
+ ],
+}
diff --git a/drm/mediacas/plugins/clearkey/tests/Android.mk b/drm/mediacas/plugins/clearkey/tests/Android.mk
deleted file mode 100644
index e1545af..0000000
--- a/drm/mediacas/plugins/clearkey/tests/Android.mk
+++ /dev/null
@@ -1,45 +0,0 @@
-#
-# Copyright (C) 2017 The Android Open Source Project
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-# http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-#
-LOCAL_PATH:= $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES := \
- ClearKeyFetcherTest.cpp
-
-LOCAL_MODULE := ClearKeyFetcherTest
-LOCAL_VENDOR_MODULE := true
-
-# LOCAL_LDFLAGS is needed here for the test to use the plugin, because
-# the plugin is not in standard library search path. Without this .so
-# loading fails at run-time (linking is okay).
-LOCAL_LDFLAGS := \
- -Wl,--rpath,\$${ORIGIN}/../../../system/vendor/lib/mediacas -Wl,--enable-new-dtags
-
-LOCAL_SHARED_LIBRARIES := \
- libutils libclearkeycasplugin libstagefright_foundation libprotobuf-cpp-lite liblog
-
-LOCAL_C_INCLUDES += \
- $(TOP)/frameworks/av/drm/mediacas/plugins/clearkey \
- $(TOP)/frameworks/av/include \
- $(TOP)/frameworks/native/include/media \
-
-LOCAL_MODULE_TAGS := tests
-
-include $(BUILD_NATIVE_TEST)
-
-
-
diff --git a/drm/mediacas/plugins/mock/Android.bp b/drm/mediacas/plugins/mock/Android.bp
new file mode 100644
index 0000000..e8a3c6f
--- /dev/null
+++ b/drm/mediacas/plugins/mock/Android.bp
@@ -0,0 +1,39 @@
+//
+// Copyright (C) 2017 The Android Open Source Project
+//
+// Licensed under the Apache License, Version 2.0 (the "License");
+// you may not use this file except in compliance with the License.
+// You may obtain a copy of the License at
+//
+// http://www.apache.org/licenses/LICENSE-2.0
+//
+// Unless required by applicable law or agreed to in writing, software
+// distributed under the License is distributed on an "AS IS" BASIS,
+// WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+// See the License for the specific language governing permissions and
+// limitations under the License.
+//
+
+cc_library_shared {
+ name: "libmockcasplugin",
+
+ srcs: [
+ "MockCasPlugin.cpp",
+ "MockSessionLibrary.cpp",
+ ],
+
+ proprietary: true,
+ relative_install_path: "mediacas",
+
+ shared_libs: [
+ "libutils",
+ "liblog",
+ ],
+
+ header_libs: ["media_plugin_headers"],
+
+ include_dirs: [
+ "frameworks/av/include",
+ "frameworks/native/include/media",
+ ],
+}
diff --git a/drm/mediacas/plugins/mock/Android.mk b/drm/mediacas/plugins/mock/Android.mk
deleted file mode 100644
index a1d61da..0000000
--- a/drm/mediacas/plugins/mock/Android.mk
+++ /dev/null
@@ -1,39 +0,0 @@
-#
-# Copyright (C) 2017 The Android Open Source Project
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-# http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-#
-LOCAL_PATH:= $(call my-dir)
-include $(CLEAR_VARS)
-
-LOCAL_SRC_FILES:= \
- MockCasPlugin.cpp \
- MockSessionLibrary.cpp \
-
-LOCAL_MODULE := libmockcasplugin
-
-LOCAL_PROPRIETARY_MODULE := true
-LOCAL_MODULE_RELATIVE_PATH := mediacas
-
-LOCAL_SHARED_LIBRARIES := \
- libutils liblog
-
-LOCAL_HEADER_LIBRARIES := media_plugin_headers
-
-LOCAL_C_INCLUDES += \
- $(TOP)/frameworks/av/include \
- $(TOP)/frameworks/native/include/media \
-
-LOCAL_MODULE_TAGS := optional
-
-include $(BUILD_SHARED_LIBRARY)
diff --git a/drm/mediacas/plugins/mock/MockCasPlugin.cpp b/drm/mediacas/plugins/mock/MockCasPlugin.cpp
index 2964791..f8bab0a 100644
--- a/drm/mediacas/plugins/mock/MockCasPlugin.cpp
+++ b/drm/mediacas/plugins/mock/MockCasPlugin.cpp
@@ -111,6 +111,12 @@
MockSessionLibrary::get()->destroyPlugin(this);
}
+status_t MockCasPlugin::setStatusCallback(
+ CasPluginStatusCallback /*callback*/) {
+ ALOGV("setStatusCallback");
+ return OK;
+}
+
status_t MockCasPlugin::setPrivateData(const CasData& /*data*/) {
ALOGV("setPrivateData");
return OK;
@@ -121,6 +127,13 @@
return MockSessionLibrary::get()->addSession(this, sessionId);
}
+status_t MockCasPlugin::openSession(uint32_t intent, uint32_t mode,
+ CasSessionId* sessionId) {
+ ALOGV("openSession with intent=%d, mode=%d", intent, mode);
+ // Clear key plugin doesn't use intent and mode.
+ return MockSessionLibrary::get()->addSession(this, sessionId);
+}
+
status_t MockCasPlugin::closeSession(const CasSessionId &sessionId) {
ALOGV("closeSession: sessionId=%s", arrayToString(sessionId).string());
Mutex::Autolock lock(mLock);
diff --git a/drm/mediacas/plugins/mock/MockCasPlugin.h b/drm/mediacas/plugins/mock/MockCasPlugin.h
index 74b540c..660fd44 100644
--- a/drm/mediacas/plugins/mock/MockCasPlugin.h
+++ b/drm/mediacas/plugins/mock/MockCasPlugin.h
@@ -65,11 +65,17 @@
MockCasPlugin();
virtual ~MockCasPlugin();
+ virtual status_t setStatusCallback(
+ CasPluginStatusCallback callback) override;
+
virtual status_t setPrivateData(
const CasData &data) override;
virtual status_t openSession(CasSessionId *sessionId) override;
+ virtual status_t openSession(uint32_t intent, uint32_t mode,
+ CasSessionId *sessionId) override;
+
virtual status_t closeSession(
const CasSessionId &sessionId) override;
diff --git a/drm/mediadrm/plugins/clearkey/hidl/Android.bp b/drm/mediadrm/plugins/clearkey/hidl/Android.bp
index e91e918..a153ce2 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/Android.bp
+++ b/drm/mediadrm/plugins/clearkey/hidl/Android.bp
@@ -48,7 +48,6 @@
"libcrypto",
"libhidlbase",
"libhidlmemory",
- "libhidltransport",
"liblog",
"libprotobuf-cpp-lite",
"libutils",
diff --git a/drm/mediadrm/plugins/clearkey/hidl/serviceLazy.cpp b/drm/mediadrm/plugins/clearkey/hidl/serviceLazy.cpp
index 99fd883..a510487 100644
--- a/drm/mediadrm/plugins/clearkey/hidl/serviceLazy.cpp
+++ b/drm/mediadrm/plugins/clearkey/hidl/serviceLazy.cpp
@@ -38,7 +38,7 @@
configureRpcThreadpool(8, true /* callerWillJoin */);
// Setup hwbinder service
- LazyServiceRegistrar serviceRegistrar;
+ auto serviceRegistrar = LazyServiceRegistrar::getInstance();
// Setup hwbinder service
CHECK_EQ(serviceRegistrar.registerService(drmFactory, "clearkey"), android::NO_ERROR)
diff --git a/include/camera b/include/camera
deleted file mode 120000
index 00848e3..0000000
--- a/include/camera
+++ /dev/null
@@ -1 +0,0 @@
-../camera/include/camera/
\ No newline at end of file
diff --git a/include/cpustats b/include/cpustats
deleted file mode 120000
index 4a02d41..0000000
--- a/include/cpustats
+++ /dev/null
@@ -1 +0,0 @@
-../media/libcpustats/include/cpustats/
\ No newline at end of file
diff --git a/include/media/AVSyncSettings.h b/include/media/AVSyncSettings.h
deleted file mode 120000
index bbe211f..0000000
--- a/include/media/AVSyncSettings.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/AVSyncSettings.h
\ No newline at end of file
diff --git a/include/media/AudioAttributes.h b/include/media/AudioAttributes.h
deleted file mode 120000
index 27ba471..0000000
--- a/include/media/AudioAttributes.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioAttributes.h
\ No newline at end of file
diff --git a/include/media/AudioBufferProvider.h b/include/media/AudioBufferProvider.h
deleted file mode 120000
index c4d6e79..0000000
--- a/include/media/AudioBufferProvider.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioBufferProvider.h
\ No newline at end of file
diff --git a/include/media/AudioClient.h b/include/media/AudioClient.h
deleted file mode 120000
index a0530e4..0000000
--- a/include/media/AudioClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioClient.h
\ No newline at end of file
diff --git a/include/media/AudioCommonTypes.h b/include/media/AudioCommonTypes.h
deleted file mode 120000
index ae7c99a..0000000
--- a/include/media/AudioCommonTypes.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioCommonTypes.h
\ No newline at end of file
diff --git a/include/media/AudioEffect.h b/include/media/AudioEffect.h
deleted file mode 120000
index bf52955..0000000
--- a/include/media/AudioEffect.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioEffect.h
\ No newline at end of file
diff --git a/include/media/AudioIoDescriptor.h b/include/media/AudioIoDescriptor.h
deleted file mode 120000
index 68f54c9..0000000
--- a/include/media/AudioIoDescriptor.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioIoDescriptor.h
\ No newline at end of file
diff --git a/include/media/AudioMixer.h b/include/media/AudioMixer.h
deleted file mode 120000
index de839c6..0000000
--- a/include/media/AudioMixer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioMixer.h
\ No newline at end of file
diff --git a/include/media/AudioParameter.h b/include/media/AudioParameter.h
deleted file mode 120000
index a5889e5..0000000
--- a/include/media/AudioParameter.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioParameter.h
\ No newline at end of file
diff --git a/include/media/AudioPolicy.h b/include/media/AudioPolicy.h
deleted file mode 120000
index dd4cd53..0000000
--- a/include/media/AudioPolicy.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioPolicy.h
\ No newline at end of file
diff --git a/include/media/AudioProductStrategy.h b/include/media/AudioProductStrategy.h
deleted file mode 120000
index 6bfaf11..0000000
--- a/include/media/AudioProductStrategy.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioProductStrategy.h
\ No newline at end of file
diff --git a/include/media/AudioRecord.h b/include/media/AudioRecord.h
deleted file mode 120000
index 7939dd3..0000000
--- a/include/media/AudioRecord.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioRecord.h
\ No newline at end of file
diff --git a/include/media/AudioSystem.h b/include/media/AudioSystem.h
deleted file mode 120000
index 9fad2b7..0000000
--- a/include/media/AudioSystem.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioSystem.h
\ No newline at end of file
diff --git a/include/media/AudioTimestamp.h b/include/media/AudioTimestamp.h
deleted file mode 120000
index b6b9278..0000000
--- a/include/media/AudioTimestamp.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioTimestamp.h
\ No newline at end of file
diff --git a/include/media/AudioTrack.h b/include/media/AudioTrack.h
deleted file mode 120000
index 303bfcd..0000000
--- a/include/media/AudioTrack.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioTrack.h
\ No newline at end of file
diff --git a/include/media/AudioVolumeGroup.h b/include/media/AudioVolumeGroup.h
deleted file mode 120000
index d6f1c99..0000000
--- a/include/media/AudioVolumeGroup.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/AudioVolumeGroup.h
\ No newline at end of file
diff --git a/include/media/BufferProviders.h b/include/media/BufferProviders.h
deleted file mode 120000
index 779bb15..0000000
--- a/include/media/BufferProviders.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/BufferProviders.h
\ No newline at end of file
diff --git a/include/media/BufferingSettings.h b/include/media/BufferingSettings.h
deleted file mode 120000
index 409203f..0000000
--- a/include/media/BufferingSettings.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/BufferingSettings.h
\ No newline at end of file
diff --git a/include/media/CharacterEncodingDetector.h b/include/media/CharacterEncodingDetector.h
deleted file mode 120000
index 2b28387..0000000
--- a/include/media/CharacterEncodingDetector.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/CharacterEncodingDetector.h
\ No newline at end of file
diff --git a/include/media/CounterMetric.h b/include/media/CounterMetric.h
deleted file mode 120000
index baba043..0000000
--- a/include/media/CounterMetric.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/CounterMetric.h
\ No newline at end of file
diff --git a/include/media/EventLog.h b/include/media/EventLog.h
deleted file mode 120000
index 9b2c4bf..0000000
--- a/include/media/EventLog.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/utils/include/mediautils/EventLog.h
\ No newline at end of file
diff --git a/include/media/EventMetric.h b/include/media/EventMetric.h
deleted file mode 120000
index 5707d9a..0000000
--- a/include/media/EventMetric.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/EventMetric.h
\ No newline at end of file
diff --git a/include/media/ExtendedAudioBufferProvider.h b/include/media/ExtendedAudioBufferProvider.h
deleted file mode 120000
index d653cc3..0000000
--- a/include/media/ExtendedAudioBufferProvider.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/ExtendedAudioBufferProvider.h
\ No newline at end of file
diff --git a/include/media/IAudioFlinger.h b/include/media/IAudioFlinger.h
deleted file mode 120000
index ef6f5be..0000000
--- a/include/media/IAudioFlinger.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IAudioFlinger.h
\ No newline at end of file
diff --git a/include/media/IAudioFlingerClient.h b/include/media/IAudioFlingerClient.h
deleted file mode 120000
index dc481e8..0000000
--- a/include/media/IAudioFlingerClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IAudioFlingerClient.h
\ No newline at end of file
diff --git a/include/media/IAudioPolicyService.h b/include/media/IAudioPolicyService.h
deleted file mode 120000
index 08101fc..0000000
--- a/include/media/IAudioPolicyService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IAudioPolicyService.h
\ No newline at end of file
diff --git a/include/media/IAudioPolicyServiceClient.h b/include/media/IAudioPolicyServiceClient.h
deleted file mode 120000
index 0d4b3e7..0000000
--- a/include/media/IAudioPolicyServiceClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IAudioPolicyServiceClient.h
\ No newline at end of file
diff --git a/include/media/IAudioTrack.h b/include/media/IAudioTrack.h
deleted file mode 120000
index 7bab1fd..0000000
--- a/include/media/IAudioTrack.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IAudioTrack.h
\ No newline at end of file
diff --git a/include/media/IDataSource.h b/include/media/IDataSource.h
deleted file mode 120000
index 41cdd8b..0000000
--- a/include/media/IDataSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IDataSource.h
\ No newline at end of file
diff --git a/include/media/IEffect.h b/include/media/IEffect.h
deleted file mode 120000
index 2fb8bfb..0000000
--- a/include/media/IEffect.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IEffect.h
\ No newline at end of file
diff --git a/include/media/IEffectClient.h b/include/media/IEffectClient.h
deleted file mode 120000
index b4e39cf..0000000
--- a/include/media/IEffectClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/IEffectClient.h
\ No newline at end of file
diff --git a/include/media/IMediaCodecList.h b/include/media/IMediaCodecList.h
deleted file mode 120000
index 2186312..0000000
--- a/include/media/IMediaCodecList.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaCodecList.h
\ No newline at end of file
diff --git a/include/media/IMediaDeathNotifier.h b/include/media/IMediaDeathNotifier.h
deleted file mode 120000
index ce3b8f0..0000000
--- a/include/media/IMediaDeathNotifier.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaDeathNotifier.h
\ No newline at end of file
diff --git a/include/media/IMediaExtractor.h b/include/media/IMediaExtractor.h
deleted file mode 120000
index 8708c8c..0000000
--- a/include/media/IMediaExtractor.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaExtractor.h
\ No newline at end of file
diff --git a/include/media/IMediaExtractorService.h b/include/media/IMediaExtractorService.h
deleted file mode 120000
index 3ee9f1e..0000000
--- a/include/media/IMediaExtractorService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaExtractorService.h
\ No newline at end of file
diff --git a/include/media/IMediaHTTPConnection.h b/include/media/IMediaHTTPConnection.h
deleted file mode 120000
index 0970c15..0000000
--- a/include/media/IMediaHTTPConnection.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaHTTPConnection.h
\ No newline at end of file
diff --git a/include/media/IMediaHTTPService.h b/include/media/IMediaHTTPService.h
deleted file mode 120000
index b90c34f..0000000
--- a/include/media/IMediaHTTPService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaHTTPService.h
\ No newline at end of file
diff --git a/include/media/IMediaLogService.h b/include/media/IMediaLogService.h
deleted file mode 120000
index 245a29d..0000000
--- a/include/media/IMediaLogService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaLogService.h
\ No newline at end of file
diff --git a/include/media/IMediaMetadataRetriever.h b/include/media/IMediaMetadataRetriever.h
deleted file mode 120000
index 959df1a..0000000
--- a/include/media/IMediaMetadataRetriever.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaMetadataRetriever.h
\ No newline at end of file
diff --git a/include/media/IMediaPlayer.h b/include/media/IMediaPlayer.h
deleted file mode 120000
index 9414d37..0000000
--- a/include/media/IMediaPlayer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaPlayer.h
\ No newline at end of file
diff --git a/include/media/IMediaPlayerClient.h b/include/media/IMediaPlayerClient.h
deleted file mode 120000
index b6547ce..0000000
--- a/include/media/IMediaPlayerClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaPlayerClient.h
\ No newline at end of file
diff --git a/include/media/IMediaPlayerService.h b/include/media/IMediaPlayerService.h
deleted file mode 120000
index 89c96cd..0000000
--- a/include/media/IMediaPlayerService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaPlayerService.h
\ No newline at end of file
diff --git a/include/media/IMediaRecorder.h b/include/media/IMediaRecorder.h
deleted file mode 120000
index 57d192c..0000000
--- a/include/media/IMediaRecorder.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaRecorder.h
\ No newline at end of file
diff --git a/include/media/IMediaRecorderClient.h b/include/media/IMediaRecorderClient.h
deleted file mode 120000
index 89f4359..0000000
--- a/include/media/IMediaRecorderClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaRecorderClient.h
\ No newline at end of file
diff --git a/include/media/IMediaSource.h b/include/media/IMediaSource.h
deleted file mode 120000
index 1330ad3..0000000
--- a/include/media/IMediaSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaSource.h
\ No newline at end of file
diff --git a/include/media/IOMX.h b/include/media/IOMX.h
deleted file mode 120000
index 6d5b375..0000000
--- a/include/media/IOMX.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IOMX.h
\ No newline at end of file
diff --git a/include/media/IRemoteDisplay.h b/include/media/IRemoteDisplay.h
deleted file mode 120000
index 4b0cf10..0000000
--- a/include/media/IRemoteDisplay.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IRemoteDisplay.h
\ No newline at end of file
diff --git a/include/media/IRemoteDisplayClient.h b/include/media/IRemoteDisplayClient.h
deleted file mode 120000
index f29a2ee..0000000
--- a/include/media/IRemoteDisplayClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IRemoteDisplayClient.h
\ No newline at end of file
diff --git a/include/media/IResourceManagerClient.h b/include/media/IResourceManagerClient.h
deleted file mode 120000
index 100af9b..0000000
--- a/include/media/IResourceManagerClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IResourceManagerClient.h
\ No newline at end of file
diff --git a/include/media/IResourceManagerService.h b/include/media/IResourceManagerService.h
deleted file mode 120000
index 9b389c6..0000000
--- a/include/media/IResourceManagerService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IResourceManagerService.h
\ No newline at end of file
diff --git a/include/media/IStreamSource.h b/include/media/IStreamSource.h
deleted file mode 120000
index 4943af9..0000000
--- a/include/media/IStreamSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IStreamSource.h
\ No newline at end of file
diff --git a/include/media/JetPlayer.h b/include/media/JetPlayer.h
deleted file mode 120000
index 5483fda..0000000
--- a/include/media/JetPlayer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/JetPlayer.h
\ No newline at end of file
diff --git a/include/media/LinearMap.h b/include/media/LinearMap.h
deleted file mode 120000
index 30d4ca8..0000000
--- a/include/media/LinearMap.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/LinearMap.h
\ No newline at end of file
diff --git a/include/media/MediaCodecBuffer.h b/include/media/MediaCodecBuffer.h
deleted file mode 120000
index 8c9aa76..0000000
--- a/include/media/MediaCodecBuffer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaCodecBuffer.h
\ No newline at end of file
diff --git a/include/media/MediaCodecInfo.h b/include/media/MediaCodecInfo.h
deleted file mode 120000
index ff44ce4..0000000
--- a/include/media/MediaCodecInfo.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaCodecInfo.h
\ No newline at end of file
diff --git a/include/media/MediaMetadataRetrieverInterface.h b/include/media/MediaMetadataRetrieverInterface.h
deleted file mode 120000
index 1c53511..0000000
--- a/include/media/MediaMetadataRetrieverInterface.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaMetadataRetrieverInterface.h
\ No newline at end of file
diff --git a/include/media/MediaProfiles.h b/include/media/MediaProfiles.h
deleted file mode 120000
index 651c6e6..0000000
--- a/include/media/MediaProfiles.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaProfiles.h
\ No newline at end of file
diff --git a/include/media/MediaRecorderBase.h b/include/media/MediaRecorderBase.h
deleted file mode 120000
index e40f992..0000000
--- a/include/media/MediaRecorderBase.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaRecorderBase.h
\ No newline at end of file
diff --git a/include/media/MediaResource.h b/include/media/MediaResource.h
deleted file mode 120000
index 91346aa..0000000
--- a/include/media/MediaResource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaResource.h
\ No newline at end of file
diff --git a/include/media/MediaResourcePolicy.h b/include/media/MediaResourcePolicy.h
deleted file mode 120000
index 5d165ee..0000000
--- a/include/media/MediaResourcePolicy.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MediaResourcePolicy.h
\ No newline at end of file
diff --git a/include/media/MemoryLeakTrackUtil.h b/include/media/MemoryLeakTrackUtil.h
deleted file mode 120000
index 504173e..0000000
--- a/include/media/MemoryLeakTrackUtil.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MemoryLeakTrackUtil.h
\ No newline at end of file
diff --git a/include/media/Metadata.h b/include/media/Metadata.h
deleted file mode 120000
index e421168..0000000
--- a/include/media/Metadata.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/Metadata.h
\ No newline at end of file
diff --git a/include/media/MidiDeviceInfo.h b/include/media/MidiDeviceInfo.h
deleted file mode 120000
index 95da7cf..0000000
--- a/include/media/MidiDeviceInfo.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MidiDeviceInfo.h
\ No newline at end of file
diff --git a/include/media/MidiIoWrapper.h b/include/media/MidiIoWrapper.h
deleted file mode 120000
index 786ec3d..0000000
--- a/include/media/MidiIoWrapper.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/MidiIoWrapper.h
\ No newline at end of file
diff --git a/include/media/Modulo.h b/include/media/Modulo.h
deleted file mode 120000
index 989c4cb..0000000
--- a/include/media/Modulo.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/Modulo.h
\ No newline at end of file
diff --git a/include/media/OMXBuffer.h b/include/media/OMXBuffer.h
deleted file mode 120000
index 00db207..0000000
--- a/include/media/OMXBuffer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/OMXBuffer.h
\ No newline at end of file
diff --git a/include/media/OMXFenceParcelable.h b/include/media/OMXFenceParcelable.h
deleted file mode 120000
index c4c1b0a..0000000
--- a/include/media/OMXFenceParcelable.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/OMXFenceParcelable.h
\ No newline at end of file
diff --git a/include/media/PluginLoader.h b/include/media/PluginLoader.h
deleted file mode 120000
index 9101735..0000000
--- a/include/media/PluginLoader.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/PluginLoader.h
\ No newline at end of file
diff --git a/include/media/PluginMetricsReporting.h b/include/media/PluginMetricsReporting.h
deleted file mode 120000
index 7d9a7a0..0000000
--- a/include/media/PluginMetricsReporting.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/PluginMetricsReporting.h
\ No newline at end of file
diff --git a/include/media/RecordBufferConverter.h b/include/media/RecordBufferConverter.h
deleted file mode 120000
index 2d7bc0c..0000000
--- a/include/media/RecordBufferConverter.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/RecordBufferConverter.h
\ No newline at end of file
diff --git a/include/media/RingBuffer.h b/include/media/RingBuffer.h
deleted file mode 120000
index 9af28d5..0000000
--- a/include/media/RingBuffer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/RingBuffer.h
\ No newline at end of file
diff --git a/include/media/SingleStateQueue.h b/include/media/SingleStateQueue.h
deleted file mode 120000
index 619f6ee..0000000
--- a/include/media/SingleStateQueue.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/SingleStateQueue.h
\ No newline at end of file
diff --git a/include/media/StringArray.h b/include/media/StringArray.h
deleted file mode 120000
index 616ce6c..0000000
--- a/include/media/StringArray.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/StringArray.h
\ No newline at end of file
diff --git a/include/media/TimeCheck.h b/include/media/TimeCheck.h
deleted file mode 120000
index 85e17f9..0000000
--- a/include/media/TimeCheck.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/utils/include/mediautils/TimeCheck.h
\ No newline at end of file
diff --git a/include/media/ToneGenerator.h b/include/media/ToneGenerator.h
deleted file mode 120000
index 33df0e3..0000000
--- a/include/media/ToneGenerator.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libaudioclient/include/media/ToneGenerator.h
\ No newline at end of file
diff --git a/include/media/TypeConverter.h b/include/media/TypeConverter.h
deleted file mode 120000
index 837af44..0000000
--- a/include/media/TypeConverter.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/TypeConverter.h
\ No newline at end of file
diff --git a/include/media/Visualizer.h b/include/media/Visualizer.h
deleted file mode 120000
index ed2ec15..0000000
--- a/include/media/Visualizer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/Visualizer.h
\ No newline at end of file
diff --git a/include/media/convert.h b/include/media/convert.h
deleted file mode 120000
index cb0d00d..0000000
--- a/include/media/convert.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/convert.h
\ No newline at end of file
diff --git a/include/media/mediametadataretriever.h b/include/media/mediametadataretriever.h
deleted file mode 120000
index b401bab..0000000
--- a/include/media/mediametadataretriever.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/mediametadataretriever.h
\ No newline at end of file
diff --git a/include/media/mediaplayer.h b/include/media/mediaplayer.h
deleted file mode 120000
index 06d537b..0000000
--- a/include/media/mediaplayer.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/mediaplayer.h
\ No newline at end of file
diff --git a/include/media/mediarecorder.h b/include/media/mediarecorder.h
deleted file mode 120000
index a24deb3..0000000
--- a/include/media/mediarecorder.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/mediarecorder.h
\ No newline at end of file
diff --git a/include/media/mediascanner.h b/include/media/mediascanner.h
deleted file mode 120000
index 91479e0..0000000
--- a/include/media/mediascanner.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/mediascanner.h
\ No newline at end of file
diff --git a/include/media/nbaio/AudioBufferProviderSource.h b/include/media/nbaio/AudioBufferProviderSource.h
deleted file mode 120000
index 55841e7..0000000
--- a/include/media/nbaio/AudioBufferProviderSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/AudioBufferProviderSource.h
\ No newline at end of file
diff --git a/include/media/nbaio/AudioStreamInSource.h b/include/media/nbaio/AudioStreamInSource.h
deleted file mode 120000
index f5bcc76..0000000
--- a/include/media/nbaio/AudioStreamInSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/AudioStreamInSource.h
\ No newline at end of file
diff --git a/include/media/nbaio/LibsndfileSink.h b/include/media/nbaio/LibsndfileSink.h
deleted file mode 120000
index 8a13b6c..0000000
--- a/include/media/nbaio/LibsndfileSink.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/LibsndfileSink.h
\ No newline at end of file
diff --git a/include/media/nbaio/LibsndfileSource.h b/include/media/nbaio/LibsndfileSource.h
deleted file mode 120000
index 2750fde..0000000
--- a/include/media/nbaio/LibsndfileSource.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/LibsndfileSource.h
\ No newline at end of file
diff --git a/include/media/nbaio/MonoPipe.h b/include/media/nbaio/MonoPipe.h
deleted file mode 120000
index 4ea43be..0000000
--- a/include/media/nbaio/MonoPipe.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include_mono/media/nbaio/MonoPipe.h
\ No newline at end of file
diff --git a/include/media/nbaio/MonoPipeReader.h b/include/media/nbaio/MonoPipeReader.h
deleted file mode 120000
index 30f426c..0000000
--- a/include/media/nbaio/MonoPipeReader.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include_mono/media/nbaio/MonoPipeReader.h
\ No newline at end of file
diff --git a/include/media/nbaio/Pipe.h b/include/media/nbaio/Pipe.h
deleted file mode 120000
index a4bbbc9..0000000
--- a/include/media/nbaio/Pipe.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/Pipe.h
\ No newline at end of file
diff --git a/include/media/nbaio/PipeReader.h b/include/media/nbaio/PipeReader.h
deleted file mode 120000
index 64b21cf..0000000
--- a/include/media/nbaio/PipeReader.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/PipeReader.h
\ No newline at end of file
diff --git a/include/media/nbaio/SingleStateQueue.h b/include/media/nbaio/SingleStateQueue.h
new file mode 120000
index 0000000..d3e0553
--- /dev/null
+++ b/include/media/nbaio/SingleStateQueue.h
@@ -0,0 +1 @@
+../../../media/libnbaio/include_mono/media/nbaio/SingleStateQueue.h
\ No newline at end of file
diff --git a/include/media/nbaio/SourceAudioBufferProvider.h b/include/media/nbaio/SourceAudioBufferProvider.h
deleted file mode 120000
index 74a3b06..0000000
--- a/include/media/nbaio/SourceAudioBufferProvider.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnbaio/include/media/nbaio/SourceAudioBufferProvider.h
\ No newline at end of file
diff --git a/include/media/nblog/NBLog.h b/include/media/nblog/NBLog.h
deleted file mode 120000
index 3cc366c..0000000
--- a/include/media/nblog/NBLog.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnblog/include/media/nblog/NBLog.h
\ No newline at end of file
diff --git a/include/media/nblog/PerformanceAnalysis.h b/include/media/nblog/PerformanceAnalysis.h
deleted file mode 120000
index 6ead3bc..0000000
--- a/include/media/nblog/PerformanceAnalysis.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnblog/include/media/nblog/PerformanceAnalysis.h
\ No newline at end of file
diff --git a/include/media/nblog/ReportPerformance.h b/include/media/nblog/ReportPerformance.h
deleted file mode 120000
index e9b8e80..0000000
--- a/include/media/nblog/ReportPerformance.h
+++ /dev/null
@@ -1 +0,0 @@
-../../../media/libnblog/include/media/nblog/ReportPerformance.h
\ No newline at end of file
diff --git a/include/mediadrm/CryptoHal.h b/include/mediadrm/CryptoHal.h
deleted file mode 120000
index 92f3137..0000000
--- a/include/mediadrm/CryptoHal.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/CryptoHal.h
\ No newline at end of file
diff --git a/include/mediadrm/DrmHal.h b/include/mediadrm/DrmHal.h
deleted file mode 120000
index 17bb667..0000000
--- a/include/mediadrm/DrmHal.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/DrmHal.h
\ No newline at end of file
diff --git a/include/mediadrm/DrmMetrics.h b/include/mediadrm/DrmMetrics.h
deleted file mode 120000
index abc966b..0000000
--- a/include/mediadrm/DrmMetrics.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/DrmMetrics.h
\ No newline at end of file
diff --git a/include/mediadrm/DrmPluginPath.h b/include/mediadrm/DrmPluginPath.h
deleted file mode 120000
index 9e05194..0000000
--- a/include/mediadrm/DrmPluginPath.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/DrmPluginPath.h
\ No newline at end of file
diff --git a/include/mediadrm/DrmSessionClientInterface.h b/include/mediadrm/DrmSessionClientInterface.h
deleted file mode 120000
index f4e3211..0000000
--- a/include/mediadrm/DrmSessionClientInterface.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/DrmSessionClientInterface.h
\ No newline at end of file
diff --git a/include/mediadrm/DrmSessionManager.h b/include/mediadrm/DrmSessionManager.h
deleted file mode 120000
index f0a47bf..0000000
--- a/include/mediadrm/DrmSessionManager.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/DrmSessionManager.h
\ No newline at end of file
diff --git a/include/mediadrm/ICrypto.h b/include/mediadrm/ICrypto.h
deleted file mode 120000
index b250e07..0000000
--- a/include/mediadrm/ICrypto.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/ICrypto.h
\ No newline at end of file
diff --git a/include/mediadrm/IDrm.h b/include/mediadrm/IDrm.h
deleted file mode 120000
index 841bb1b..0000000
--- a/include/mediadrm/IDrm.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IDrm.h
\ No newline at end of file
diff --git a/include/mediadrm/IDrmClient.h b/include/mediadrm/IDrmClient.h
deleted file mode 120000
index 10aa5c0..0000000
--- a/include/mediadrm/IDrmClient.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IDrmClient.h
\ No newline at end of file
diff --git a/include/mediadrm/IMediaDrmService.h b/include/mediadrm/IMediaDrmService.h
deleted file mode 120000
index f3c260f..0000000
--- a/include/mediadrm/IMediaDrmService.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/IMediaDrmService.h
\ No newline at end of file
diff --git a/include/mediadrm/SharedLibrary.h b/include/mediadrm/SharedLibrary.h
deleted file mode 120000
index 9f8f5a4..0000000
--- a/include/mediadrm/SharedLibrary.h
+++ /dev/null
@@ -1 +0,0 @@
-../../media/libmedia/include/media/SharedLibrary.h
\ No newline at end of file
diff --git a/include/private/media/AudioTrackShared.h b/include/private/media/AudioTrackShared.h
index 5f19f74..1b1f149 100644
--- a/include/private/media/AudioTrackShared.h
+++ b/include/private/media/AudioTrackShared.h
@@ -28,7 +28,7 @@
#include <media/AudioResamplerPublic.h>
#include <media/AudioTimestamp.h>
#include <media/Modulo.h>
-#include <media/SingleStateQueue.h>
+#include <media/nbaio/SingleStateQueue.h>
namespace android {
diff --git a/media/audioserver/Android.mk b/media/audioserver/Android.mk
index 33b36b8..6697cb5 100644
--- a/media/audioserver/Android.mk
+++ b/media/audioserver/Android.mk
@@ -9,12 +9,11 @@
libaaudioservice \
libaudioflinger \
libaudiopolicyservice \
+ libaudioprocessing \
libbinder \
libcutils \
liblog \
libhidlbase \
- libhidltransport \
- libhwbinder \
libmedia \
libmedialogservice \
libmediautils \
@@ -34,13 +33,11 @@
frameworks/av/services/audiopolicy/service \
frameworks/av/services/medialog \
frameworks/av/services/oboeservice \
- frameworks/av/services/radio \
frameworks/av/services/soundtrigger \
frameworks/av/media/libaaudio/include \
frameworks/av/media/libaaudio/src \
frameworks/av/media/libaaudio/src/binding \
frameworks/av/media/libmedia \
- $(call include-path-for, audio-utils) \
external/sonic \
# If AUDIOSERVER_MULTILIB in device.mk is non-empty then it is used to control
diff --git a/media/audioserver/audioserver.rc b/media/audioserver/audioserver.rc
index dfb1a3f..5484613 100644
--- a/media/audioserver/audioserver.rc
+++ b/media/audioserver/audioserver.rc
@@ -2,14 +2,14 @@
class core
user audioserver
# media gid needed for /dev/fm (radio) and for /data/misc/media (tee)
- group audio camera drmrpc inet media mediadrm net_bt net_bt_admin net_bw_acct wakelock
+ group audio camera drmrpc media mediadrm net_bt net_bt_admin net_bw_acct wakelock
capabilities BLOCK_SUSPEND
ioprio rt 4
writepid /dev/cpuset/foreground/tasks /dev/stune/foreground/tasks
- onrestart restart vendor.audio-hal-2-0
+ onrestart restart vendor.audio-hal
onrestart restart vendor.audio-hal-4-0-msd
- # Keep the original service name for backward compatibility when upgrading
- # O-MR1 devices with framework-only.
+ # Keep the original service names for backward compatibility
+ onrestart restart vendor.audio-hal-2-0
onrestart restart audio-hal-2-0
on property:vts.native_server.on=1
diff --git a/media/bufferpool/1.0/Android.bp b/media/bufferpool/1.0/Android.bp
index c7ea70f..f817c76 100644
--- a/media/bufferpool/1.0/Android.bp
+++ b/media/bufferpool/1.0/Android.bp
@@ -16,8 +16,6 @@
"libcutils",
"libfmq",
"libhidlbase",
- "libhwbinder",
- "libhidltransport",
"liblog",
"libutils",
"android.hardware.media.bufferpool@1.0",
diff --git a/media/bufferpool/2.0/Android.bp b/media/bufferpool/2.0/Android.bp
index e8a69c9..97f114a 100644
--- a/media/bufferpool/2.0/Android.bp
+++ b/media/bufferpool/2.0/Android.bp
@@ -16,8 +16,6 @@
"libcutils",
"libfmq",
"libhidlbase",
- "libhwbinder",
- "libhidltransport",
"liblog",
"libutils",
"android.hardware.media.bufferpool@2.0",
diff --git a/media/codec2/components/cmds/Android.bp b/media/codec2/components/cmds/Android.bp
index 35f689e..a081e28 100644
--- a/media/codec2/components/cmds/Android.bp
+++ b/media/codec2/components/cmds/Android.bp
@@ -9,10 +9,15 @@
include_dirs: [
],
+ header_libs: [
+ "libmediadrm_headers",
+ ],
+
shared_libs: [
"libbase",
"libbinder",
"libcutils",
+ "libdatasource",
"libgui",
"liblog",
"libstagefright",
diff --git a/media/codec2/components/cmds/codec2.cpp b/media/codec2/components/cmds/codec2.cpp
index f2cf545..38eaf88 100644
--- a/media/codec2/components/cmds/codec2.cpp
+++ b/media/codec2/components/cmds/codec2.cpp
@@ -30,15 +30,15 @@
#include <binder/IServiceManager.h>
#include <binder/ProcessState.h>
+#include <datasource/DataSourceFactory.h>
#include <media/DataSource.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/IMediaHTTPService.h>
#include <media/MediaSource.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ALooper.h>
#include <media/stagefright/foundation/AMessage.h>
#include <media/stagefright/foundation/AUtils.h>
-#include <media/stagefright/DataSourceFactory.h>
#include <media/stagefright/MediaDefs.h>
#include <media/stagefright/MediaErrors.h>
#include <media/stagefright/MediaExtractorFactory.h>
@@ -418,7 +418,7 @@
const char *filename = argv[k];
sp<DataSource> dataSource =
- DataSourceFactory::CreateFromURI(nullptr /* httpService */, filename);
+ DataSourceFactory::getInstance()->CreateFromURI(nullptr /* httpService */, filename);
if (strncasecmp(filename, "sine:", 5) && dataSource == nullptr) {
fprintf(stderr, "Unable to create data source.\n");
diff --git a/media/codec2/components/hevc/C2SoftHevcEnc.cpp b/media/codec2/components/hevc/C2SoftHevcEnc.cpp
index b129b1b..19ccbf9 100644
--- a/media/codec2/components/hevc/C2SoftHevcEnc.cpp
+++ b/media/codec2/components/hevc/C2SoftHevcEnc.cpp
@@ -42,6 +42,36 @@
constexpr char COMPONENT_NAME[] = "c2.android.hevc.encoder";
+void ParseGop(
+ const C2StreamGopTuning::output &gop,
+ uint32_t *syncInterval, uint32_t *iInterval, uint32_t *maxBframes) {
+ uint32_t syncInt = 1;
+ uint32_t iInt = 1;
+ for (size_t i = 0; i < gop.flexCount(); ++i) {
+ const C2GopLayerStruct &layer = gop.m.values[i];
+ if (layer.count == UINT32_MAX) {
+ syncInt = 0;
+ } else if (syncInt <= UINT32_MAX / (layer.count + 1)) {
+ syncInt *= (layer.count + 1);
+ }
+ if ((layer.type_ & I_FRAME) == 0) {
+ if (layer.count == UINT32_MAX) {
+ iInt = 0;
+ } else if (iInt <= UINT32_MAX / (layer.count + 1)) {
+ iInt *= (layer.count + 1);
+ }
+ }
+ if (layer.type_ == C2Config::picture_type_t(P_FRAME | B_FRAME) && maxBframes) {
+ *maxBframes = layer.count;
+ }
+ }
+ if (syncInterval) {
+ *syncInterval = syncInt;
+ }
+ if (iInterval) {
+ *iInterval = iInt;
+ }
+}
} // namepsace
class C2SoftHevcEnc::IntfImpl : public SimpleInterface<void>::BaseParams {
@@ -60,13 +90,21 @@
setDerivedInstance(this);
addParameter(
+ DefineParam(mGop, C2_PARAMKEY_GOP)
+ .withDefault(C2StreamGopTuning::output::AllocShared(
+ 0 /* flexCount */, 0u /* stream */))
+ .withFields({C2F(mGop, m.values[0].type_).any(),
+ C2F(mGop, m.values[0].count).any()})
+ .withSetter(GopSetter)
+ .build());
+
+ addParameter(
DefineParam(mActualInputDelay, C2_PARAMKEY_INPUT_DELAY)
.withDefault(new C2PortActualDelayTuning::input(
DEFAULT_B_FRAMES + DEFAULT_RC_LOOKAHEAD))
.withFields({C2F(mActualInputDelay, value).inRange(
0, MAX_B_FRAMES + MAX_RC_LOOKAHEAD)})
- .withSetter(
- Setter<decltype(*mActualInputDelay)>::StrictValueWithNoDeps)
+ .calculatedAs(InputDelaySetter, mGop)
.build());
addParameter(
@@ -172,6 +210,17 @@
.build());
}
+ static C2R InputDelaySetter(
+ bool mayBlock,
+ C2P<C2PortActualDelayTuning::input> &me,
+ const C2P<C2StreamGopTuning::output> &gop) {
+ (void)mayBlock;
+ uint32_t maxBframes = 0;
+ ParseGop(gop.v, nullptr, nullptr, &maxBframes);
+ me.set().value = maxBframes + DEFAULT_RC_LOOKAHEAD;
+ return C2R::Ok();
+ }
+
static C2R BitrateSetter(bool mayBlock,
C2P<C2StreamBitrateInfo::output>& me) {
(void)mayBlock;
@@ -270,6 +319,18 @@
return C2R::Ok();
}
+ static C2R GopSetter(bool mayBlock, C2P<C2StreamGopTuning::output> &me) {
+ (void)mayBlock;
+ for (size_t i = 0; i < me.v.flexCount(); ++i) {
+ const C2GopLayerStruct &layer = me.v.m.values[0];
+ if (layer.type_ == C2Config::picture_type_t(P_FRAME | B_FRAME)
+ && layer.count > MAX_B_FRAMES) {
+ me.set().m.values[i].count = MAX_B_FRAMES;
+ }
+ }
+ return C2R::Ok();
+ }
+
UWORD32 getProfile_l() const {
switch (mProfileLevel->profile) {
case PROFILE_HEVC_MAIN: [[fallthrough]];
@@ -338,6 +399,9 @@
std::shared_ptr<C2StreamQualityTuning::output> getQuality_l() const {
return mQuality;
}
+ std::shared_ptr<C2StreamGopTuning::output> getGop_l() const {
+ return mGop;
+ }
private:
std::shared_ptr<C2StreamUsageTuning::input> mUsage;
@@ -350,6 +414,7 @@
std::shared_ptr<C2StreamQualityTuning::output> mQuality;
std::shared_ptr<C2StreamProfileLevelInfo::output> mProfileLevel;
std::shared_ptr<C2StreamSyncFrameIntervalTuning::output> mSyncFramePeriod;
+ std::shared_ptr<C2StreamGopTuning::output> mGop;
};
static size_t GetCPUCoreCount() {
@@ -449,7 +514,25 @@
ALOGE("HEVC default init failed : 0x%x", err);
return C2_CORRUPTED;
}
-
+ mBframes = 0;
+ if (mGop && mGop->flexCount() > 0) {
+ uint32_t syncInterval = 1;
+ uint32_t iInterval = 1;
+ uint32_t maxBframes = 0;
+ ParseGop(*mGop, &syncInterval, &iInterval, &maxBframes);
+ if (syncInterval > 0) {
+ ALOGD("Updating IDR interval from GOP: old %u new %u", mIDRInterval, syncInterval);
+ mIDRInterval = syncInterval;
+ }
+ if (iInterval > 0) {
+ ALOGD("Updating I interval from GOP: old %u new %u", mIInterval, iInterval);
+ mIInterval = iInterval;
+ }
+ if (mBframes != maxBframes) {
+ ALOGD("Updating max B frames from GOP: old %u new %u", mBframes, maxBframes);
+ mBframes = maxBframes;
+ }
+ }
// update configuration
mEncParams.s_src_prms.i4_width = mSize->width;
mEncParams.s_src_prms.i4_height = mSize->height;
@@ -463,12 +546,20 @@
mBitrate->value << 1;
mEncParams.s_tgt_lyr_prms.as_tgt_params[0].i4_codec_level = mHevcEncLevel;
mEncParams.s_coding_tools_prms.i4_max_i_open_gop_period = mIDRInterval;
- mEncParams.s_coding_tools_prms.i4_max_cra_open_gop_period = mIDRInterval;
+ mEncParams.s_coding_tools_prms.i4_max_cra_open_gop_period = mIInterval;
mIvVideoColorFormat = IV_YUV_420P;
mEncParams.s_multi_thrd_prms.i4_max_num_cores = mNumCores;
mEncParams.s_out_strm_prms.i4_codec_profile = mHevcEncProfile;
mEncParams.s_lap_prms.i4_rc_look_ahead_pics = DEFAULT_RC_LOOKAHEAD;
- mEncParams.s_coding_tools_prms.i4_max_temporal_layers = DEFAULT_B_FRAMES;
+ if (mBframes == 0) {
+ mEncParams.s_coding_tools_prms.i4_max_temporal_layers = 0;
+ } else if (mBframes <= 2) {
+ mEncParams.s_coding_tools_prms.i4_max_temporal_layers = 1;
+ } else if (mBframes <= 6) {
+ mEncParams.s_coding_tools_prms.i4_max_temporal_layers = 2;
+ } else {
+ mEncParams.s_coding_tools_prms.i4_max_temporal_layers = 3;
+ }
switch (mBitrateMode->value) {
case C2Config::BITRATE_IGNORE:
@@ -523,6 +614,7 @@
c2_status_t C2SoftHevcEnc::initEncoder() {
CHECK(!mCodecCtx);
+
{
IntfImpl::Lock lock = mIntf->lock();
mSize = mIntf->getSize_l();
@@ -532,8 +624,10 @@
mHevcEncProfile = mIntf->getProfile_l();
mHevcEncLevel = mIntf->getLevel_l();
mIDRInterval = mIntf->getSyncFramePeriod_l();
+ mIInterval = mIntf->getSyncFramePeriod_l();
mComplexity = mIntf->getComplexity_l();
mQuality = mIntf->getQuality_l();
+ mGop = mIntf->getGop_l();
}
c2_status_t status = initEncParams();
diff --git a/media/codec2/components/hevc/C2SoftHevcEnc.h b/media/codec2/components/hevc/C2SoftHevcEnc.h
index f2c7642..140b4a9 100644
--- a/media/codec2/components/hevc/C2SoftHevcEnc.h
+++ b/media/codec2/components/hevc/C2SoftHevcEnc.h
@@ -67,6 +67,8 @@
ihevce_static_cfg_params_t mEncParams;
size_t mNumCores;
UWORD32 mIDRInterval;
+ UWORD32 mIInterval;
+ UWORD32 mBframes;
IV_COLOR_FORMAT_T mIvVideoColorFormat;
UWORD32 mHevcEncProfile;
UWORD32 mHevcEncLevel;
@@ -85,7 +87,7 @@
std::shared_ptr<C2StreamBitrateModeTuning::output> mBitrateMode;
std::shared_ptr<C2StreamComplexityTuning::output> mComplexity;
std::shared_ptr<C2StreamQualityTuning::output> mQuality;
-
+ std::shared_ptr<C2StreamGopTuning::output> mGop;
#ifdef FILE_DUMP_ENABLE
char mInFile[200];
char mOutFile[200];
diff --git a/media/codec2/components/mpeg4_h263/C2SoftMpeg4Enc.cpp b/media/codec2/components/mpeg4_h263/C2SoftMpeg4Enc.cpp
index 36053f6..54c8c47 100644
--- a/media/codec2/components/mpeg4_h263/C2SoftMpeg4Enc.cpp
+++ b/media/codec2/components/mpeg4_h263/C2SoftMpeg4Enc.cpp
@@ -517,9 +517,11 @@
if (layout.planes[layout.PLANE_Y].colInc == 1
&& layout.planes[layout.PLANE_U].colInc == 1
&& layout.planes[layout.PLANE_V].colInc == 1
+ && yStride == align(width, 16)
&& uStride == vStride
&& yStride == 2 * vStride) {
- // I420 compatible - planes are already set up above
+ // I420 compatible with yStride being equal to aligned width
+ // planes are already set up above
break;
}
diff --git a/media/codec2/components/vpx/C2SoftVpxEnc.cpp b/media/codec2/components/vpx/C2SoftVpxEnc.cpp
index 6dab70b..ebc7a8f 100644
--- a/media/codec2/components/vpx/C2SoftVpxEnc.cpp
+++ b/media/codec2/components/vpx/C2SoftVpxEnc.cpp
@@ -514,7 +514,7 @@
return;
}
vpx_img_wrap(&raw_frame, VPX_IMG_FMT_I420, stride, vstride,
- mStrideAlign, (uint8_t*)rView->data()[0]);
+ mStrideAlign, mConversionBuffer.data());
vpx_img_set_rect(&raw_frame, 0, 0, width, height);
} else {
ALOGE("Conversion buffer is too small: %u x %u for %zu",
diff --git a/media/codec2/hidl/1.0/utils/Android.bp b/media/codec2/hidl/1.0/utils/Android.bp
index f1f1536..bdff29a 100644
--- a/media/codec2/hidl/1.0/utils/Android.bp
+++ b/media/codec2/hidl/1.0/utils/Android.bp
@@ -80,8 +80,6 @@
"libcodec2_vndk",
"libcutils",
"libhidlbase",
- "libhidltransport",
- "libhwbinder",
"liblog",
"libstagefright_bufferpool@2.0.1",
"libstagefright_bufferqueue_helper",
diff --git a/media/codec2/hidl/client/Android.bp b/media/codec2/hidl/client/Android.bp
index e184223..89c1c4a 100644
--- a/media/codec2/hidl/client/Android.bp
+++ b/media/codec2/hidl/client/Android.bp
@@ -17,7 +17,6 @@
"libcutils",
"libgui",
"libhidlbase",
- "libhidltransport",
"liblog",
"libstagefright_bufferpool@2.0.1",
"libui",
diff --git a/media/codec2/hidl/services/Android.bp b/media/codec2/hidl/services/Android.bp
index 216525e..0403a1f 100644
--- a/media/codec2/hidl/services/Android.bp
+++ b/media/codec2/hidl/services/Android.bp
@@ -17,8 +17,6 @@
"libcodec2_hidl@1.0",
"libcodec2_vndk",
"libhidlbase",
- "libhidltransport",
- "libhwbinder",
"liblog",
"libstagefright_omx",
"libstagefright_xmlparser",
diff --git a/media/codec2/sfplugin/Android.bp b/media/codec2/sfplugin/Android.bp
index 9c84c71..ec576c9 100644
--- a/media/codec2/sfplugin/Android.bp
+++ b/media/codec2/sfplugin/Android.bp
@@ -22,6 +22,8 @@
header_libs: [
"libcodec2_internal",
+ "libmediadrm_headers",
+ "media_ndk_headers",
],
shared_libs: [
@@ -39,7 +41,7 @@
"libhidlallocatorutils",
"libhidlbase",
"liblog",
- "libmedia",
+ "libmedia_codeclist",
"libmedia_omx",
"libsfplugin_ccodec_utils",
"libstagefright_bufferqueue_helper",
diff --git a/media/codec2/sfplugin/CCodecBufferChannel.h b/media/codec2/sfplugin/CCodecBufferChannel.h
index ee3455d..c0fa138 100644
--- a/media/codec2/sfplugin/CCodecBufferChannel.h
+++ b/media/codec2/sfplugin/CCodecBufferChannel.h
@@ -29,7 +29,6 @@
#include <codec2/hidl/client.h>
#include <media/stagefright/foundation/Mutexed.h>
#include <media/stagefright/CodecBase.h>
-#include <media/ICrypto.h>
#include "CCodecBuffers.h"
#include "InputSurfaceWrapper.h"
diff --git a/media/codec2/sfplugin/CCodecBuffers.cpp b/media/codec2/sfplugin/CCodecBuffers.cpp
index 26c702d..ed8b832 100644
--- a/media/codec2/sfplugin/CCodecBuffers.cpp
+++ b/media/codec2/sfplugin/CCodecBuffers.cpp
@@ -878,9 +878,10 @@
switch (c2buffer->data().type()) {
case C2BufferData::LINEAR: {
uint32_t size = kLinearBufferSize;
- const C2ConstLinearBlock &block = c2buffer->data().linearBlocks().front();
- if (block.size() < kMaxLinearBufferSize / 2) {
- size = block.size() * 2;
+ const std::vector<C2ConstLinearBlock> &linear_blocks = c2buffer->data().linearBlocks();
+ const uint32_t block_size = linear_blocks.front().size();
+ if (block_size < kMaxLinearBufferSize / 2) {
+ size = block_size * 2;
} else {
size = kMaxLinearBufferSize;
}
diff --git a/media/codec2/sfplugin/Codec2Buffer.h b/media/codec2/sfplugin/Codec2Buffer.h
index 36dcab9..6f87101 100644
--- a/media/codec2/sfplugin/Codec2Buffer.h
+++ b/media/codec2/sfplugin/Codec2Buffer.h
@@ -25,7 +25,7 @@
#include <media/hardware/VideoAPI.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/MediaCodecBuffer.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
namespace android {
diff --git a/media/codec2/sfplugin/tests/Android.bp b/media/codec2/sfplugin/tests/Android.bp
index be7f55c..b6eb2b4 100644
--- a/media/codec2/sfplugin/tests/Android.bp
+++ b/media/codec2/sfplugin/tests/Android.bp
@@ -33,6 +33,10 @@
"frameworks/av/media/codec2/sfplugin",
],
+ header_libs: [
+ "libmediadrm_headers",
+ ],
+
shared_libs: [
"libbinder",
"libcodec2",
diff --git a/media/codec2/sfplugin/tests/MediaCodec_sanity_test.cpp b/media/codec2/sfplugin/tests/MediaCodec_sanity_test.cpp
index ba3687b..6deede0 100644
--- a/media/codec2/sfplugin/tests/MediaCodec_sanity_test.cpp
+++ b/media/codec2/sfplugin/tests/MediaCodec_sanity_test.cpp
@@ -21,7 +21,7 @@
#include <binder/ProcessState.h>
#include <gtest/gtest.h>
#include <gui/Surface.h>
-#include <media/ICrypto.h>
+#include <mediadrm/ICrypto.h>
#include <media/MediaCodecBuffer.h>
#include <media/hardware/VideoAPI.h>
#include <media/stagefright/MediaCodec.h>
diff --git a/media/extractors/flac/FLACExtractor.h b/media/extractors/flac/FLACExtractor.h
index 5a73d20..223d359 100644
--- a/media/extractors/flac/FLACExtractor.h
+++ b/media/extractors/flac/FLACExtractor.h
@@ -17,7 +17,6 @@
#ifndef FLAC_EXTRACTOR_H_
#define FLAC_EXTRACTOR_H_
-#include <media/DataSourceBase.h>
#include <media/MediaExtractorPluginApi.h>
#include <media/MediaExtractorPluginHelper.h>
#include <media/NdkMediaFormat.h>
diff --git a/media/extractors/midi/Android.bp b/media/extractors/midi/Android.bp
index 7d42e70..d36cb49 100644
--- a/media/extractors/midi/Android.bp
+++ b/media/extractors/midi/Android.bp
@@ -6,6 +6,10 @@
"frameworks/av/media/libstagefright/include",
],
+ header_libs: [
+ "libmedia_headers",
+ ],
+
shared_libs: [
"liblog",
"libmediandk",
diff --git a/media/extractors/midi/MidiExtractor.h b/media/extractors/midi/MidiExtractor.h
index 2e78086..b486fc6 100644
--- a/media/extractors/midi/MidiExtractor.h
+++ b/media/extractors/midi/MidiExtractor.h
@@ -17,7 +17,6 @@
#ifndef MIDI_EXTRACTOR_H_
#define MIDI_EXTRACTOR_H_
-#include <media/DataSourceBase.h>
#include <media/MediaExtractorPluginApi.h>
#include <media/MediaExtractorPluginHelper.h>
#include <media/stagefright/MediaBufferBase.h>
diff --git a/media/extractors/mp4/SampleIterator.cpp b/media/extractors/mp4/SampleIterator.cpp
index 2890b26..85fbf97 100644
--- a/media/extractors/mp4/SampleIterator.cpp
+++ b/media/extractors/mp4/SampleIterator.cpp
@@ -22,7 +22,6 @@
#include <arpa/inet.h>
-#include <media/DataSourceBase.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ByteUtils.h>
@@ -355,7 +354,7 @@
if (offset > 0) {
*time += offset;
} else {
- *time -= (offset == INT64_MIN ? INT64_MAX : (-offset));
+ *time -= (offset == INT32_MIN ? INT64_MAX : (-offset));
}
*duration = mTTSDuration;
diff --git a/media/extractors/mpeg2/Android.bp b/media/extractors/mpeg2/Android.bp
index 0f0c72c..1d9e1e6 100644
--- a/media/extractors/mpeg2/Android.bp
+++ b/media/extractors/mpeg2/Android.bp
@@ -16,6 +16,7 @@
"android.hardware.cas.native@1.0",
"android.hidl.token@1.0-utils",
"android.hidl.allocator@1.0",
+ "libcrypto",
"libhidlmemory",
"libhidlbase",
"liblog",
@@ -23,13 +24,13 @@
],
header_libs: [
+ "libaudioclient_headers",
"libbase_headers",
"libstagefright_headers",
"libmedia_headers",
],
static_libs: [
- "libcrypto",
"libstagefright_foundation_without_imemory",
"libstagefright_mpeg2support",
"libutils",
diff --git a/media/extractors/mpeg2/MPEG2PSExtractor.cpp b/media/extractors/mpeg2/MPEG2PSExtractor.cpp
index 92ba039..002a855 100644
--- a/media/extractors/mpeg2/MPEG2PSExtractor.cpp
+++ b/media/extractors/mpeg2/MPEG2PSExtractor.cpp
@@ -23,7 +23,6 @@
#include "mpeg2ts/AnotherPacketSource.h"
#include "mpeg2ts/ESQueue.h"
-#include <media/DataSourceBase.h>
#include <media/stagefright/foundation/ABitReader.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/ADebug.h>
diff --git a/media/libaaudio/Android.bp b/media/libaaudio/Android.bp
index 16958f9..140052f 100644
--- a/media/libaaudio/Android.bp
+++ b/media/libaaudio/Android.bp
@@ -24,7 +24,7 @@
ndk_library {
name: "libaaudio",
// deliberately includes symbols from AAudioTesting.h
- symbol_file: "libaaudio.map.txt",
+ symbol_file: "src/libaaudio.map.txt",
first_version: "26",
unversioned_until: "current",
}
@@ -32,6 +32,5 @@
cc_library_headers {
name: "libaaudio_headers",
export_include_dirs: ["include"],
- version_script: "libaaudio.map.txt",
}
diff --git a/media/libaaudio/include/aaudio/AAudio.h b/media/libaaudio/include/aaudio/AAudio.h
index ee5d089..8173e3c 100644
--- a/media/libaaudio/include/aaudio/AAudio.h
+++ b/media/libaaudio/include/aaudio/AAudio.h
@@ -472,6 +472,8 @@
* This is intended for developers to use when debugging.
* It is not for display to users.
*
+ * Available since API level 26.
+ *
* @return pointer to a text representation of an AAudio result code.
*/
AAUDIO_API const char * AAudio_convertResultToText(aaudio_result_t returnCode) __INTRODUCED_IN(26);
@@ -482,6 +484,8 @@
* This is intended for developers to use when debugging.
* It is not for display to users.
*
+ * Available since API level 26.
+ *
* @return pointer to a text representation of an AAudio state.
*/
AAUDIO_API const char * AAudio_convertStreamStateToText(aaudio_stream_state_t state)
@@ -502,6 +506,8 @@
* chosen by the device when it is opened.
*
* AAudioStreamBuilder_delete() must be called when you are done using the builder.
+ *
+ * Available since API level 26.
*/
AAUDIO_API aaudio_result_t AAudio_createStreamBuilder(AAudioStreamBuilder** builder)
__INTRODUCED_IN(26);
@@ -513,6 +519,8 @@
* The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED},
* in which case the primary device will be used.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param deviceId device identifier or {@link #AAUDIO_UNSPECIFIED}
*/
@@ -530,6 +538,8 @@
* If an exact value is specified then an opened stream will use that value.
* If a stream cannot be opened with the specified value then the open will fail.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param sampleRate frames per second. Common rates include 44100 and 48000 Hz.
*/
@@ -547,6 +557,8 @@
* If an exact value is specified then an opened stream will use that value.
* If a stream cannot be opened with the specified value then the open will fail.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param channelCount Number of channels desired.
*/
@@ -556,6 +568,8 @@
/**
* Identical to AAudioStreamBuilder_setChannelCount().
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param samplesPerFrame Number of samples in a frame.
*/
@@ -573,6 +587,8 @@
* If an exact value is specified then an opened stream will use that value.
* If a stream cannot be opened with the specified value then the open will fail.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param format common formats are {@link #AAUDIO_FORMAT_PCM_FLOAT} and
* {@link #AAUDIO_FORMAT_PCM_I16}.
@@ -588,6 +604,8 @@
* The requested sharing mode may not be available.
* The application can query for the actual mode after the stream is opened.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param sharingMode {@link #AAUDIO_SHARING_MODE_SHARED} or {@link #AAUDIO_SHARING_MODE_EXCLUSIVE}
*/
@@ -599,6 +617,8 @@
*
* The default, if you do not call this function, is {@link #AAUDIO_DIRECTION_OUTPUT}.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param direction {@link #AAUDIO_DIRECTION_OUTPUT} or {@link #AAUDIO_DIRECTION_INPUT}
*/
@@ -611,6 +631,8 @@
*
* The default, if you do not call this function, is {@link #AAUDIO_UNSPECIFIED}.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param numFrames the desired buffer capacity in frames or {@link #AAUDIO_UNSPECIFIED}
*/
@@ -629,6 +651,8 @@
* You can call AAudioStream_getPerformanceMode()
* to find out the final mode for the stream.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param mode the desired performance mode, eg. {@link #AAUDIO_PERFORMANCE_MODE_LOW_LATENCY}
*/
@@ -644,7 +668,7 @@
*
* The default, if you do not call this function, is {@link #AAUDIO_USAGE_MEDIA}.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param usage the desired usage, eg. {@link #AAUDIO_USAGE_GAME}
@@ -661,7 +685,7 @@
*
* The default, if you do not call this function, is {@link #AAUDIO_CONTENT_TYPE_MUSIC}.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param contentType the type of audio data, eg. {@link #AAUDIO_CONTENT_TYPE_SPEECH}
@@ -681,7 +705,7 @@
* That is because VOICE_RECOGNITION is the preset with the lowest latency
* on many platforms.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param inputPreset the desired configuration for recording
@@ -697,7 +721,7 @@
* Note that an application can also set its global policy, in which case the most restrictive
* policy is always applied. See {@link android.media.AudioAttributes#setAllowedCapturePolicy(int)}
*
- * Added in API level 29.
+ * Available since API level 29.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param inputPreset the desired level of opt-out from being captured.
@@ -727,7 +751,7 @@
*
* Allocated session IDs will always be positive and nonzero.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param builder reference provided by AAudio_createStreamBuilder()
* @param sessionId an allocated sessionID or {@link #AAUDIO_SESSION_ID_ALLOCATE}
@@ -826,6 +850,8 @@
*
* Note that the AAudio callbacks will never be called simultaneously from multiple threads.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param callback pointer to a function that will process audio data.
* @param userData pointer to an application data structure that will be passed
@@ -854,6 +880,8 @@
* If you do call this function then the requested size should be less than
* half the buffer capacity, to allow double buffering.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param numFrames the desired buffer size in frames or {@link #AAUDIO_UNSPECIFIED}
*/
@@ -905,6 +933,8 @@
*
* Note that the AAudio callbacks will never be called simultaneously from multiple threads.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param callback pointer to a function that will be called if an error occurs.
* @param userData pointer to an application data structure that will be passed
@@ -919,6 +949,8 @@
* AAudioStream_close() must be called when finished with the stream to recover
* the memory and to free the associated resources.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @param stream pointer to a variable to receive the new stream reference
* @return {@link #AAUDIO_OK} or a negative error.
@@ -929,6 +961,8 @@
/**
* Delete the resources associated with the StreamBuilder.
*
+ * Available since API level 26.
+ *
* @param builder reference provided by AAudio_createStreamBuilder()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -942,6 +976,8 @@
/**
* Free the resources associated with a stream created by AAudioStreamBuilder_openStream()
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -954,6 +990,8 @@
* After this call the state will be in {@link #AAUDIO_STREAM_STATE_STARTING} or
* {@link #AAUDIO_STREAM_STATE_STARTED}.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -969,6 +1007,8 @@
* This will return {@link #AAUDIO_ERROR_UNIMPLEMENTED} for input streams.
* For input streams use AAudioStream_requestStop().
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -984,6 +1024,8 @@
*
* This will return {@link #AAUDIO_ERROR_UNIMPLEMENTED} for input streams.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -995,6 +1037,8 @@
* After this call the state will be in {@link #AAUDIO_STREAM_STATE_STOPPING} or
* {@link #AAUDIO_STREAM_STATE_STOPPED}.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return {@link #AAUDIO_OK} or a negative error.
*/
@@ -1008,6 +1052,8 @@
* call AAudioStream_waitForStateChange() with currentState
* set to {@link #AAUDIO_STREAM_STATE_UNKNOWN} and a zero timeout.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
*/
AAUDIO_API aaudio_stream_state_t AAudioStream_getState(AAudioStream* stream) __INTRODUCED_IN(26);
@@ -1028,6 +1074,8 @@
* }
* </code></pre>
*
+ * Available since API level 26.
+ *
* @param stream A reference provided by AAudioStreamBuilder_openStream()
* @param inputState The state we want to avoid.
* @param nextState Pointer to a variable that will be set to the new state.
@@ -1056,6 +1104,8 @@
*
* If the call times out then zero or a partial frame count will be returned.
*
+ * Available since API level 26.
+ *
* @param stream A stream created using AAudioStreamBuilder_openStream().
* @param buffer The address of the first sample.
* @param numFrames Number of frames to read. Only complete frames will be written.
@@ -1079,6 +1129,8 @@
*
* If the call times out then zero or a partial frame count will be returned.
*
+ * Available since API level 26.
+ *
* @param stream A stream created using AAudioStreamBuilder_openStream().
* @param buffer The address of the first sample.
* @param numFrames Number of frames to write. Only complete frames will be written.
@@ -1104,6 +1156,8 @@
* You can check the return value or call AAudioStream_getBufferSizeInFrames()
* to see what the actual final size is.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @param numFrames requested number of frames that can be filled without blocking
* @return actual buffer size in frames or a negative error
@@ -1114,6 +1168,8 @@
/**
* Query the maximum number of frames that can be filled without blocking.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return buffer size in frames.
*/
@@ -1129,6 +1185,8 @@
* For some endpoints, the burst size can vary dynamically.
* But these tend to be devices with high latency.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return burst size
*/
@@ -1137,6 +1195,8 @@
/**
* Query maximum buffer capacity in frames.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return buffer capacity in frames
*/
@@ -1158,6 +1218,8 @@
* {@link #AAUDIO_UNSPECIFIED} indicates that the callback buffer size for this stream
* may vary from one dataProc callback to the next.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return callback buffer size in frames or {@link #AAUDIO_UNSPECIFIED}
*/
@@ -1175,12 +1237,16 @@
* Note that some INPUT devices may not support this function.
* In that case a 0 will always be returned.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return the underrun or overrun count
*/
AAUDIO_API int32_t AAudioStream_getXRunCount(AAudioStream* stream) __INTRODUCED_IN(26);
/**
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual sample rate
*/
@@ -1190,6 +1256,8 @@
* A stream has one or more channels of data.
* A frame will contain one sample for each channel.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual number of channels
*/
@@ -1198,18 +1266,24 @@
/**
* Identical to AAudioStream_getChannelCount().
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual number of samples frame
*/
AAUDIO_API int32_t AAudioStream_getSamplesPerFrame(AAudioStream* stream) __INTRODUCED_IN(26);
/**
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual device ID
*/
AAUDIO_API int32_t AAudioStream_getDeviceId(AAudioStream* stream) __INTRODUCED_IN(26);
/**
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual data format
*/
@@ -1217,6 +1291,9 @@
/**
* Provide actual sharing mode.
+ *
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return actual sharing mode
*/
@@ -1226,12 +1303,16 @@
/**
* Get the performance mode used by the stream.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
*/
AAUDIO_API aaudio_performance_mode_t AAudioStream_getPerformanceMode(AAudioStream* stream)
__INTRODUCED_IN(26);
/**
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return direction
*/
@@ -1245,6 +1326,8 @@
*
* The frame position is monotonically increasing.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return frames written
*/
@@ -1258,6 +1341,8 @@
*
* The frame position is monotonically increasing.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return frames read
*/
@@ -1281,7 +1366,7 @@
*
* The sessionID for a stream should not change once the stream has been opened.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return session ID or {@link #AAUDIO_SESSION_ID_NONE}
@@ -1304,6 +1389,8 @@
*
* The position and time passed back are monotonically increasing.
*
+ * Available since API level 26.
+ *
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @param clockid CLOCK_MONOTONIC or CLOCK_BOOTTIME
* @param framePosition pointer to a variable to receive the position
@@ -1316,7 +1403,7 @@
/**
* Return the use case for the stream.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return frames read
@@ -1326,7 +1413,7 @@
/**
* Return the content type for the stream.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return content type, for example {@link #AAUDIO_CONTENT_TYPE_MUSIC}
@@ -1337,7 +1424,7 @@
/**
* Return the input preset for the stream.
*
- * Added in API level 28.
+ * Available since API level 28.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return input preset, for example {@link #AAUDIO_INPUT_PRESET_CAMCORDER}
@@ -1349,7 +1436,7 @@
* Return the policy that determines whether the audio may or may not be captured
* by other apps or the system.
*
- * Added in API level 29.
+ * Available since API level 29.
*
* @param stream reference provided by AAudioStreamBuilder_openStream()
* @return the allowed capture policy, for example {@link #AAUDIO_ALLOW_CAPTURE_BY_ALL}
diff --git a/media/libaaudio/src/Android.bp b/media/libaaudio/src/Android.bp
index 4090286..850b1d0 100644
--- a/media/libaaudio/src/Android.bp
+++ b/media/libaaudio/src/Android.bp
@@ -10,14 +10,76 @@
"legacy",
"utility",
],
- export_include_dirs: ["."],
- header_libs: ["libaaudio_headers"],
+ header_libs: [
+ "libaaudio_headers",
+ ],
export_header_lib_headers: ["libaaudio_headers"],
+ version_script: "libaaudio.map.txt",
srcs: [
+ "core/AAudioAudio.cpp",
+ ],
+
+ cflags: [
+ "-Wno-unused-parameter",
+ "-Wall",
+ "-Werror",
+
+ // By default, all symbols are hidden.
+ // "-fvisibility=hidden",
+ // AAUDIO_API is used to explicitly export a function or a variable as a visible symbol.
+ "-DAAUDIO_API=__attribute__((visibility(\"default\")))",
+ ],
+
+ shared_libs: [
+ "libaaudio_internal",
+ "libaudioclient",
+ "libaudioutils",
+ "liblog",
+ "libcutils",
+ "libutils",
+ "libbinder",
+ ],
+}
+
+cc_library {
+ name: "libaaudio_internal",
+
+ local_include_dirs: [
+ "binding",
+ "client",
+ "core",
+ "fifo",
+ "legacy",
+ "utility",
+ ],
+
+ export_include_dirs: ["."],
+ header_libs: [
+ "libaaudio_headers",
+ "libmedia_headers"
+ ],
+ export_header_lib_headers: ["libaaudio_headers"],
+
+ shared_libs: [
+ "libaudioclient",
+ "libaudioutils",
+ "liblog",
+ "libcutils",
+ "libutils",
+ "libbinder",
+ ],
+
+ cflags: [
+ "-Wno-unused-parameter",
+ "-Wall",
+ "-Werror",
+ ],
+
+ srcs: [
+ "core/AudioGlobal.cpp",
"core/AudioStream.cpp",
"core/AudioStreamBuilder.cpp",
- "core/AAudioAudio.cpp",
"core/AAudioStreamParameters.cpp",
"legacy/AudioStreamLegacy.cpp",
"legacy/AudioStreamRecord.cpp",
@@ -54,25 +116,4 @@
"flowgraph/SourceI16.cpp",
"flowgraph/SourceI24.cpp",
],
-
- cflags: [
- "-Wno-unused-parameter",
- "-Wall",
- "-Werror",
-
- // By default, all symbols are hidden.
- // "-fvisibility=hidden",
- // AAUDIO_API is used to explicitly export a function or a variable as a visible symbol.
- "-DAAUDIO_API=__attribute__((visibility(\"default\")))",
- ],
-
- shared_libs: [
- "libaudioclient",
- "libaudioutils",
- "liblog",
- "libcutils",
- "libutils",
- "libbinder",
- "libaudiomanager",
- ],
}
diff --git a/media/libaaudio/src/client/AudioStreamInternal.cpp b/media/libaaudio/src/client/AudioStreamInternal.cpp
index 52eadd4..fb276c2 100644
--- a/media/libaaudio/src/client/AudioStreamInternal.cpp
+++ b/media/libaaudio/src/client/AudioStreamInternal.cpp
@@ -36,6 +36,7 @@
#include "binding/AAudioStreamConfiguration.h"
#include "binding/IAAudioService.h"
#include "binding/AAudioServiceMessage.h"
+#include "core/AudioGlobal.h"
#include "core/AudioStreamBuilder.h"
#include "fifo/FifoBuffer.h"
#include "utility/AudioClock.h"
diff --git a/media/libaaudio/src/core/AAudioAudio.cpp b/media/libaaudio/src/core/AAudioAudio.cpp
index 44d5122..8040e6a 100644
--- a/media/libaaudio/src/core/AAudioAudio.cpp
+++ b/media/libaaudio/src/core/AAudioAudio.cpp
@@ -27,6 +27,7 @@
#include <aaudio/AAudioTesting.h>
#include "AudioClock.h"
+#include "AudioGlobal.h"
#include "AudioStreamBuilder.h"
#include "AudioStream.h"
#include "binding/AAudioCommon.h"
@@ -45,63 +46,14 @@
return AAUDIO_ERROR_NULL; \
}
-#define AAUDIO_CASE_ENUM(name) case name: return #name
-
AAUDIO_API const char * AAudio_convertResultToText(aaudio_result_t returnCode) {
- switch (returnCode) {
- AAUDIO_CASE_ENUM(AAUDIO_OK);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_DISCONNECTED);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_ILLEGAL_ARGUMENT);
- // reserved
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_INTERNAL);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_STATE);
- // reserved
- // reserved
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_HANDLE);
- // reserved
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_UNIMPLEMENTED);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_UNAVAILABLE);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_FREE_HANDLES);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_MEMORY);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_NULL);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_TIMEOUT);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_WOULD_BLOCK);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_FORMAT);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_OUT_OF_RANGE);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_SERVICE);
- AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_RATE);
- }
- return "Unrecognized AAudio error.";
+ return AudioGlobal_convertResultToText(returnCode);
}
AAUDIO_API const char * AAudio_convertStreamStateToText(aaudio_stream_state_t state) {
- switch (state) {
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_UNINITIALIZED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_UNKNOWN);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_OPEN);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STARTING);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STARTED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_PAUSING);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_PAUSED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_FLUSHING);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_FLUSHED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STOPPING);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STOPPED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_DISCONNECTED);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_CLOSING);
- AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_CLOSED);
- }
- return "Unrecognized AAudio state.";
+ return AudioGlobal_convertStreamStateToText(state);
}
-#undef AAUDIO_CASE_ENUM
-
-
-/******************************************
- * Static globals.
- */
-static aaudio_policy_t s_MMapPolicy = AAUDIO_UNSPECIFIED;
-
static AudioStream *convertAAudioStreamToAudioStream(AAudioStream* stream)
{
return (AudioStream*) stream;
@@ -543,23 +495,11 @@
}
AAUDIO_API aaudio_policy_t AAudio_getMMapPolicy() {
- return s_MMapPolicy;
+ return AudioGlobal_getMMapPolicy();
}
AAUDIO_API aaudio_result_t AAudio_setMMapPolicy(aaudio_policy_t policy) {
- aaudio_result_t result = AAUDIO_OK;
- switch(policy) {
- case AAUDIO_UNSPECIFIED:
- case AAUDIO_POLICY_NEVER:
- case AAUDIO_POLICY_AUTO:
- case AAUDIO_POLICY_ALWAYS:
- s_MMapPolicy = policy;
- break;
- default:
- result = AAUDIO_ERROR_ILLEGAL_ARGUMENT;
- break;
- }
- return result;
+ return AudioGlobal_setMMapPolicy(policy);
}
AAUDIO_API bool AAudioStream_isMMapUsed(AAudioStream* stream)
diff --git a/media/libaaudio/src/core/AudioGlobal.cpp b/media/libaaudio/src/core/AudioGlobal.cpp
new file mode 100644
index 0000000..e6d9a0d
--- /dev/null
+++ b/media/libaaudio/src/core/AudioGlobal.cpp
@@ -0,0 +1,99 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#include <aaudio/AAudio.h>
+#include <aaudio/AAudioTesting.h>
+
+#include "AudioGlobal.h"
+
+/******************************************
+ * Static globals.
+ */
+namespace aaudio {
+
+static aaudio_policy_t g_MMapPolicy = AAUDIO_UNSPECIFIED;
+
+aaudio_policy_t AudioGlobal_getMMapPolicy() {
+ return g_MMapPolicy;
+}
+
+aaudio_result_t AudioGlobal_setMMapPolicy(aaudio_policy_t policy) {
+ aaudio_result_t result = AAUDIO_OK;
+ switch(policy) {
+ case AAUDIO_UNSPECIFIED:
+ case AAUDIO_POLICY_NEVER:
+ case AAUDIO_POLICY_AUTO:
+ case AAUDIO_POLICY_ALWAYS:
+ g_MMapPolicy = policy;
+ break;
+ default:
+ result = AAUDIO_ERROR_ILLEGAL_ARGUMENT;
+ break;
+ }
+ return result;
+}
+
+#define AAUDIO_CASE_ENUM(name) case name: return #name
+
+const char* AudioGlobal_convertResultToText(aaudio_result_t returnCode) {
+ switch (returnCode) {
+ AAUDIO_CASE_ENUM(AAUDIO_OK);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_DISCONNECTED);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_ILLEGAL_ARGUMENT);
+ // reserved
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INTERNAL);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_STATE);
+ // reserved
+ // reserved
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_HANDLE);
+ // reserved
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_UNIMPLEMENTED);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_UNAVAILABLE);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_FREE_HANDLES);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_MEMORY);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_NULL);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_TIMEOUT);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_WOULD_BLOCK);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_FORMAT);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_OUT_OF_RANGE);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_NO_SERVICE);
+ AAUDIO_CASE_ENUM(AAUDIO_ERROR_INVALID_RATE);
+ }
+ return "Unrecognized AAudio error.";
+}
+
+const char* AudioGlobal_convertStreamStateToText(aaudio_stream_state_t state) {
+ switch (state) {
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_UNINITIALIZED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_UNKNOWN);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_OPEN);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STARTING);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STARTED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_PAUSING);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_PAUSED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_FLUSHING);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_FLUSHED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STOPPING);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_STOPPED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_DISCONNECTED);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_CLOSING);
+ AAUDIO_CASE_ENUM(AAUDIO_STREAM_STATE_CLOSED);
+ }
+ return "Unrecognized AAudio state.";
+}
+
+#undef AAUDIO_CASE_ENUM
+
+} // namespace aaudio
diff --git a/media/libaaudio/src/core/AudioGlobal.h b/media/libaaudio/src/core/AudioGlobal.h
new file mode 100644
index 0000000..312cef2
--- /dev/null
+++ b/media/libaaudio/src/core/AudioGlobal.h
@@ -0,0 +1,34 @@
+/*
+ * Copyright 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+#ifndef AAUDIO_AUDIOGLOBAL_H
+#define AAUDIO_AUDIOGLOBAL_H
+
+#include <aaudio/AAudio.h>
+#include <aaudio/AAudioTesting.h>
+
+
+namespace aaudio {
+
+aaudio_policy_t AudioGlobal_getMMapPolicy();
+aaudio_result_t AudioGlobal_setMMapPolicy(aaudio_policy_t policy);
+
+const char* AudioGlobal_convertResultToText(aaudio_result_t returnCode);
+const char* AudioGlobal_convertStreamStateToText(aaudio_stream_state_t state);
+
+}
+
+#endif // AAUDIO_AUDIOGLOBAL_H
+
diff --git a/media/libaaudio/src/core/AudioStream.cpp b/media/libaaudio/src/core/AudioStream.cpp
index 9b77223..5303631 100644
--- a/media/libaaudio/src/core/AudioStream.cpp
+++ b/media/libaaudio/src/core/AudioStream.cpp
@@ -25,8 +25,9 @@
#include "AudioStreamBuilder.h"
#include "AudioStream.h"
#include "AudioClock.h"
+#include "AudioGlobal.h"
-using namespace aaudio;
+namespace aaudio {
// Sequential number assigned to streams solely for debugging purposes.
@@ -51,7 +52,7 @@
|| getState() == AAUDIO_STREAM_STATE_UNINITIALIZED
|| getState() == AAUDIO_STREAM_STATE_DISCONNECTED),
"~AudioStream() - still in use, state = %s",
- AAudio_convertStreamStateToText(getState()));
+ AudioGlobal_convertStreamStateToText(getState()));
mPlayerBase->clearParentReference(); // remove reference to this AudioStream
}
@@ -155,7 +156,7 @@
case AAUDIO_STREAM_STATE_CLOSED:
default:
ALOGW("safePause() stream not running, state = %s",
- AAudio_convertStreamStateToText(getState()));
+ AudioGlobal_convertStreamStateToText(getState()));
return AAUDIO_ERROR_INVALID_STATE;
}
@@ -240,7 +241,7 @@
case AAUDIO_STREAM_STATE_CLOSED:
default:
ALOGW("%s() stream not running, state = %s", __func__,
- AAudio_convertStreamStateToText(getState()));
+ AudioGlobal_convertStreamStateToText(getState()));
return AAUDIO_ERROR_INVALID_STATE;
}
@@ -488,3 +489,5 @@
void AudioStream::MyPlayerBase::destroy() {
unregisterWithAudioManager();
}
+
+} // namespace aaudio
diff --git a/media/libaaudio/src/core/AudioStreamBuilder.cpp b/media/libaaudio/src/core/AudioStreamBuilder.cpp
index 08f4958..44f45b3 100644
--- a/media/libaaudio/src/core/AudioStreamBuilder.cpp
+++ b/media/libaaudio/src/core/AudioStreamBuilder.cpp
@@ -27,6 +27,7 @@
#include "binding/AAudioBinderClient.h"
#include "client/AudioStreamInternalCapture.h"
#include "client/AudioStreamInternalPlay.h"
+#include "core/AudioGlobal.h"
#include "core/AudioStream.h"
#include "core/AudioStreamBuilder.h"
#include "legacy/AudioStreamRecord.h"
@@ -112,7 +113,7 @@
}
// The API setting is the highest priority.
- aaudio_policy_t mmapPolicy = AAudio_getMMapPolicy();
+ aaudio_policy_t mmapPolicy = AudioGlobal_getMMapPolicy();
// If not specified then get from a system property.
if (mmapPolicy == AAUDIO_UNSPECIFIED) {
mmapPolicy = AAudioProperty_getMMapPolicy();
diff --git a/media/libaaudio/libaaudio.map.txt b/media/libaaudio/src/libaaudio.map.txt
similarity index 100%
rename from media/libaaudio/libaaudio.map.txt
rename to media/libaaudio/src/libaaudio.map.txt
diff --git a/media/libaaudio/src/utility/AAudioUtilities.cpp b/media/libaaudio/src/utility/AAudioUtilities.cpp
index 96ed56a..cdd02c0 100644
--- a/media/libaaudio/src/utility/AAudioUtilities.cpp
+++ b/media/libaaudio/src/utility/AAudioUtilities.cpp
@@ -24,6 +24,7 @@
#include <utils/Errors.h>
#include "aaudio/AAudio.h"
+#include "core/AudioGlobal.h"
#include <aaudio/AAudioTesting.h>
#include <math.h>
#include <system/audio-base.h>
@@ -355,7 +356,7 @@
case AAUDIO_STREAM_STATE_DISCONNECTED:
default:
ALOGE("can only flush stream when PAUSED, OPEN or STOPPED, state = %s",
- AAudio_convertStreamStateToText(state));
+ aaudio::AudioGlobal_convertStreamStateToText(state));
result = AAUDIO_ERROR_INVALID_STATE;
break;
}
diff --git a/media/libaaudio/tests/Android.bp b/media/libaaudio/tests/Android.bp
index 6101e99..19cd0a0 100644
--- a/media/libaaudio/tests/Android.bp
+++ b/media/libaaudio/tests/Android.bp
@@ -11,7 +11,7 @@
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_marshalling.cpp"],
shared_libs: [
- "libaaudio",
+ "libaaudio_internal",
"libbinder",
"libcutils",
"libutils",
@@ -23,7 +23,7 @@
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_clock_model.cpp"],
shared_libs: [
- "libaaudio",
+ "libaaudio_internal",
"libaudioutils",
"libcutils",
"libutils",
@@ -34,7 +34,7 @@
name: "test_block_adapter",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_block_adapter.cpp"],
- shared_libs: ["libaaudio"],
+ shared_libs: ["libaaudio_internal"],
}
cc_test {
@@ -170,7 +170,7 @@
name: "test_atomic_fifo",
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_atomic_fifo.cpp"],
- shared_libs: ["libaaudio"],
+ shared_libs: ["libaaudio_internal"],
}
cc_test {
@@ -178,7 +178,7 @@
defaults: ["libaaudio_tests_defaults"],
srcs: ["test_flowgraph.cpp"],
shared_libs: [
- "libaaudio",
+ "libaaudio_internal",
"libbinder",
"libcutils",
"libutils",
diff --git a/media/libaudioclient/Android.bp b/media/libaudioclient/Android.bp
index 03bd6ce..32904bb 100644
--- a/media/libaudioclient/Android.bp
+++ b/media/libaudioclient/Android.bp
@@ -42,7 +42,7 @@
// AIDL files for audioclient interfaces
// The headers for these interfaces will be available to any modules that
// include libaudioclient, at the path "aidl/package/path/BnFoo.h"
- "aidl/android/media/IAudioRecord.aidl",
+ ":libaudioclient_aidl_private",
":libaudioclient_aidl",
"AudioEffect.cpp",
@@ -84,6 +84,7 @@
header_libs: [
"libaudioclient_headers",
"libbase_headers",
+ "libmedia_headers",
],
export_header_lib_headers: ["libaudioclient_headers"],
@@ -110,4 +111,15 @@
srcs: [
"aidl/android/media/IPlayer.aidl",
],
+ path: "aidl",
+}
+
+// Used to strip the "aidl/" from the path, so the build system can predict the
+// output filename.
+filegroup {
+ name: "libaudioclient_aidl_private",
+ srcs: [
+ "aidl/android/media/IAudioRecord.aidl",
+ ],
+ path: "aidl",
}
diff --git a/media/libaudioclient/IAudioFlinger.cpp b/media/libaudioclient/IAudioFlinger.cpp
index dd95e34..e3e64af 100644
--- a/media/libaudioclient/IAudioFlinger.cpp
+++ b/media/libaudioclient/IAudioFlinger.cpp
@@ -24,8 +24,8 @@
#include <binder/IPCThreadState.h>
#include <binder/Parcel.h>
-#include <media/TimeCheck.h>
#include <mediautils/ServiceUtilities.h>
+#include <mediautils/TimeCheck.h>
#include "IAudioFlinger.h"
namespace android {
diff --git a/media/libaudioclient/IAudioPolicyService.cpp b/media/libaudioclient/IAudioPolicyService.cpp
index 64f0aca..7cc95e5 100644
--- a/media/libaudioclient/IAudioPolicyService.cpp
+++ b/media/libaudioclient/IAudioPolicyService.cpp
@@ -26,8 +26,8 @@
#include <binder/Parcel.h>
#include <media/AudioEffect.h>
#include <media/IAudioPolicyService.h>
-#include <media/TimeCheck.h>
#include <mediautils/ServiceUtilities.h>
+#include <mediautils/TimeCheck.h>
#include <system/audio.h>
namespace android {
diff --git a/media/libaudioclient/include/media/AudioMixer.h b/media/libaudioclient/include/media/AudioMixer.h
index 783eef3..3f7cd48 100644
--- a/media/libaudioclient/include/media/AudioMixer.h
+++ b/media/libaudioclient/include/media/AudioMixer.h
@@ -18,87 +18,38 @@
#ifndef ANDROID_AUDIO_MIXER_H
#define ANDROID_AUDIO_MIXER_H
-#include <map>
#include <pthread.h>
-#include <sstream>
#include <stdint.h>
#include <sys/types.h>
-#include <unordered_map>
-#include <vector>
#include <android/os/IExternalVibratorService.h>
-#include <media/AudioBufferProvider.h>
-#include <media/AudioResampler.h>
-#include <media/AudioResamplerPublic.h>
+#include <media/AudioMixerBase.h>
#include <media/BufferProviders.h>
-#include <system/audio.h>
-#include <utils/Compat.h>
#include <utils/threads.h>
// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
-#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT
-
-// This must match frameworks/av/services/audioflinger/Configuration.h
-#define FLOAT_AUX
+#define MAX_GAIN_INT AudioMixerBase::UNITY_GAIN_INT
namespace android {
-namespace NBLog {
-class Writer;
-} // namespace NBLog
-
// ----------------------------------------------------------------------------
-class AudioMixer
+// AudioMixer extends AudioMixerBase by adding support for down- and up-mixing
+// and time stretch that are implemented via Effects HAL, and adding support
+// for haptic channels which depends on Vibrator service. This is the version
+// that is used by Audioflinger.
+
+class AudioMixer : public AudioMixerBase
{
public:
- // Do not change these unless underlying code changes.
- // This mixer has a hard-coded upper limit of 8 channels for output.
- static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8;
- static constexpr uint32_t MAX_NUM_VOLUMES = FCC_2; // stereo volume only
// maximum number of channels supported for the content
static const uint32_t MAX_NUM_CHANNELS_TO_DOWNMIX = AUDIO_CHANNEL_COUNT_MAX;
- static const uint16_t UNITY_GAIN_INT = 0x1000;
- static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
-
- enum { // names
- // setParameter targets
- TRACK = 0x3000,
- RESAMPLE = 0x3001,
- RAMP_VOLUME = 0x3002, // ramp to new volume
- VOLUME = 0x3003, // don't ramp
- TIMESTRETCH = 0x3004,
-
- // set Parameter names
- // for target TRACK
- CHANNEL_MASK = 0x4000,
- FORMAT = 0x4001,
- MAIN_BUFFER = 0x4002,
- AUX_BUFFER = 0x4003,
- DOWNMIX_TYPE = 0X4004,
- MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
- MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
+ enum { // extension of AudioMixerBase parameters
+ DOWNMIX_TYPE = 0x4004,
// for haptic
HAPTIC_ENABLED = 0x4007, // Set haptic data from this track should be played or not.
HAPTIC_INTENSITY = 0x4008, // Set the intensity to play haptic data.
- // for target RESAMPLE
- SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
- // parameter 'value' is the new sample rate in Hz.
- // Only creates a sample rate converter the first time that
- // the track sample rate is different from the mix sample rate.
- // If the new sample rate is the same as the mix sample rate,
- // and a sample rate converter already exists,
- // then the sample rate converter remains present but is a no-op.
- RESET = 0x4101, // Reset sample rate converter without changing sample rate.
- // This clears out the resampler's input buffer.
- REMOVE = 0x4102, // Remove the sample rate converter on this track name;
- // the track is restored to the mix sample rate.
- // for target RAMP_VOLUME and VOLUME (8 channels max)
- // FIXME use float for these 3 to improve the dynamic range
- VOLUME0 = 0x4200,
- VOLUME1 = 0x4201,
- AUXLEVEL = 0x4210,
// for target TIMESTRETCH
PLAYBACK_RATE = 0x4300, // Configure timestretch on this track name;
// parameter 'value' is a pointer to the new playback rate.
@@ -131,142 +82,23 @@
}
AudioMixer(size_t frameCount, uint32_t sampleRate)
- : mSampleRate(sampleRate)
- , mFrameCount(frameCount) {
+ : AudioMixerBase(frameCount, sampleRate) {
pthread_once(&sOnceControl, &sInitRoutine);
}
- // Create a new track in the mixer.
- //
- // \param name a unique user-provided integer associated with the track.
- // If name already exists, the function will abort.
- // \param channelMask output channel mask.
- // \param format PCM format
- // \param sessionId Session id for the track. Tracks with the same
- // session id will be submixed together.
- //
- // \return OK on success.
- // BAD_VALUE if the format does not satisfy isValidFormat()
- // or the channelMask does not satisfy isValidChannelMask().
- status_t create(
- int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId);
+ bool isValidChannelMask(audio_channel_mask_t channelMask) const override;
- bool exists(int name) const {
- return mTracks.count(name) > 0;
- }
-
- // Free an allocated track by name.
- void destroy(int name);
-
- // Enable or disable an allocated track by name
- void enable(int name);
- void disable(int name);
-
- void setParameter(int name, int target, int param, void *value);
-
- void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
-
- void process() {
- for (const auto &pair : mTracks) {
- // Clear contracted buffer before processing if contracted channels are saved
- const std::shared_ptr<Track> &t = pair.second;
- if (t->mKeepContractedChannels) {
- t->clearContractedBuffer();
- }
- }
- (this->*mHook)();
- processHapticData();
- }
-
- size_t getUnreleasedFrames(int name) const;
-
- std::string trackNames() const {
- std::stringstream ss;
- for (const auto &pair : mTracks) {
- ss << pair.first << " ";
- }
- return ss.str();
- }
-
- void setNBLogWriter(NBLog::Writer *logWriter) {
- mNBLogWriter = logWriter;
- }
-
- static inline bool isValidFormat(audio_format_t format) {
- switch (format) {
- case AUDIO_FORMAT_PCM_8_BIT:
- case AUDIO_FORMAT_PCM_16_BIT:
- case AUDIO_FORMAT_PCM_24_BIT_PACKED:
- case AUDIO_FORMAT_PCM_32_BIT:
- case AUDIO_FORMAT_PCM_FLOAT:
- return true;
- default:
- return false;
- }
- }
-
- static inline bool isValidChannelMask(audio_channel_mask_t channelMask) {
- return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
- }
+ void setParameter(int name, int target, int param, void *value) override;
+ void setBufferProvider(int name, AudioBufferProvider* bufferProvider);
private:
- /* For multi-format functions (calls template functions
- * in AudioMixerOps.h). The template parameters are as follows:
- *
- * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * USEFLOATVOL (set to true if float volume is used)
- * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27)
- */
-
- enum {
- // FIXME this representation permits up to 8 channels
- NEEDS_CHANNEL_COUNT__MASK = 0x00000007,
- };
-
- enum {
- NEEDS_CHANNEL_1 = 0x00000000, // mono
- NEEDS_CHANNEL_2 = 0x00000001, // stereo
-
- // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
-
- NEEDS_MUTE = 0x00000100,
- NEEDS_RESAMPLE = 0x00001000,
- NEEDS_AUX = 0x00010000,
- };
-
- // hook types
- enum {
- PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere
- };
-
- enum {
- TRACKTYPE_NOP,
- TRACKTYPE_RESAMPLE,
- TRACKTYPE_NORESAMPLE,
- TRACKTYPE_NORESAMPLEMONO,
- };
-
- // process hook functionality
- using process_hook_t = void(AudioMixer::*)();
-
- struct Track;
- using hook_t = void(Track::*)(int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
-
- struct Track {
- Track()
- : bufferProvider(nullptr)
- {
- // TODO: move additional initialization here.
- }
+ struct Track : public TrackBase {
+ Track() : TrackBase() {}
~Track()
{
- // bufferProvider, mInputBufferProvider need not be deleted.
- mResampler.reset(nullptr);
+ // mInputBufferProvider need not be deleted.
// Ensure the order of destruction of buffer providers as they
// release the upstream provider in the destructor.
mTimestretchBufferProvider.reset(nullptr);
@@ -277,13 +109,12 @@
mAdjustChannelsBufferProvider.reset(nullptr);
}
- bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
- bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
- bool doesResample() const { return mResampler.get() != nullptr; }
- void resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); }
- void adjustVolumeRamp(bool aux, bool useFloat = false);
- size_t getUnreleasedFrames() const { return mResampler.get() != nullptr ?
- mResampler->getUnreleasedFrames() : 0; };
+ uint32_t getOutputChannelCount() override {
+ return mDownmixerBufferProvider.get() != nullptr ? mMixerChannelCount : channelCount;
+ }
+ uint32_t getMixerChannelCount() override {
+ return mMixerChannelCount + mMixerHapticChannelCount;
+ }
status_t prepareForDownmix();
void unprepareForDownmix();
@@ -297,51 +128,9 @@
bool setPlaybackRate(const AudioPlaybackRate &playbackRate);
void reconfigureBufferProviders();
- static hook_t getTrackHook(int trackType, uint32_t channelCount,
- audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
-
- void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-
- template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
- typename TO, typename TI, typename TA>
- void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp);
-
- uint32_t needs;
-
- // TODO: Eventually remove legacy integer volume settings
- union {
- int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
- int32_t volumeRL;
- };
-
- int32_t prevVolume[MAX_NUM_VOLUMES];
- int32_t volumeInc[MAX_NUM_VOLUMES];
- int32_t auxInc;
- int32_t prevAuxLevel;
- int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
-
- uint16_t frameCount;
-
- uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
- uint8_t unused_padding; // formerly format, was always 16
- uint16_t enabled; // actually bool
- audio_channel_mask_t channelMask;
-
- // actual buffer provider used by the track hooks, see DownmixerBufferProvider below
- // for how the Track buffer provider is wrapped by another one when dowmixing is required
- AudioBufferProvider* bufferProvider;
-
- mutable AudioBufferProvider::Buffer buffer; // 8 bytes
-
- hook_t hook;
- const void *mIn; // current location in buffer
-
- std::unique_ptr<AudioResampler> mResampler;
- uint32_t sampleRate;
- int32_t* mainBuffer;
- int32_t* auxBuffer;
-
/* Buffer providers are constructed to translate the track input data as needed.
+ * See DownmixerBufferProvider below for how the Track buffer provider
+ * is wrapped by another one when dowmixing is required.
*
* TODO: perhaps make a single PlaybackConverterProvider class to move
* all pre-mixer track buffer conversions outside the AudioMixer class.
@@ -363,7 +152,7 @@
* the downmixer requirements to the mixer engine input requirements.
* 7) mTimestretchBufferProvider: Adds timestretching for playback rate
*/
- AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
+ AudioBufferProvider* mInputBufferProvider; // externally provided buffer provider.
// TODO: combine mAdjustChannelsBufferProvider and
// mContractChannelsNonDestructiveBufferProvider
std::unique_ptr<PassthruBufferProvider> mAdjustChannelsBufferProvider;
@@ -373,27 +162,10 @@
std::unique_ptr<PassthruBufferProvider> mPostDownmixReformatBufferProvider;
std::unique_ptr<PassthruBufferProvider> mTimestretchBufferProvider;
- int32_t sessionId;
-
- audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
- audio_format_t mFormat; // input track format
- audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
- // each track must be converted to this format.
audio_format_t mDownmixRequiresFormat; // required downmixer format
// AUDIO_FORMAT_PCM_16_BIT if 16 bit necessary
// AUDIO_FORMAT_INVALID if no required format
- float mVolume[MAX_NUM_VOLUMES]; // floating point set volume
- float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
- float mVolumeInc[MAX_NUM_VOLUMES]; // floating point volume increment
-
- float mAuxLevel; // floating point set aux level
- float mPrevAuxLevel; // floating point prev aux level
- float mAuxInc; // floating point aux increment
-
- audio_channel_mask_t mMixerChannelMask;
- uint32_t mMixerChannelCount;
-
AudioPlaybackRate mPlaybackRate;
// Haptic
@@ -440,76 +212,23 @@
return 0.0f;
}
}
-
- private:
- // hooks
- void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
- void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
- void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
-
- void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
- void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
-
- // multi-format track hooks
- template <int MIXTYPE, typename TO, typename TI, typename TA>
- void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
- template <int MIXTYPE, typename TO, typename TI, typename TA>
- void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
};
- // TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
- static constexpr int BLOCKSIZE = 16;
-
- bool setChannelMasks(int name,
- audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
-
- // Called when track info changes and a new process hook should be determined.
- void invalidate() {
- mHook = &AudioMixer::process__validate;
+ inline std::shared_ptr<Track> getTrack(int name) {
+ return std::static_pointer_cast<Track>(mTracks[name]);
}
- void process__validate();
- void process__nop();
- void process__genericNoResampling();
- void process__genericResampling();
- void process__oneTrack16BitsStereoNoResampling();
+ std::shared_ptr<TrackBase> preCreateTrack() override;
+ status_t postCreateTrack(TrackBase *track) override;
- template <int MIXTYPE, typename TO, typename TI, typename TA>
- void process__noResampleOneTrack();
+ void preProcess() override;
+ void postProcess() override;
- void processHapticData();
-
- static process_hook_t getProcessHook(int processType, uint32_t channelCount,
- audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
-
- static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
- void *in, audio_format_t mixerInFormat, size_t sampleCount);
+ bool setChannelMasks(int name,
+ audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) override;
static void sInitRoutine();
- // initialization constants
- const uint32_t mSampleRate;
- const size_t mFrameCount;
-
- NBLog::Writer *mNBLogWriter = nullptr; // associated NBLog::Writer
-
- process_hook_t mHook = &AudioMixer::process__nop; // one of process__*, never nullptr
-
- // the size of the type (int32_t) should be the largest of all types supported
- // by the mixer.
- std::unique_ptr<int32_t[]> mOutputTemp;
- std::unique_ptr<int32_t[]> mResampleTemp;
-
- // track names grouped by main buffer, in no particular order of main buffer.
- // however names for a particular main buffer are in order (by construction).
- std::unordered_map<void * /* mainBuffer */, std::vector<int /* name */>> mGroups;
-
- // track names that are enabled, in increasing order (by construction).
- std::vector<int /* name */> mEnabled;
-
- // track smart pointers, by name, in increasing order of name.
- std::map<int /* name */, std::shared_ptr<Track>> mTracks;
-
static pthread_once_t sOnceControl; // initialized in constructor by first new
};
diff --git a/media/libaudioclient/include/media/AudioParameter.h b/media/libaudioclient/include/media/AudioParameter.h
index 24837e3..3c190f2 100644
--- a/media/libaudioclient/include/media/AudioParameter.h
+++ b/media/libaudioclient/include/media/AudioParameter.h
@@ -67,9 +67,12 @@
// keyAudioLanguagePreferred: Preferred audio language
static const char * const keyAudioLanguagePreferred;
- // keyStreamConnect / Disconnect: value is an int in audio_devices_t
- static const char * const keyStreamConnect;
- static const char * const keyStreamDisconnect;
+ // keyDeviceConnect / Disconnect: value is an int in audio_devices_t
+ static const char * const keyDeviceConnect;
+ static const char * const keyDeviceDisconnect;
+ // Need to be here because vendors still use them.
+ static const char * const keyStreamConnect; // Deprecated: DO NOT USE.
+ static const char * const keyStreamDisconnect; // Deprecated: DO NOT USE.
// For querying stream capabilities. All the returned values are lists.
// keyStreamSupportedFormats: audio_format_t
diff --git a/media/libmedia/include/media/ExtendedAudioBufferProvider.h b/media/libaudioclient/include/media/ExtendedAudioBufferProvider.h
similarity index 100%
rename from media/libmedia/include/media/ExtendedAudioBufferProvider.h
rename to media/libaudioclient/include/media/ExtendedAudioBufferProvider.h
diff --git a/media/libaudioclient/tests/Android.bp b/media/libaudioclient/tests/Android.bp
index 52bb2fb..d509be6 100644
--- a/media/libaudioclient/tests/Android.bp
+++ b/media/libaudioclient/tests/Android.bp
@@ -11,6 +11,9 @@
defaults: ["libaudioclient_tests_defaults"],
srcs: ["test_create_audiotrack.cpp",
"test_create_utils.cpp"],
+ header_libs: [
+ "libmedia_headers",
+ ],
shared_libs: [
"libaudioclient",
"libbinder",
@@ -25,6 +28,9 @@
defaults: ["libaudioclient_tests_defaults"],
srcs: ["test_create_audiorecord.cpp",
"test_create_utils.cpp"],
+ header_libs: [
+ "libmedia_headers",
+ ],
shared_libs: [
"libaudioclient",
"libbinder",
diff --git a/media/libaudiohal/Android.bp b/media/libaudiohal/Android.bp
index 584c2c0..74b48f3 100644
--- a/media/libaudiohal/Android.bp
+++ b/media/libaudiohal/Android.bp
@@ -13,20 +13,16 @@
],
shared_libs: [
- "android.hardware.audio.effect@2.0",
- "android.hardware.audio.effect@4.0",
- "android.hardware.audio.effect@5.0",
- "android.hardware.audio@2.0",
- "android.hardware.audio@4.0",
- "android.hardware.audio@5.0",
"libaudiohal@2.0",
"libaudiohal@4.0",
"libaudiohal@5.0",
+ "libaudiohal@6.0",
"libutils",
],
header_libs: [
- "libaudiohal_headers"
+ "libaudiohal_headers",
+ "libbase_headers",
]
}
@@ -57,4 +53,10 @@
name: "libaudiohal_headers",
export_include_dirs: ["include"],
+
+ // This is needed because the stream interface includes media/MicrophoneInfo.h
+ // which is not in any library but has a dependency on headers from libbinder.
+ header_libs: ["libbinder_headers"],
+
+ export_header_lib_headers: ["libbinder_headers"],
}
diff --git a/media/libaudiohal/DevicesFactoryHalInterface.cpp b/media/libaudiohal/DevicesFactoryHalInterface.cpp
index f86009c..d5336fa 100644
--- a/media/libaudiohal/DevicesFactoryHalInterface.cpp
+++ b/media/libaudiohal/DevicesFactoryHalInterface.cpp
@@ -14,26 +14,16 @@
* limitations under the License.
*/
-#include <android/hardware/audio/2.0/IDevicesFactory.h>
-#include <android/hardware/audio/4.0/IDevicesFactory.h>
-#include <android/hardware/audio/5.0/IDevicesFactory.h>
-
#include <libaudiohal/FactoryHalHidl.h>
+#include <media/audiohal/DevicesFactoryHalInterface.h>
+
namespace android {
// static
sp<DevicesFactoryHalInterface> DevicesFactoryHalInterface::create() {
- if (hardware::audio::V5_0::IDevicesFactory::getService() != nullptr) {
- return V5_0::createDevicesFactoryHal();
- }
- if (hardware::audio::V4_0::IDevicesFactory::getService() != nullptr) {
- return V4_0::createDevicesFactoryHal();
- }
- if (hardware::audio::V2_0::IDevicesFactory::getService() != nullptr) {
- return V2_0::createDevicesFactoryHal();
- }
- return nullptr;
+ return createPreferedImpl<DevicesFactoryHalInterface>();
}
} // namespace android
+
diff --git a/media/libaudiohal/EffectsFactoryHalInterface.cpp b/media/libaudiohal/EffectsFactoryHalInterface.cpp
index bd3ef61..d15b14e 100644
--- a/media/libaudiohal/EffectsFactoryHalInterface.cpp
+++ b/media/libaudiohal/EffectsFactoryHalInterface.cpp
@@ -1,5 +1,5 @@
/*
- * Copyright (C) 2016 The Android Open Source Project
+ * Copyright (C) 2017 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
@@ -14,26 +14,15 @@
* limitations under the License.
*/
-#include <android/hardware/audio/effect/2.0/IEffectsFactory.h>
-#include <android/hardware/audio/effect/4.0/IEffectsFactory.h>
-#include <android/hardware/audio/effect/5.0/IEffectsFactory.h>
-
#include <libaudiohal/FactoryHalHidl.h>
+#include <media/audiohal/EffectsFactoryHalInterface.h>
+
namespace android {
// static
sp<EffectsFactoryHalInterface> EffectsFactoryHalInterface::create() {
- if (hardware::audio::effect::V5_0::IEffectsFactory::getService() != nullptr) {
- return effect::V5_0::createEffectsFactoryHal();
- }
- if (hardware::audio::effect::V4_0::IEffectsFactory::getService() != nullptr) {
- return effect::V4_0::createEffectsFactoryHal();
- }
- if (hardware::audio::effect::V2_0::IEffectsFactory::getService() != nullptr) {
- return effect::V2_0::createEffectsFactoryHal();
- }
- return nullptr;
+ return createPreferedImpl<EffectsFactoryHalInterface>();
}
// static
diff --git a/media/libaudiohal/impl/Android.bp b/media/libaudiohal/impl/Android.bp
index 88533da..8669e2a 100644
--- a/media/libaudiohal/impl/Android.bp
+++ b/media/libaudiohal/impl/Android.bp
@@ -36,8 +36,6 @@
"libhardware",
"libhidlbase",
"libhidlmemory",
- "libhidltransport",
- "libhwbinder",
"liblog",
"libmedia_helper",
"libmediautils",
@@ -45,6 +43,7 @@
],
header_libs: [
"android.hardware.audio.common.util@all-versions",
+ "libaudioclient_headers",
"libaudiohal_headers"
],
@@ -100,3 +99,20 @@
"-include common/all-versions/VersionMacro.h",
]
}
+
+cc_library_shared {
+ name: "libaudiohal@6.0",
+ defaults: ["libaudiohal_default"],
+ shared_libs: [
+ "android.hardware.audio.common@6.0",
+ "android.hardware.audio.common@6.0-util",
+ "android.hardware.audio.effect@6.0",
+ "android.hardware.audio@6.0",
+ ],
+ cflags: [
+ "-DMAJOR_VERSION=6",
+ "-DMINOR_VERSION=0",
+ "-include common/all-versions/VersionMacro.h",
+ ]
+}
+
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp b/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp
index 5e01e42..1335a0c 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/DevicesFactoryHalHidl.cpp
@@ -35,13 +35,10 @@
namespace android {
namespace CPP_VERSION {
-DevicesFactoryHalHidl::DevicesFactoryHalHidl() {
- sp<IDevicesFactory> defaultFactory{IDevicesFactory::getService()};
- if (!defaultFactory) {
- ALOGE("Failed to obtain IDevicesFactory/default service, terminating process.");
- exit(1);
- }
- mDeviceFactories.push_back(defaultFactory);
+DevicesFactoryHalHidl::DevicesFactoryHalHidl(sp<IDevicesFactory> devicesFactory) {
+ ALOG_ASSERT(devicesFactory != nullptr, "Provided IDevicesFactory service is NULL");
+
+ mDeviceFactories.push_back(devicesFactory);
if (MAJOR_VERSION >= 4) {
// The MSD factory is optional and only available starting at HAL 4.0
sp<IDevicesFactory> msdFactory{IDevicesFactory::getService(AUDIO_HAL_SERVICE_NAME_MSD)};
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHidl.h b/media/libaudiohal/impl/DevicesFactoryHalHidl.h
index 27e0649..8775e7b 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHidl.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalHidl.h
@@ -32,18 +32,14 @@
class DevicesFactoryHalHidl : public DevicesFactoryHalInterface
{
public:
+ DevicesFactoryHalHidl(sp<IDevicesFactory> devicesFactory);
+
// Opens a device with the specified name. To close the device, it is
// necessary to release references to the returned object.
virtual status_t openDevice(const char *name, sp<DeviceHalInterface> *device);
-
private:
- friend class DevicesFactoryHalHybrid;
-
std::vector<sp<IDevicesFactory>> mDeviceFactories;
- // Can not be constructed directly by clients.
- DevicesFactoryHalHidl();
-
virtual ~DevicesFactoryHalHidl() = default;
};
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp b/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
index f337a8b..0e1f1bb 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
+++ b/media/libaudiohal/impl/DevicesFactoryHalHybrid.cpp
@@ -17,16 +17,17 @@
#define LOG_TAG "DevicesFactoryHalHybrid"
//#define LOG_NDEBUG 0
+#include "DevicesFactoryHalHidl.h"
#include "DevicesFactoryHalHybrid.h"
#include "DevicesFactoryHalLocal.h"
-#include "DevicesFactoryHalHidl.h"
+#include <libaudiohal/FactoryHalHidl.h>
namespace android {
namespace CPP_VERSION {
-DevicesFactoryHalHybrid::DevicesFactoryHalHybrid()
+DevicesFactoryHalHybrid::DevicesFactoryHalHybrid(sp<IDevicesFactory> hidlFactory)
: mLocalFactory(new DevicesFactoryHalLocal()),
- mHidlFactory(new DevicesFactoryHalHidl()) {
+ mHidlFactory(new DevicesFactoryHalHidl(hidlFactory)) {
}
status_t DevicesFactoryHalHybrid::openDevice(const char *name, sp<DeviceHalInterface> *device) {
@@ -36,6 +37,12 @@
}
return mLocalFactory->openDevice(name, device);
}
-
} // namespace CPP_VERSION
+
+template <>
+sp<DevicesFactoryHalInterface> createFactoryHal<AudioHALVersion::CPP_VERSION>() {
+ auto service = hardware::audio::CPP_VERSION::IDevicesFactory::getService();
+ return service ? new CPP_VERSION::DevicesFactoryHalHybrid(service) : nullptr;
+}
+
} // namespace android
diff --git a/media/libaudiohal/impl/DevicesFactoryHalHybrid.h b/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
index 5ac0d0d..545bb70 100644
--- a/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
+++ b/media/libaudiohal/impl/DevicesFactoryHalHybrid.h
@@ -17,17 +17,20 @@
#ifndef ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_H
#define ANDROID_HARDWARE_DEVICES_FACTORY_HAL_HYBRID_H
+#include PATH(android/hardware/audio/FILE_VERSION/IDevicesFactory.h)
#include <media/audiohal/DevicesFactoryHalInterface.h>
#include <utils/Errors.h>
#include <utils/RefBase.h>
+using ::android::hardware::audio::CPP_VERSION::IDevicesFactory;
+
namespace android {
namespace CPP_VERSION {
class DevicesFactoryHalHybrid : public DevicesFactoryHalInterface
{
public:
- DevicesFactoryHalHybrid();
+ DevicesFactoryHalHybrid(sp<IDevicesFactory> hidlFactory);
// Opens a device with the specified name. To close the device, it is
// necessary to release references to the returned object.
@@ -38,10 +41,6 @@
sp<DevicesFactoryHalInterface> mHidlFactory;
};
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal() {
- return new DevicesFactoryHalHybrid();
-}
-
} // namespace CPP_VERSION
} // namespace android
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
index 7fd6bde..ba7b195 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.cpp
@@ -19,11 +19,12 @@
#include <cutils/native_handle.h>
-#include "EffectsFactoryHalHidl.h"
#include "ConversionHelperHidl.h"
#include "EffectBufferHalHidl.h"
#include "EffectHalHidl.h"
+#include "EffectsFactoryHalHidl.h"
#include "HidlUtils.h"
+#include <libaudiohal/FactoryHalHidl.h>
using ::android::hardware::audio::common::CPP_VERSION::implementation::HidlUtils;
using ::android::hardware::Return;
@@ -35,12 +36,10 @@
using namespace ::android::hardware::audio::common::CPP_VERSION;
using namespace ::android::hardware::audio::effect::CPP_VERSION;
-EffectsFactoryHalHidl::EffectsFactoryHalHidl() : ConversionHelperHidl("EffectsFactory") {
- mEffectsFactory = IEffectsFactory::getService();
- if (mEffectsFactory == 0) {
- ALOGE("Failed to obtain IEffectsFactory service, terminating process.");
- exit(1);
- }
+EffectsFactoryHalHidl::EffectsFactoryHalHidl(sp<IEffectsFactory> effectsFactory)
+ : ConversionHelperHidl("EffectsFactory") {
+ ALOG_ASSERT(effectsFactory != nullptr, "Provided IDevicesFactory service is NULL");
+ mEffectsFactory = effectsFactory;
}
status_t EffectsFactoryHalHidl::queryAllDescriptors() {
@@ -147,4 +146,11 @@
} // namespace CPP_VERSION
} // namespace effect
+
+template<>
+sp<EffectsFactoryHalInterface> createFactoryHal<AudioHALVersion::CPP_VERSION>() {
+ auto service = hardware::audio::effect::CPP_VERSION::IEffectsFactory::getService();
+ return service ? new effect::CPP_VERSION::EffectsFactoryHalHidl(service) : nullptr;
+}
+
} // namespace android
diff --git a/media/libaudiohal/impl/EffectsFactoryHalHidl.h b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
index 01178ff..2828513 100644
--- a/media/libaudiohal/impl/EffectsFactoryHalHidl.h
+++ b/media/libaudiohal/impl/EffectsFactoryHalHidl.h
@@ -18,7 +18,6 @@
#define ANDROID_HARDWARE_EFFECTS_FACTORY_HAL_HIDL_H
#include PATH(android/hardware/audio/effect/FILE_VERSION/IEffectsFactory.h)
-#include PATH(android/hardware/audio/effect/FILE_VERSION/types.h)
#include <media/audiohal/EffectsFactoryHalInterface.h>
#include "ConversionHelperHidl.h"
@@ -34,7 +33,7 @@
class EffectsFactoryHalHidl : public EffectsFactoryHalInterface, public ConversionHelperHidl
{
public:
- EffectsFactoryHalHidl();
+ EffectsFactoryHalHidl(sp<IEffectsFactory> effectsFactory);
// Returns the number of different effects in all loaded libraries.
virtual status_t queryNumberEffects(uint32_t *pNumEffects);
@@ -66,10 +65,6 @@
status_t queryAllDescriptors();
};
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal() {
- return new EffectsFactoryHalHidl();
-}
-
} // namespace CPP_VERSION
} // namespace effect
} // namespace android
diff --git a/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h b/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
index c7319d0..271bafc 100644
--- a/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
+++ b/media/libaudiohal/impl/include/libaudiohal/FactoryHalHidl.h
@@ -23,33 +23,43 @@
#include <media/audiohal/EffectsFactoryHalInterface.h>
#include <utils/StrongPointer.h>
+#include <array>
+#include <utility>
+
namespace android {
-namespace effect {
-namespace V2_0 {
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
-} // namespace V2_0
+/** Supported HAL versions, in order of preference.
+ * Implementation should use specialize the `create*FactoryHal` for their version.
+ * Client should use `createPreferedImpl<*FactoryHal>()` to instantiate
+ * the preferred available impl.
+ */
+enum class AudioHALVersion {
+ V6_0,
+ V5_0,
+ V4_0,
+ V2_0,
+ end, // used for iterating over supported versions
+};
-namespace V4_0 {
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
-} // namespace V4_0
+/** Template function to fully specialized for each version and each Interface. */
+template <AudioHALVersion, class Interface>
+sp<Interface> createFactoryHal();
-namespace V5_0 {
-sp<EffectsFactoryHalInterface> createEffectsFactoryHal();
-} // namespace V5_0
-} // namespace effect
+/** @Return the preferred available implementation or nullptr if none are available. */
+template <class Interface, AudioHALVersion version = AudioHALVersion{}>
+static sp<Interface> createPreferedImpl() {
+ if constexpr (version == AudioHALVersion::end) {
+ return nullptr; // tried all version, all returned nullptr
+ } else {
+ if (auto created = createFactoryHal<version, Interface>(); created != nullptr) {
+ return created;
+ }
-namespace V2_0 {
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
-} // namespace V2_0
+ using Raw = std::underlying_type_t<AudioHALVersion>; // cast as enum class do not support ++
+ return createPreferedImpl<Interface, AudioHALVersion(Raw(version) + 1)>();
+ }
+}
-namespace V4_0 {
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
-} // namespace V4_0
-
-namespace V5_0 {
-sp<DevicesFactoryHalInterface> createDevicesFactoryHal();
-} // namespace V5_0
} // namespace android
diff --git a/media/libaudioprocessing/Android.bp b/media/libaudioprocessing/Android.bp
index cb78063..9b5d58c 100644
--- a/media/libaudioprocessing/Android.bp
+++ b/media/libaudioprocessing/Android.bp
@@ -3,20 +3,13 @@
export_include_dirs: ["include"],
+ header_libs: ["libaudioclient_headers"],
+
shared_libs: [
- "libaudiohal",
"libaudioutils",
"libcutils",
"liblog",
- "libnbaio",
- "libnblog",
- "libsonic",
"libutils",
- "libvibrator",
- ],
-
- header_libs: [
- "libbase_headers",
],
cflags: [
@@ -33,18 +26,32 @@
defaults: ["libaudioprocessing_defaults"],
srcs: [
+ "AudioMixer.cpp",
"BufferProviders.cpp",
"RecordBufferConverter.cpp",
],
- whole_static_libs: ["libaudioprocessing_arm"],
+
+ header_libs: [
+ "libbase_headers",
+ "libmedia_headers"
+ ],
+
+ shared_libs: [
+ "libaudiohal",
+ "libsonic",
+ "libvibrator",
+ ],
+
+ whole_static_libs: ["libaudioprocessing_base"],
}
cc_library_static {
- name: "libaudioprocessing_arm",
+ name: "libaudioprocessing_base",
defaults: ["libaudioprocessing_defaults"],
+ vendor_available: true,
srcs: [
- "AudioMixer.cpp",
+ "AudioMixerBase.cpp",
"AudioResampler.cpp",
"AudioResamplerCubic.cpp",
"AudioResamplerSinc.cpp",
diff --git a/media/libaudioprocessing/AudioMixer.cpp b/media/libaudioprocessing/AudioMixer.cpp
index f7cc096..c0b11a4 100644
--- a/media/libaudioprocessing/AudioMixer.cpp
+++ b/media/libaudioprocessing/AudioMixer.cpp
@@ -18,6 +18,7 @@
#define LOG_TAG "AudioMixer"
//#define LOG_NDEBUG 0
+#include <sstream>
#include <stdint.h>
#include <string.h>
#include <stdlib.h>
@@ -27,9 +28,6 @@
#include <utils/Errors.h>
#include <utils/Log.h>
-#include <cutils/compiler.h>
-#include <utils/Debug.h>
-
#include <system/audio.h>
#include <audio_utils/primitives.h>
@@ -58,138 +56,15 @@
#define ALOGVV(a...) do { } while (0)
#endif
-#ifndef ARRAY_SIZE
-#define ARRAY_SIZE(x) (sizeof(x)/sizeof((x)[0]))
-#endif
-
-// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
-// original code will be used for stereo sinks, the new mixer for multichannel.
-static constexpr bool kUseNewMixer = true;
-
-// Set kUseFloat to true to allow floating input into the mixer engine.
-// If kUseNewMixer is false, this is ignored or may be overridden internally
-// because of downmix/upmix support.
-static constexpr bool kUseFloat = true;
-
-#ifdef FLOAT_AUX
-using TYPE_AUX = float;
-static_assert(kUseNewMixer && kUseFloat,
- "kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
-#else
-using TYPE_AUX = int32_t; // q4.27
-#endif
-
// Set to default copy buffer size in frames for input processing.
-static const size_t kCopyBufferFrameCount = 256;
+static constexpr size_t kCopyBufferFrameCount = 256;
namespace android {
// ----------------------------------------------------------------------------
-static inline audio_format_t selectMixerInFormat(audio_format_t inputFormat __unused) {
- return kUseFloat && kUseNewMixer ? AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
-}
-
-status_t AudioMixer::create(
- int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
-{
- LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);
-
- if (!isValidChannelMask(channelMask)) {
- ALOGE("%s invalid channelMask: %#x", __func__, channelMask);
- return BAD_VALUE;
- }
- if (!isValidFormat(format)) {
- ALOGE("%s invalid format: %#x", __func__, format);
- return BAD_VALUE;
- }
-
- auto t = std::make_shared<Track>();
- {
- // TODO: move initialization to the Track constructor.
- // assume default parameters for the track, except where noted below
- t->needs = 0;
-
- // Integer volume.
- // Currently integer volume is kept for the legacy integer mixer.
- // Will be removed when the legacy mixer path is removed.
- t->volume[0] = 0;
- t->volume[1] = 0;
- t->prevVolume[0] = 0 << 16;
- t->prevVolume[1] = 0 << 16;
- t->volumeInc[0] = 0;
- t->volumeInc[1] = 0;
- t->auxLevel = 0;
- t->auxInc = 0;
- t->prevAuxLevel = 0;
-
- // Floating point volume.
- t->mVolume[0] = 0.f;
- t->mVolume[1] = 0.f;
- t->mPrevVolume[0] = 0.f;
- t->mPrevVolume[1] = 0.f;
- t->mVolumeInc[0] = 0.;
- t->mVolumeInc[1] = 0.;
- t->mAuxLevel = 0.;
- t->mAuxInc = 0.;
- t->mPrevAuxLevel = 0.;
-
- // no initialization needed
- // t->frameCount
- t->mHapticChannelMask = channelMask & AUDIO_CHANNEL_HAPTIC_ALL;
- t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
- channelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
- t->channelCount = audio_channel_count_from_out_mask(channelMask);
- t->enabled = false;
- ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
- "Non-stereo channel mask: %d\n", channelMask);
- t->channelMask = channelMask;
- t->sessionId = sessionId;
- // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
- t->bufferProvider = NULL;
- t->buffer.raw = NULL;
- // no initialization needed
- // t->buffer.frameCount
- t->hook = NULL;
- t->mIn = NULL;
- t->sampleRate = mSampleRate;
- // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
- t->mainBuffer = NULL;
- t->auxBuffer = NULL;
- t->mInputBufferProvider = NULL;
- t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
- t->mFormat = format;
- t->mMixerInFormat = selectMixerInFormat(format);
- t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
- t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
- AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
- t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
- t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
- // haptic
- t->mHapticPlaybackEnabled = false;
- t->mHapticIntensity = HAPTIC_SCALE_NONE;
- t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
- t->mMixerHapticChannelCount = 0;
- t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
- t->mAdjustOutChannelCount = t->channelCount + t->mMixerHapticChannelCount;
- t->mAdjustNonDestructiveInChannelCount = t->mAdjustOutChannelCount;
- t->mAdjustNonDestructiveOutChannelCount = t->channelCount;
- t->mKeepContractedChannels = false;
- // Check the downmixing (or upmixing) requirements.
- status_t status = t->prepareForDownmix();
- if (status != OK) {
- ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
- return BAD_VALUE;
- }
- // prepareForDownmix() may change mDownmixRequiresFormat
- ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
- t->prepareForReformat();
- t->prepareForAdjustChannelsNonDestructive(mFrameCount);
- t->prepareForAdjustChannels();
-
- mTracks[name] = t;
- return OK;
- }
+bool AudioMixer::isValidChannelMask(audio_channel_mask_t channelMask) const {
+ return audio_channel_mask_is_valid(channelMask); // the RemixBufferProvider is flexible.
}
// Called when channel masks have changed for a track name
@@ -198,7 +73,7 @@
bool AudioMixer::setChannelMasks(int name,
audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask) {
LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
+ const std::shared_ptr<Track> &track = getTrack(name);
if (trackChannelMask == (track->channelMask | track->mHapticChannelMask)
&& mixerChannelMask == (track->mMixerChannelMask | track->mMixerHapticChannelMask)) {
@@ -255,14 +130,8 @@
track->prepareForAdjustChannelsNonDestructive(mFrameCount);
track->prepareForAdjustChannels();
- if (track->mResampler.get() != nullptr) {
- // resampler channels may have changed.
- const uint32_t resetToSampleRate = track->sampleRate;
- track->mResampler.reset(nullptr);
- track->sampleRate = mSampleRate; // without resampler, track rate is device sample rate.
- // recreate the resampler with updated format, channels, saved sampleRate.
- track->setResampler(resetToSampleRate /*trackSampleRate*/, mSampleRate /*devSampleRate*/);
- }
+ // Resampler channels may have changed.
+ track->recreateResampler(mSampleRate);
return true;
}
@@ -477,171 +346,10 @@
}
}
-void AudioMixer::destroy(int name)
-{
- LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- ALOGV("deleteTrackName(%d)", name);
-
- if (mTracks[name]->enabled) {
- invalidate();
- }
- mTracks.erase(name); // deallocate track
-}
-
-void AudioMixer::enable(int name)
-{
- LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
-
- if (!track->enabled) {
- track->enabled = true;
- ALOGV("enable(%d)", name);
- invalidate();
- }
-}
-
-void AudioMixer::disable(int name)
-{
- LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
-
- if (track->enabled) {
- track->enabled = false;
- ALOGV("disable(%d)", name);
- invalidate();
- }
-}
-
-/* Sets the volume ramp variables for the AudioMixer.
- *
- * The volume ramp variables are used to transition from the previous
- * volume to the set volume. ramp controls the duration of the transition.
- * Its value is typically one state framecount period, but may also be 0,
- * meaning "immediate."
- *
- * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
- * even if there is a nonzero floating point increment (in that case, the volume
- * change is immediate). This restriction should be changed when the legacy mixer
- * is removed (see #2).
- * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
- * when no longer needed.
- *
- * @param newVolume set volume target in floating point [0.0, 1.0].
- * @param ramp number of frames to increment over. if ramp is 0, the volume
- * should be set immediately. Currently ramp should not exceed 65535 (frames).
- * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
- * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
- * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
- * @param pSetVolume pointer to the float target volume, set on return.
- * @param pPrevVolume pointer to the float previous volume, set on return.
- * @param pVolumeInc pointer to the float increment per output audio frame, set on return.
- * @return true if the volume has changed, false if volume is same.
- */
-static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
- int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
- float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
- // check floating point volume to see if it is identical to the previously
- // set volume.
- // We do not use a tolerance here (and reject changes too small)
- // as it may be confusing to use a different value than the one set.
- // If the resulting volume is too small to ramp, it is a direct set of the volume.
- if (newVolume == *pSetVolume) {
- return false;
- }
- if (newVolume < 0) {
- newVolume = 0; // should not have negative volumes
- } else {
- switch (fpclassify(newVolume)) {
- case FP_SUBNORMAL:
- case FP_NAN:
- newVolume = 0;
- break;
- case FP_ZERO:
- break; // zero volume is fine
- case FP_INFINITE:
- // Infinite volume could be handled consistently since
- // floating point math saturates at infinities,
- // but we limit volume to unity gain float.
- // ramp = 0; break;
- //
- newVolume = AudioMixer::UNITY_GAIN_FLOAT;
- break;
- case FP_NORMAL:
- default:
- // Floating point does not have problems with overflow wrap
- // that integer has. However, we limit the volume to
- // unity gain here.
- // TODO: Revisit the volume limitation and perhaps parameterize.
- if (newVolume > AudioMixer::UNITY_GAIN_FLOAT) {
- newVolume = AudioMixer::UNITY_GAIN_FLOAT;
- }
- break;
- }
- }
-
- // set floating point volume ramp
- if (ramp != 0) {
- // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
- // is no computational mismatch; hence equality is checked here.
- ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
- " prev:%f set_to:%f", *pPrevVolume, *pSetVolume);
- const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
- // could be inf, cannot be nan, subnormal
- const float maxv = std::max(newVolume, *pPrevVolume);
-
- if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
- && maxv + inc != maxv) { // inc must make forward progress
- *pVolumeInc = inc;
- // ramp is set now.
- // Note: if newVolume is 0, then near the end of the ramp,
- // it may be possible that the ramped volume may be subnormal or
- // temporarily negative by a small amount or subnormal due to floating
- // point inaccuracies.
- } else {
- ramp = 0; // ramp not allowed
- }
- }
-
- // compute and check integer volume, no need to check negative values
- // The integer volume is limited to "unity_gain" to avoid wrapping and other
- // audio artifacts, so it never reaches the range limit of U4.28.
- // We safely use signed 16 and 32 bit integers here.
- const float scaledVolume = newVolume * AudioMixer::UNITY_GAIN_INT; // not neg, subnormal, nan
- const int32_t intVolume = (scaledVolume >= (float)AudioMixer::UNITY_GAIN_INT) ?
- AudioMixer::UNITY_GAIN_INT : (int32_t)scaledVolume;
-
- // set integer volume ramp
- if (ramp != 0) {
- // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
- // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
- // is no computational mismatch; hence equality is checked here.
- ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
- " prev:%d set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
- const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
-
- if (inc != 0) { // inc must make forward progress
- *pIntVolumeInc = inc;
- } else {
- ramp = 0; // ramp not allowed
- }
- }
-
- // if no ramp, or ramp not allowed, then clear float and integer increments
- if (ramp == 0) {
- *pVolumeInc = 0;
- *pPrevVolume = newVolume;
- *pIntVolumeInc = 0;
- *pIntPrevVolume = intVolume << 16;
- }
- *pSetVolume = newVolume;
- *pIntSetVolume = intVolume;
- return true;
-}
-
void AudioMixer::setParameter(int name, int target, int param, void *value)
{
LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
+ const std::shared_ptr<Track> &track = getTrack(name);
int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
@@ -670,11 +378,7 @@
}
break;
case AUX_BUFFER:
- if (track->auxBuffer != valueBuf) {
- track->auxBuffer = valueBuf;
- ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
- invalidate();
- }
+ AudioMixerBase::setParameter(name, target, param, value);
break;
case FORMAT: {
audio_format_t format = static_cast<audio_format_t>(valueInt);
@@ -730,127 +434,38 @@
break;
case RESAMPLE:
- switch (param) {
- case SAMPLE_RATE:
- ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
- if (track->setResampler(uint32_t(valueInt), mSampleRate)) {
- ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
- uint32_t(valueInt));
- invalidate();
- }
- break;
- case RESET:
- track->resetResampler();
- invalidate();
- break;
- case REMOVE:
- track->mResampler.reset(nullptr);
- track->sampleRate = mSampleRate;
- invalidate();
- break;
- default:
- LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
- }
- break;
-
case RAMP_VOLUME:
case VOLUME:
+ AudioMixerBase::setParameter(name, target, param, value);
+ break;
+ case TIMESTRETCH:
switch (param) {
- case AUXLEVEL:
- if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
- target == RAMP_VOLUME ? mFrameCount : 0,
- &track->auxLevel, &track->prevAuxLevel, &track->auxInc,
- &track->mAuxLevel, &track->mPrevAuxLevel, &track->mAuxInc)) {
- ALOGV("setParameter(%s, AUXLEVEL: %04x)",
- target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track->auxLevel);
- invalidate();
+ case PLAYBACK_RATE: {
+ const AudioPlaybackRate *playbackRate =
+ reinterpret_cast<AudioPlaybackRate*>(value);
+ ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
+ "bad parameters speed %f, pitch %f",
+ playbackRate->mSpeed, playbackRate->mPitch);
+ if (track->setPlaybackRate(*playbackRate)) {
+ ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
+ "%f %f %d %d",
+ playbackRate->mSpeed,
+ playbackRate->mPitch,
+ playbackRate->mStretchMode,
+ playbackRate->mFallbackMode);
+ // invalidate(); (should not require reconfigure)
}
- break;
+ } break;
default:
- if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
- if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
- target == RAMP_VOLUME ? mFrameCount : 0,
- &track->volume[param - VOLUME0],
- &track->prevVolume[param - VOLUME0],
- &track->volumeInc[param - VOLUME0],
- &track->mVolume[param - VOLUME0],
- &track->mPrevVolume[param - VOLUME0],
- &track->mVolumeInc[param - VOLUME0])) {
- ALOGV("setParameter(%s, VOLUME%d: %04x)",
- target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
- track->volume[param - VOLUME0]);
- invalidate();
- }
- } else {
- LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
- }
+ LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
}
break;
- case TIMESTRETCH:
- switch (param) {
- case PLAYBACK_RATE: {
- const AudioPlaybackRate *playbackRate =
- reinterpret_cast<AudioPlaybackRate*>(value);
- ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
- "bad parameters speed %f, pitch %f",
- playbackRate->mSpeed, playbackRate->mPitch);
- if (track->setPlaybackRate(*playbackRate)) {
- ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
- "%f %f %d %d",
- playbackRate->mSpeed,
- playbackRate->mPitch,
- playbackRate->mStretchMode,
- playbackRate->mFallbackMode);
- // invalidate(); (should not require reconfigure)
- }
- } break;
- default:
- LOG_ALWAYS_FATAL("setParameter timestretch: bad param %d", param);
- }
- break;
default:
LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
}
}
-bool AudioMixer::Track::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
-{
- if (trackSampleRate != devSampleRate || mResampler.get() != nullptr) {
- if (sampleRate != trackSampleRate) {
- sampleRate = trackSampleRate;
- if (mResampler.get() == nullptr) {
- ALOGV("Creating resampler from track %d Hz to device %d Hz",
- trackSampleRate, devSampleRate);
- AudioResampler::src_quality quality;
- // force lowest quality level resampler if use case isn't music or video
- // FIXME this is flawed for dynamic sample rates, as we choose the resampler
- // quality level based on the initial ratio, but that could change later.
- // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
- if (isMusicRate(trackSampleRate)) {
- quality = AudioResampler::DEFAULT_QUALITY;
- } else {
- quality = AudioResampler::DYN_LOW_QUALITY;
- }
-
- // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
- // but if none exists, it is the channel count (1 for mono).
- const int resamplerChannelCount = mDownmixerBufferProvider.get() != nullptr
- ? mMixerChannelCount : channelCount;
- ALOGVV("Creating resampler:"
- " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
- mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
- mResampler.reset(AudioResampler::create(
- mMixerInFormat,
- resamplerChannelCount,
- devSampleRate, quality));
- }
- return true;
- }
- }
- return false;
-}
-
bool AudioMixer::Track::setPlaybackRate(const AudioPlaybackRate &playbackRate)
{
if ((mTimestretchBufferProvider.get() == nullptr &&
@@ -863,8 +478,7 @@
if (mTimestretchBufferProvider.get() == nullptr) {
// TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
// but if none exists, it is the channel count (1 for mono).
- const int timestretchChannelCount = mDownmixerBufferProvider.get() != nullptr
- ? mMixerChannelCount : channelCount;
+ const int timestretchChannelCount = getOutputChannelCount();
mTimestretchBufferProvider.reset(new TimestretchBufferProvider(timestretchChannelCount,
mMixerInFormat, sampleRate, playbackRate));
reconfigureBufferProviders();
@@ -875,84 +489,10 @@
return true;
}
-/* Checks to see if the volume ramp has completed and clears the increment
- * variables appropriately.
- *
- * FIXME: There is code to handle int/float ramp variable switchover should it not
- * complete within a mixer buffer processing call, but it is preferred to avoid switchover
- * due to precision issues. The switchover code is included for legacy code purposes
- * and can be removed once the integer volume is removed.
- *
- * It is not sufficient to clear only the volumeInc integer variable because
- * if one channel requires ramping, all channels are ramped.
- *
- * There is a bit of duplicated code here, but it keeps backward compatibility.
- */
-inline void AudioMixer::Track::adjustVolumeRamp(bool aux, bool useFloat)
-{
- if (useFloat) {
- for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
- if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) ||
- (mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) {
- volumeInc[i] = 0;
- prevVolume[i] = volume[i] << 16;
- mVolumeInc[i] = 0.;
- mPrevVolume[i] = mVolume[i];
- } else {
- //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
- prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
- }
- }
- } else {
- for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
- if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
- ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
- volumeInc[i] = 0;
- prevVolume[i] = volume[i] << 16;
- mVolumeInc[i] = 0.;
- mPrevVolume[i] = mVolume[i];
- } else {
- //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
- mPrevVolume[i] = float_from_u4_28(prevVolume[i]);
- }
- }
- }
-
- if (aux) {
-#ifdef FLOAT_AUX
- if (useFloat) {
- if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||
- (mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
- auxInc = 0;
- prevAuxLevel = auxLevel << 16;
- mAuxInc = 0.f;
- mPrevAuxLevel = mAuxLevel;
- }
- } else
-#endif
- if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||
- (auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
- auxInc = 0;
- prevAuxLevel = auxLevel << 16;
- mAuxInc = 0.f;
- mPrevAuxLevel = mAuxLevel;
- }
- }
-}
-
-size_t AudioMixer::getUnreleasedFrames(int name) const
-{
- const auto it = mTracks.find(name);
- if (it != mTracks.end()) {
- return it->second->getUnreleasedFrames();
- }
- return 0;
-}
-
void AudioMixer::setBufferProvider(int name, AudioBufferProvider* bufferProvider)
{
LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
- const std::shared_ptr<Track> &track = mTracks[name];
+ const std::shared_ptr<Track> &track = getTrack(name);
if (track->mInputBufferProvider == bufferProvider) {
return; // don't reset any buffer providers if identical.
@@ -976,679 +516,6 @@
track->reconfigureBufferProviders();
}
-void AudioMixer::process__validate()
-{
- // TODO: fix all16BitsStereNoResample logic to
- // either properly handle muted tracks (it should ignore them)
- // or remove altogether as an obsolete optimization.
- bool all16BitsStereoNoResample = true;
- bool resampling = false;
- bool volumeRamp = false;
-
- mEnabled.clear();
- mGroups.clear();
- for (const auto &pair : mTracks) {
- const int name = pair.first;
- const std::shared_ptr<Track> &t = pair.second;
- if (!t->enabled) continue;
-
- mEnabled.emplace_back(name); // we add to mEnabled in order of name.
- mGroups[t->mainBuffer].emplace_back(name); // mGroups also in order of name.
-
- uint32_t n = 0;
- // FIXME can overflow (mask is only 3 bits)
- n |= NEEDS_CHANNEL_1 + t->channelCount - 1;
- if (t->doesResample()) {
- n |= NEEDS_RESAMPLE;
- }
- if (t->auxLevel != 0 && t->auxBuffer != NULL) {
- n |= NEEDS_AUX;
- }
-
- if (t->volumeInc[0]|t->volumeInc[1]) {
- volumeRamp = true;
- } else if (!t->doesResample() && t->volumeRL == 0) {
- n |= NEEDS_MUTE;
- }
- t->needs = n;
-
- if (n & NEEDS_MUTE) {
- t->hook = &Track::track__nop;
- } else {
- if (n & NEEDS_AUX) {
- all16BitsStereoNoResample = false;
- }
- if (n & NEEDS_RESAMPLE) {
- all16BitsStereoNoResample = false;
- resampling = true;
- t->hook = Track::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
- t->mMixerInFormat, t->mMixerFormat);
- ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
- "Track %d needs downmix + resample", name);
- } else {
- if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
- t->hook = Track::getTrackHook(
- (t->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO // TODO: MONO_HACK
- && t->channelMask == AUDIO_CHANNEL_OUT_MONO)
- ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
- t->mMixerChannelCount,
- t->mMixerInFormat, t->mMixerFormat);
- all16BitsStereoNoResample = false;
- }
- if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
- t->hook = Track::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
- t->mMixerInFormat, t->mMixerFormat);
- ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
- "Track %d needs downmix", name);
- }
- }
- }
- }
-
- // select the processing hooks
- mHook = &AudioMixer::process__nop;
- if (mEnabled.size() > 0) {
- if (resampling) {
- if (mOutputTemp.get() == nullptr) {
- mOutputTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
- }
- if (mResampleTemp.get() == nullptr) {
- mResampleTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
- }
- mHook = &AudioMixer::process__genericResampling;
- } else {
- // we keep temp arrays around.
- mHook = &AudioMixer::process__genericNoResampling;
- if (all16BitsStereoNoResample && !volumeRamp) {
- if (mEnabled.size() == 1) {
- const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
- if ((t->needs & NEEDS_MUTE) == 0) {
- // The check prevents a muted track from acquiring a process hook.
- //
- // This is dangerous if the track is MONO as that requires
- // special case handling due to implicit channel duplication.
- // Stereo or Multichannel should actually be fine here.
- mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
- t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
- }
- }
- }
- }
- }
-
- ALOGV("mixer configuration change: %zu "
- "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
- mEnabled.size(), all16BitsStereoNoResample, resampling, volumeRamp);
-
- process();
-
- // Now that the volume ramp has been done, set optimal state and
- // track hooks for subsequent mixer process
- if (mEnabled.size() > 0) {
- bool allMuted = true;
-
- for (const int name : mEnabled) {
- const std::shared_ptr<Track> &t = mTracks[name];
- if (!t->doesResample() && t->volumeRL == 0) {
- t->needs |= NEEDS_MUTE;
- t->hook = &Track::track__nop;
- } else {
- allMuted = false;
- }
- }
- if (allMuted) {
- mHook = &AudioMixer::process__nop;
- } else if (all16BitsStereoNoResample) {
- if (mEnabled.size() == 1) {
- //const int i = 31 - __builtin_clz(enabledTracks);
- const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
- // Muted single tracks handled by allMuted above.
- mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
- t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
- }
- }
- }
-}
-
-void AudioMixer::Track::track__genericResample(
- int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
-{
- ALOGVV("track__genericResample\n");
- mResampler->setSampleRate(sampleRate);
-
- // ramp gain - resample to temp buffer and scale/mix in 2nd step
- if (aux != NULL) {
- // always resample with unity gain when sending to auxiliary buffer to be able
- // to apply send level after resampling
- mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
- memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(int32_t));
- mResampler->resample(temp, outFrameCount, bufferProvider);
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
- volumeRampStereo(out, outFrameCount, temp, aux);
- } else {
- volumeStereo(out, outFrameCount, temp, aux);
- }
- } else {
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
- mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
- memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
- mResampler->resample(temp, outFrameCount, bufferProvider);
- volumeRampStereo(out, outFrameCount, temp, aux);
- }
-
- // constant gain
- else {
- mResampler->setVolume(mVolume[0], mVolume[1]);
- mResampler->resample(out, outFrameCount, bufferProvider);
- }
- }
-}
-
-void AudioMixer::Track::track__nop(int32_t* out __unused,
- size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
-{
-}
-
-void AudioMixer::Track::volumeRampStereo(
- int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
-{
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
-
- //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
-
- // ramp volume
- if (CC_UNLIKELY(aux != NULL)) {
- int32_t va = prevAuxLevel;
- const int32_t vaInc = auxInc;
- int32_t l;
- int32_t r;
-
- do {
- l = (*temp++ >> 12);
- r = (*temp++ >> 12);
- *out++ += (vl >> 16) * l;
- *out++ += (vr >> 16) * r;
- *aux++ += (va >> 17) * (l + r);
- vl += vlInc;
- vr += vrInc;
- va += vaInc;
- } while (--frameCount);
- prevAuxLevel = va;
- } else {
- do {
- *out++ += (vl >> 16) * (*temp++ >> 12);
- *out++ += (vr >> 16) * (*temp++ >> 12);
- vl += vlInc;
- vr += vrInc;
- } while (--frameCount);
- }
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- adjustVolumeRamp(aux != NULL);
-}
-
-void AudioMixer::Track::volumeStereo(
- int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
-{
- const int16_t vl = volume[0];
- const int16_t vr = volume[1];
-
- if (CC_UNLIKELY(aux != NULL)) {
- const int16_t va = auxLevel;
- do {
- int16_t l = (int16_t)(*temp++ >> 12);
- int16_t r = (int16_t)(*temp++ >> 12);
- out[0] = mulAdd(l, vl, out[0]);
- int16_t a = (int16_t)(((int32_t)l + r) >> 1);
- out[1] = mulAdd(r, vr, out[1]);
- out += 2;
- aux[0] = mulAdd(a, va, aux[0]);
- aux++;
- } while (--frameCount);
- } else {
- do {
- int16_t l = (int16_t)(*temp++ >> 12);
- int16_t r = (int16_t)(*temp++ >> 12);
- out[0] = mulAdd(l, vl, out[0]);
- out[1] = mulAdd(r, vr, out[1]);
- out += 2;
- } while (--frameCount);
- }
-}
-
-void AudioMixer::Track::track__16BitsStereo(
- int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
-{
- ALOGVV("track__16BitsStereo\n");
- const int16_t *in = static_cast<const int16_t *>(mIn);
-
- if (CC_UNLIKELY(aux != NULL)) {
- int32_t l;
- int32_t r;
- // ramp gain
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- int32_t va = prevAuxLevel;
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
- const int32_t vaInc = auxInc;
- // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
-
- do {
- l = (int32_t)*in++;
- r = (int32_t)*in++;
- *out++ += (vl >> 16) * l;
- *out++ += (vr >> 16) * r;
- *aux++ += (va >> 17) * (l + r);
- vl += vlInc;
- vr += vrInc;
- va += vaInc;
- } while (--frameCount);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- prevAuxLevel = va;
- adjustVolumeRamp(true);
- }
-
- // constant gain
- else {
- const uint32_t vrl = volumeRL;
- const int16_t va = (int16_t)auxLevel;
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
- in += 2;
- out[0] = mulAddRL(1, rl, vrl, out[0]);
- out[1] = mulAddRL(0, rl, vrl, out[1]);
- out += 2;
- aux[0] = mulAdd(a, va, aux[0]);
- aux++;
- } while (--frameCount);
- }
- } else {
- // ramp gain
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
-
- // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
-
- do {
- *out++ += (vl >> 16) * (int32_t) *in++;
- *out++ += (vr >> 16) * (int32_t) *in++;
- vl += vlInc;
- vr += vrInc;
- } while (--frameCount);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- adjustVolumeRamp(false);
- }
-
- // constant gain
- else {
- const uint32_t vrl = volumeRL;
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- out[0] = mulAddRL(1, rl, vrl, out[0]);
- out[1] = mulAddRL(0, rl, vrl, out[1]);
- out += 2;
- } while (--frameCount);
- }
- }
- mIn = in;
-}
-
-void AudioMixer::Track::track__16BitsMono(
- int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
-{
- ALOGVV("track__16BitsMono\n");
- const int16_t *in = static_cast<int16_t const *>(mIn);
-
- if (CC_UNLIKELY(aux != NULL)) {
- // ramp gain
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- int32_t va = prevAuxLevel;
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
- const int32_t vaInc = auxInc;
-
- // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
-
- do {
- int32_t l = *in++;
- *out++ += (vl >> 16) * l;
- *out++ += (vr >> 16) * l;
- *aux++ += (va >> 16) * l;
- vl += vlInc;
- vr += vrInc;
- va += vaInc;
- } while (--frameCount);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- prevAuxLevel = va;
- adjustVolumeRamp(true);
- }
- // constant gain
- else {
- const int16_t vl = volume[0];
- const int16_t vr = volume[1];
- const int16_t va = (int16_t)auxLevel;
- do {
- int16_t l = *in++;
- out[0] = mulAdd(l, vl, out[0]);
- out[1] = mulAdd(l, vr, out[1]);
- out += 2;
- aux[0] = mulAdd(l, va, aux[0]);
- aux++;
- } while (--frameCount);
- }
- } else {
- // ramp gain
- if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
- int32_t vl = prevVolume[0];
- int32_t vr = prevVolume[1];
- const int32_t vlInc = volumeInc[0];
- const int32_t vrInc = volumeInc[1];
-
- // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
- // t, vlInc/65536.0f, vl/65536.0f, volume[0],
- // (vl + vlInc*frameCount)/65536.0f, frameCount);
-
- do {
- int32_t l = *in++;
- *out++ += (vl >> 16) * l;
- *out++ += (vr >> 16) * l;
- vl += vlInc;
- vr += vrInc;
- } while (--frameCount);
-
- prevVolume[0] = vl;
- prevVolume[1] = vr;
- adjustVolumeRamp(false);
- }
- // constant gain
- else {
- const int16_t vl = volume[0];
- const int16_t vr = volume[1];
- do {
- int16_t l = *in++;
- out[0] = mulAdd(l, vl, out[0]);
- out[1] = mulAdd(l, vr, out[1]);
- out += 2;
- } while (--frameCount);
- }
- }
- mIn = in;
-}
-
-// no-op case
-void AudioMixer::process__nop()
-{
- ALOGVV("process__nop\n");
-
- for (const auto &pair : mGroups) {
- // process by group of tracks with same output buffer to
- // avoid multiple memset() on same buffer
- const auto &group = pair.second;
-
- const std::shared_ptr<Track> &t = mTracks[group[0]];
- memset(t->mainBuffer, 0,
- mFrameCount * audio_bytes_per_frame(
- t->mMixerChannelCount + t->mMixerHapticChannelCount, t->mMixerFormat));
-
- // now consume data
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- size_t outFrames = mFrameCount;
- while (outFrames) {
- t->buffer.frameCount = outFrames;
- t->bufferProvider->getNextBuffer(&t->buffer);
- if (t->buffer.raw == NULL) break;
- outFrames -= t->buffer.frameCount;
- t->bufferProvider->releaseBuffer(&t->buffer);
- }
- }
- }
-}
-
-// generic code without resampling
-void AudioMixer::process__genericNoResampling()
-{
- ALOGVV("process__genericNoResampling\n");
- int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
-
- for (const auto &pair : mGroups) {
- // process by group of tracks with same output main buffer to
- // avoid multiple memset() on same buffer
- const auto &group = pair.second;
-
- // acquire buffer
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- t->buffer.frameCount = mFrameCount;
- t->bufferProvider->getNextBuffer(&t->buffer);
- t->frameCount = t->buffer.frameCount;
- t->mIn = t->buffer.raw;
- }
-
- int32_t *out = (int *)pair.first;
- size_t numFrames = 0;
- do {
- const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames);
- memset(outTemp, 0, sizeof(outTemp));
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- int32_t *aux = NULL;
- if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
- aux = t->auxBuffer + numFrames;
- }
- for (int outFrames = frameCount; outFrames > 0; ) {
- // t->in == nullptr can happen if the track was flushed just after having
- // been enabled for mixing.
- if (t->mIn == nullptr) {
- break;
- }
- size_t inFrames = (t->frameCount > outFrames)?outFrames:t->frameCount;
- if (inFrames > 0) {
- (t.get()->*t->hook)(
- outTemp + (frameCount - outFrames) * t->mMixerChannelCount,
- inFrames, mResampleTemp.get() /* naked ptr */, aux);
- t->frameCount -= inFrames;
- outFrames -= inFrames;
- if (CC_UNLIKELY(aux != NULL)) {
- aux += inFrames;
- }
- }
- if (t->frameCount == 0 && outFrames) {
- t->bufferProvider->releaseBuffer(&t->buffer);
- t->buffer.frameCount = (mFrameCount - numFrames) -
- (frameCount - outFrames);
- t->bufferProvider->getNextBuffer(&t->buffer);
- t->mIn = t->buffer.raw;
- if (t->mIn == nullptr) {
- break;
- }
- t->frameCount = t->buffer.frameCount;
- }
- }
- }
-
- const std::shared_ptr<Track> &t1 = mTracks[group[0]];
- convertMixerFormat(out, t1->mMixerFormat, outTemp, t1->mMixerInFormat,
- frameCount * t1->mMixerChannelCount);
- // TODO: fix ugly casting due to choice of out pointer type
- out = reinterpret_cast<int32_t*>((uint8_t*)out
- + frameCount * t1->mMixerChannelCount
- * audio_bytes_per_sample(t1->mMixerFormat));
- numFrames += frameCount;
- } while (numFrames < mFrameCount);
-
- // release each track's buffer
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- t->bufferProvider->releaseBuffer(&t->buffer);
- }
- }
-}
-
-// generic code with resampling
-void AudioMixer::process__genericResampling()
-{
- ALOGVV("process__genericResampling\n");
- int32_t * const outTemp = mOutputTemp.get(); // naked ptr
- size_t numFrames = mFrameCount;
-
- for (const auto &pair : mGroups) {
- const auto &group = pair.second;
- const std::shared_ptr<Track> &t1 = mTracks[group[0]];
-
- // clear temp buffer
- memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount);
- for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
- int32_t *aux = NULL;
- if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
- aux = t->auxBuffer;
- }
-
- // this is a little goofy, on the resampling case we don't
- // acquire/release the buffers because it's done by
- // the resampler.
- if (t->needs & NEEDS_RESAMPLE) {
- (t.get()->*t->hook)(outTemp, numFrames, mResampleTemp.get() /* naked ptr */, aux);
- } else {
-
- size_t outFrames = 0;
-
- while (outFrames < numFrames) {
- t->buffer.frameCount = numFrames - outFrames;
- t->bufferProvider->getNextBuffer(&t->buffer);
- t->mIn = t->buffer.raw;
- // t->mIn == nullptr can happen if the track was flushed just after having
- // been enabled for mixing.
- if (t->mIn == nullptr) break;
-
- (t.get()->*t->hook)(
- outTemp + outFrames * t->mMixerChannelCount, t->buffer.frameCount,
- mResampleTemp.get() /* naked ptr */,
- aux != nullptr ? aux + outFrames : nullptr);
- outFrames += t->buffer.frameCount;
-
- t->bufferProvider->releaseBuffer(&t->buffer);
- }
- }
- }
- convertMixerFormat(t1->mainBuffer, t1->mMixerFormat,
- outTemp, t1->mMixerInFormat, numFrames * t1->mMixerChannelCount);
- }
-}
-
-// one track, 16 bits stereo without resampling is the most common case
-void AudioMixer::process__oneTrack16BitsStereoNoResampling()
-{
- ALOGVV("process__oneTrack16BitsStereoNoResampling\n");
- LOG_ALWAYS_FATAL_IF(mEnabled.size() != 0,
- "%zu != 1 tracks enabled", mEnabled.size());
- const int name = mEnabled[0];
- const std::shared_ptr<Track> &t = mTracks[name];
-
- AudioBufferProvider::Buffer& b(t->buffer);
-
- int32_t* out = t->mainBuffer;
- float *fout = reinterpret_cast<float*>(out);
- size_t numFrames = mFrameCount;
-
- const int16_t vl = t->volume[0];
- const int16_t vr = t->volume[1];
- const uint32_t vrl = t->volumeRL;
- while (numFrames) {
- b.frameCount = numFrames;
- t->bufferProvider->getNextBuffer(&b);
- const int16_t *in = b.i16;
-
- // in == NULL can happen if the track was flushed just after having
- // been enabled for mixing.
- if (in == NULL || (((uintptr_t)in) & 3)) {
- if ( AUDIO_FORMAT_PCM_FLOAT == t->mMixerFormat ) {
- memset((char*)fout, 0, numFrames
- * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
- } else {
- memset((char*)out, 0, numFrames
- * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
- }
- ALOGE_IF((((uintptr_t)in) & 3),
- "process__oneTrack16BitsStereoNoResampling: misaligned buffer"
- " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
- in, name, t->channelCount, t->needs, vrl, t->mVolume[0], t->mVolume[1]);
- return;
- }
- size_t outFrames = b.frameCount;
-
- switch (t->mMixerFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- int32_t l = mulRL(1, rl, vrl);
- int32_t r = mulRL(0, rl, vrl);
- *fout++ = float_from_q4_27(l);
- *fout++ = float_from_q4_27(r);
- // Note: In case of later int16_t sink output,
- // conversion and clamping is done by memcpy_to_i16_from_float().
- } while (--outFrames);
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
- // volume is boosted, so we might need to clamp even though
- // we process only one track.
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- int32_t l = mulRL(1, rl, vrl) >> 12;
- int32_t r = mulRL(0, rl, vrl) >> 12;
- // clamping...
- l = clamp16(l);
- r = clamp16(r);
- *out++ = (r<<16) | (l & 0xFFFF);
- } while (--outFrames);
- } else {
- do {
- uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
- in += 2;
- int32_t l = mulRL(1, rl, vrl) >> 12;
- int32_t r = mulRL(0, rl, vrl) >> 12;
- *out++ = (r<<16) | (l & 0xFFFF);
- } while (--outFrames);
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixer format: %d", t->mMixerFormat);
- }
- numFrames -= b.frameCount;
- t->bufferProvider->releaseBuffer(&b);
- }
-}
-
/*static*/ pthread_once_t AudioMixer::sOnceControl = PTHREAD_ONCE_INIT;
/*static*/ void AudioMixer::sInitRoutine()
@@ -1656,211 +523,71 @@
DownmixerBufferProvider::init(); // for the downmixer
}
-/* TODO: consider whether this level of optimization is necessary.
- * Perhaps just stick with a single for loop.
- */
-
-// Needs to derive a compile time constant (constexpr). Could be targeted to go
-// to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication.
-#define MIXTYPE_MONOVOL(mixtype) ((mixtype) == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \
- (mixtype) == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : (mixtype))
-
-/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE,
- typename TO, typename TI, typename TV, typename TA, typename TAV>
-static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
- const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
+std::shared_ptr<AudioMixerBase::TrackBase> AudioMixer::preCreateTrack()
{
- switch (channels) {
- case 1:
- volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 2:
- volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 3:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 4:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 5:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 6:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 7:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- case 8:
- volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out,
- frameCount, in, aux, vol, volinc, vola, volainc);
- break;
- }
+ return std::make_shared<Track>();
}
-/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE,
- typename TO, typename TI, typename TV, typename TA, typename TAV>
-static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
- const TI* in, TA* aux, const TV *vol, TAV vola)
+status_t AudioMixer::postCreateTrack(TrackBase *track)
{
- switch (channels) {
- case 1:
- volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola);
- break;
- case 2:
- volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola);
- break;
- case 3:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola);
- break;
- case 4:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola);
- break;
- case 5:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola);
- break;
- case 6:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola);
- break;
- case 7:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola);
- break;
- case 8:
- volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola);
- break;
+ Track* t = static_cast<Track*>(track);
+
+ audio_channel_mask_t channelMask = t->channelMask;
+ t->mHapticChannelMask = channelMask & AUDIO_CHANNEL_HAPTIC_ALL;
+ t->mHapticChannelCount = audio_channel_count_from_out_mask(t->mHapticChannelMask);
+ channelMask &= ~AUDIO_CHANNEL_HAPTIC_ALL;
+ t->channelCount = audio_channel_count_from_out_mask(channelMask);
+ ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
+ "Non-stereo channel mask: %d\n", channelMask);
+ t->channelMask = channelMask;
+ t->mInputBufferProvider = NULL;
+ t->mDownmixRequiresFormat = AUDIO_FORMAT_INVALID; // no format required
+ t->mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
+ // haptic
+ t->mHapticPlaybackEnabled = false;
+ t->mHapticIntensity = HAPTIC_SCALE_NONE;
+ t->mMixerHapticChannelMask = AUDIO_CHANNEL_NONE;
+ t->mMixerHapticChannelCount = 0;
+ t->mAdjustInChannelCount = t->channelCount + t->mHapticChannelCount;
+ t->mAdjustOutChannelCount = t->channelCount + t->mMixerHapticChannelCount;
+ t->mAdjustNonDestructiveInChannelCount = t->mAdjustOutChannelCount;
+ t->mAdjustNonDestructiveOutChannelCount = t->channelCount;
+ t->mKeepContractedChannels = false;
+ // Check the downmixing (or upmixing) requirements.
+ status_t status = t->prepareForDownmix();
+ if (status != OK) {
+ ALOGE("AudioMixer::getTrackName invalid channelMask (%#x)", channelMask);
+ return BAD_VALUE;
}
+ // prepareForDownmix() may change mDownmixRequiresFormat
+ ALOGVV("mMixerFormat:%#x mMixerInFormat:%#x\n", t->mMixerFormat, t->mMixerInFormat);
+ t->prepareForReformat();
+ t->prepareForAdjustChannelsNonDestructive(mFrameCount);
+ t->prepareForAdjustChannels();
+ return OK;
}
-/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * USEFLOATVOL (set to true if float volume is used)
- * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
- typename TO, typename TI, typename TA>
-void AudioMixer::Track::volumeMix(TO *out, size_t outFrames,
- const TI *in, TA *aux, bool ramp)
+void AudioMixer::preProcess()
{
- if (USEFLOATVOL) {
- if (ramp) {
- volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- mPrevVolume, mVolumeInc,
-#ifdef FLOAT_AUX
- &mPrevAuxLevel, mAuxInc
-#else
- &prevAuxLevel, auxInc
-#endif
- );
- if (ADJUSTVOL) {
- adjustVolumeRamp(aux != NULL, true);
- }
- } else {
- volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- mVolume,
-#ifdef FLOAT_AUX
- mAuxLevel
-#else
- auxLevel
-#endif
- );
- }
- } else {
- if (ramp) {
- volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- prevVolume, volumeInc, &prevAuxLevel, auxInc);
- if (ADJUSTVOL) {
- adjustVolumeRamp(aux != NULL);
- }
- } else {
- volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
- volume, auxLevel);
+ for (const auto &pair : mTracks) {
+ // Clear contracted buffer before processing if contracted channels are saved
+ const std::shared_ptr<TrackBase> &tb = pair.second;
+ Track *t = static_cast<Track*>(tb.get());
+ if (t->mKeepContractedChannels) {
+ t->clearContractedBuffer();
}
}
}
-/* This process hook is called when there is a single track without
- * aux buffer, volume ramp, or resampling.
- * TODO: Update the hook selection: this can properly handle aux and ramp.
- *
- * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27)
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::process__noResampleOneTrack()
+void AudioMixer::postProcess()
{
- ALOGVV("process__noResampleOneTrack\n");
- LOG_ALWAYS_FATAL_IF(mEnabled.size() != 1,
- "%zu != 1 tracks enabled", mEnabled.size());
- const std::shared_ptr<Track> &t = mTracks[mEnabled[0]];
- const uint32_t channels = t->mMixerChannelCount;
- TO* out = reinterpret_cast<TO*>(t->mainBuffer);
- TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
- const bool ramp = t->needsRamp();
-
- for (size_t numFrames = mFrameCount; numFrames > 0; ) {
- AudioBufferProvider::Buffer& b(t->buffer);
- // get input buffer
- b.frameCount = numFrames;
- t->bufferProvider->getNextBuffer(&b);
- const TI *in = reinterpret_cast<TI*>(b.raw);
-
- // in == NULL can happen if the track was flushed just after having
- // been enabled for mixing.
- if (in == NULL || (((uintptr_t)in) & 3)) {
- memset(out, 0, numFrames
- * channels * audio_bytes_per_sample(t->mMixerFormat));
- ALOGE_IF((((uintptr_t)in) & 3), "process__noResampleOneTrack: bus error: "
- "buffer %p track %p, channels %d, needs %#x",
- in, &t, t->channelCount, t->needs);
- return;
- }
-
- const size_t outFrames = b.frameCount;
- t->volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (
- out, outFrames, in, aux, ramp);
-
- out += outFrames * channels;
- if (aux != NULL) {
- aux += outFrames;
- }
- numFrames -= b.frameCount;
-
- // release buffer
- t->bufferProvider->releaseBuffer(&b);
- }
- if (ramp) {
- t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
- }
-}
-
-void AudioMixer::processHapticData()
-{
+ // Process haptic data.
// Need to keep consistent with VibrationEffect.scale(int, float, int)
for (const auto &pair : mGroups) {
// process by group of tracks with same output main buffer.
const auto &group = pair.second;
for (const int name : group) {
- const std::shared_ptr<Track> &t = mTracks[name];
+ const std::shared_ptr<Track> &t = getTrack(name);
if (t->mHapticPlaybackEnabled) {
size_t sampleCount = mFrameCount * t->mMixerHapticChannelCount;
float gamma = t->getHapticScaleGamma();
@@ -1887,225 +614,5 @@
}
}
-/* This track hook is called to do resampling then mixing,
- * pulling from the track's upstream AudioBufferProvider.
- *
- * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::Track::track__Resample(TO* out, size_t outFrameCount, TO* temp, TA* aux)
-{
- ALOGVV("track__Resample\n");
- mResampler->setSampleRate(sampleRate);
- const bool ramp = needsRamp();
- if (ramp || aux != NULL) {
- // if ramp: resample with unity gain to temp buffer and scale/mix in 2nd step.
- // if aux != NULL: resample with unity gain to temp buffer then apply send level.
-
- mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
- memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(TO));
- mResampler->resample((int32_t*)temp, outFrameCount, bufferProvider);
-
- volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
- out, outFrameCount, temp, aux, ramp);
-
- } else { // constant volume gain
- mResampler->setVolume(mVolume[0], mVolume[1]);
- mResampler->resample((int32_t*)out, outFrameCount, bufferProvider);
- }
-}
-
-/* This track hook is called to mix a track, when no resampling is required.
- * The input buffer should be present in in.
- *
- * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
- * TO: int32_t (Q4.27) or float
- * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
- * TA: int32_t (Q4.27) or float
- */
-template <int MIXTYPE, typename TO, typename TI, typename TA>
-void AudioMixer::Track::track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux)
-{
- ALOGVV("track__NoResample\n");
- const TI *in = static_cast<const TI *>(mIn);
-
- volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
- out, frameCount, in, aux, needsRamp());
-
- // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
- // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
- in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * mMixerChannelCount;
- mIn = in;
-}
-
-/* The Mixer engine generates either int32_t (Q4_27) or float data.
- * We use this function to convert the engine buffers
- * to the desired mixer output format, either int16_t (Q.15) or float.
- */
-/* static */
-void AudioMixer::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
- void *in, audio_format_t mixerInFormat, size_t sampleCount)
-{
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- memcpy_to_float_from_q4_27((float*)out, (const int32_t*)in, sampleCount);
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- memcpy_to_i16_from_q4_27((int16_t*)out, (const int32_t*)in, sampleCount);
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
-}
-
-/* Returns the proper track hook to use for mixing the track into the output buffer.
- */
-/* static */
-AudioMixer::hook_t AudioMixer::Track::getTrackHook(int trackType, uint32_t channelCount,
- audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
-{
- if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
- switch (trackType) {
- case TRACKTYPE_NOP:
- return &Track::track__nop;
- case TRACKTYPE_RESAMPLE:
- return &Track::track__genericResample;
- case TRACKTYPE_NORESAMPLEMONO:
- return &Track::track__16BitsMono;
- case TRACKTYPE_NORESAMPLE:
- return &Track::track__16BitsStereo;
- default:
- LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
- break;
- }
- }
- LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
- switch (trackType) {
- case TRACKTYPE_NOP:
- return &Track::track__nop;
- case TRACKTYPE_RESAMPLE:
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return (AudioMixer::hook_t) &Track::track__Resample<
- MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return (AudioMixer::hook_t) &Track::track__Resample<
- MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- break;
- case TRACKTYPE_NORESAMPLEMONO:
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- break;
- case TRACKTYPE_NORESAMPLE:
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return (AudioMixer::hook_t) &Track::track__NoResample<
- MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
- break;
- }
- return NULL;
-}
-
-/* Returns the proper process hook for mixing tracks. Currently works only for
- * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
- *
- * TODO: Due to the special mixing considerations of duplicating to
- * a stereo output track, the input track cannot be MONO. This should be
- * prevented by the caller.
- */
-/* static */
-AudioMixer::process_hook_t AudioMixer::getProcessHook(
- int processType, uint32_t channelCount,
- audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
-{
- if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
- LOG_ALWAYS_FATAL("bad processType: %d", processType);
- return NULL;
- }
- if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
- return &AudioMixer::process__oneTrack16BitsStereoNoResampling;
- }
- LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
- switch (mixerInFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- case AUDIO_FORMAT_PCM_16_BIT:
- switch (mixerOutFormat) {
- case AUDIO_FORMAT_PCM_FLOAT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- case AUDIO_FORMAT_PCM_16_BIT:
- return &AudioMixer::process__noResampleOneTrack<
- MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
- default:
- LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
- break;
- }
- break;
- default:
- LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
- break;
- }
- return NULL;
-}
-
// ----------------------------------------------------------------------------
} // namespace android
diff --git a/media/libaudioprocessing/AudioMixerBase.cpp b/media/libaudioprocessing/AudioMixerBase.cpp
new file mode 100644
index 0000000..75c077d
--- /dev/null
+++ b/media/libaudioprocessing/AudioMixerBase.cpp
@@ -0,0 +1,1692 @@
+/*
+**
+** Copyright 2019, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#define LOG_TAG "AudioMixer"
+//#define LOG_NDEBUG 0
+
+#include <sstream>
+#include <string.h>
+
+#include <audio_utils/primitives.h>
+#include <cutils/compiler.h>
+#include <media/AudioMixerBase.h>
+#include <utils/Log.h>
+
+#include "AudioMixerOps.h"
+
+// The FCC_2 macro refers to the Fixed Channel Count of 2 for the legacy integer mixer.
+#ifndef FCC_2
+#define FCC_2 2
+#endif
+
+// Look for MONO_HACK for any Mono hack involving legacy mono channel to
+// stereo channel conversion.
+
+/* VERY_VERY_VERBOSE_LOGGING will show exactly which process hook and track hook is
+ * being used. This is a considerable amount of log spam, so don't enable unless you
+ * are verifying the hook based code.
+ */
+//#define VERY_VERY_VERBOSE_LOGGING
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+//define ALOGVV printf // for test-mixer.cpp
+#else
+#define ALOGVV(a...) do { } while (0)
+#endif
+
+// TODO: remove BLOCKSIZE unit of processing - it isn't needed anymore.
+static constexpr int BLOCKSIZE = 16;
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+bool AudioMixerBase::isValidFormat(audio_format_t format) const
+{
+ switch (format) {
+ case AUDIO_FORMAT_PCM_8_BIT:
+ case AUDIO_FORMAT_PCM_16_BIT:
+ case AUDIO_FORMAT_PCM_24_BIT_PACKED:
+ case AUDIO_FORMAT_PCM_32_BIT:
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return true;
+ default:
+ return false;
+ }
+}
+
+bool AudioMixerBase::isValidChannelMask(audio_channel_mask_t channelMask) const
+{
+ return audio_channel_count_from_out_mask(channelMask) <= MAX_NUM_CHANNELS;
+}
+
+std::shared_ptr<AudioMixerBase::TrackBase> AudioMixerBase::preCreateTrack()
+{
+ return std::make_shared<TrackBase>();
+}
+
+status_t AudioMixerBase::create(
+ int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId)
+{
+ LOG_ALWAYS_FATAL_IF(exists(name), "name %d already exists", name);
+
+ if (!isValidChannelMask(channelMask)) {
+ ALOGE("%s invalid channelMask: %#x", __func__, channelMask);
+ return BAD_VALUE;
+ }
+ if (!isValidFormat(format)) {
+ ALOGE("%s invalid format: %#x", __func__, format);
+ return BAD_VALUE;
+ }
+
+ auto t = preCreateTrack();
+ {
+ // TODO: move initialization to the Track constructor.
+ // assume default parameters for the track, except where noted below
+ t->needs = 0;
+
+ // Integer volume.
+ // Currently integer volume is kept for the legacy integer mixer.
+ // Will be removed when the legacy mixer path is removed.
+ t->volume[0] = 0;
+ t->volume[1] = 0;
+ t->prevVolume[0] = 0 << 16;
+ t->prevVolume[1] = 0 << 16;
+ t->volumeInc[0] = 0;
+ t->volumeInc[1] = 0;
+ t->auxLevel = 0;
+ t->auxInc = 0;
+ t->prevAuxLevel = 0;
+
+ // Floating point volume.
+ t->mVolume[0] = 0.f;
+ t->mVolume[1] = 0.f;
+ t->mPrevVolume[0] = 0.f;
+ t->mPrevVolume[1] = 0.f;
+ t->mVolumeInc[0] = 0.;
+ t->mVolumeInc[1] = 0.;
+ t->mAuxLevel = 0.;
+ t->mAuxInc = 0.;
+ t->mPrevAuxLevel = 0.;
+
+ // no initialization needed
+ // t->frameCount
+ t->channelCount = audio_channel_count_from_out_mask(channelMask);
+ t->enabled = false;
+ ALOGV_IF(audio_channel_mask_get_bits(channelMask) != AUDIO_CHANNEL_OUT_STEREO,
+ "Non-stereo channel mask: %d\n", channelMask);
+ t->channelMask = channelMask;
+ t->sessionId = sessionId;
+ // setBufferProvider(name, AudioBufferProvider *) is required before enable(name)
+ t->bufferProvider = NULL;
+ t->buffer.raw = NULL;
+ // no initialization needed
+ // t->buffer.frameCount
+ t->hook = NULL;
+ t->mIn = NULL;
+ t->sampleRate = mSampleRate;
+ // setParameter(name, TRACK, MAIN_BUFFER, mixBuffer) is required before enable(name)
+ t->mainBuffer = NULL;
+ t->auxBuffer = NULL;
+ t->mMixerFormat = AUDIO_FORMAT_PCM_16_BIT;
+ t->mFormat = format;
+ t->mMixerInFormat = kUseFloat && kUseNewMixer ?
+ AUDIO_FORMAT_PCM_FLOAT : AUDIO_FORMAT_PCM_16_BIT;
+ t->mMixerChannelMask = audio_channel_mask_from_representation_and_bits(
+ AUDIO_CHANNEL_REPRESENTATION_POSITION, AUDIO_CHANNEL_OUT_STEREO);
+ t->mMixerChannelCount = audio_channel_count_from_out_mask(t->mMixerChannelMask);
+ status_t status = postCreateTrack(t.get());
+ if (status != OK) return status;
+ mTracks[name] = t;
+ return OK;
+ }
+}
+
+// Called when channel masks have changed for a track name
+bool AudioMixerBase::setChannelMasks(int name,
+ audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask)
+{
+ LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+ const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+ if (trackChannelMask == track->channelMask && mixerChannelMask == track->mMixerChannelMask) {
+ return false; // no need to change
+ }
+ // always recompute for both channel masks even if only one has changed.
+ const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
+ const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
+
+ ALOG_ASSERT(trackChannelCount && mixerChannelCount);
+ track->channelMask = trackChannelMask;
+ track->channelCount = trackChannelCount;
+ track->mMixerChannelMask = mixerChannelMask;
+ track->mMixerChannelCount = mixerChannelCount;
+
+ // Resampler channels may have changed.
+ track->recreateResampler(mSampleRate);
+ return true;
+}
+
+void AudioMixerBase::destroy(int name)
+{
+ LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+ ALOGV("deleteTrackName(%d)", name);
+
+ if (mTracks[name]->enabled) {
+ invalidate();
+ }
+ mTracks.erase(name); // deallocate track
+}
+
+void AudioMixerBase::enable(int name)
+{
+ LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+ const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+ if (!track->enabled) {
+ track->enabled = true;
+ ALOGV("enable(%d)", name);
+ invalidate();
+ }
+}
+
+void AudioMixerBase::disable(int name)
+{
+ LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+ const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+ if (track->enabled) {
+ track->enabled = false;
+ ALOGV("disable(%d)", name);
+ invalidate();
+ }
+}
+
+/* Sets the volume ramp variables for the AudioMixer.
+ *
+ * The volume ramp variables are used to transition from the previous
+ * volume to the set volume. ramp controls the duration of the transition.
+ * Its value is typically one state framecount period, but may also be 0,
+ * meaning "immediate."
+ *
+ * FIXME: 1) Volume ramp is enabled only if there is a nonzero integer increment
+ * even if there is a nonzero floating point increment (in that case, the volume
+ * change is immediate). This restriction should be changed when the legacy mixer
+ * is removed (see #2).
+ * FIXME: 2) Integer volume variables are used for Legacy mixing and should be removed
+ * when no longer needed.
+ *
+ * @param newVolume set volume target in floating point [0.0, 1.0].
+ * @param ramp number of frames to increment over. if ramp is 0, the volume
+ * should be set immediately. Currently ramp should not exceed 65535 (frames).
+ * @param pIntSetVolume pointer to the U4.12 integer target volume, set on return.
+ * @param pIntPrevVolume pointer to the U4.28 integer previous volume, set on return.
+ * @param pIntVolumeInc pointer to the U4.28 increment per output audio frame, set on return.
+ * @param pSetVolume pointer to the float target volume, set on return.
+ * @param pPrevVolume pointer to the float previous volume, set on return.
+ * @param pVolumeInc pointer to the float increment per output audio frame, set on return.
+ * @return true if the volume has changed, false if volume is same.
+ */
+static inline bool setVolumeRampVariables(float newVolume, int32_t ramp,
+ int16_t *pIntSetVolume, int32_t *pIntPrevVolume, int32_t *pIntVolumeInc,
+ float *pSetVolume, float *pPrevVolume, float *pVolumeInc) {
+ // check floating point volume to see if it is identical to the previously
+ // set volume.
+ // We do not use a tolerance here (and reject changes too small)
+ // as it may be confusing to use a different value than the one set.
+ // If the resulting volume is too small to ramp, it is a direct set of the volume.
+ if (newVolume == *pSetVolume) {
+ return false;
+ }
+ if (newVolume < 0) {
+ newVolume = 0; // should not have negative volumes
+ } else {
+ switch (fpclassify(newVolume)) {
+ case FP_SUBNORMAL:
+ case FP_NAN:
+ newVolume = 0;
+ break;
+ case FP_ZERO:
+ break; // zero volume is fine
+ case FP_INFINITE:
+ // Infinite volume could be handled consistently since
+ // floating point math saturates at infinities,
+ // but we limit volume to unity gain float.
+ // ramp = 0; break;
+ //
+ newVolume = AudioMixerBase::UNITY_GAIN_FLOAT;
+ break;
+ case FP_NORMAL:
+ default:
+ // Floating point does not have problems with overflow wrap
+ // that integer has. However, we limit the volume to
+ // unity gain here.
+ // TODO: Revisit the volume limitation and perhaps parameterize.
+ if (newVolume > AudioMixerBase::UNITY_GAIN_FLOAT) {
+ newVolume = AudioMixerBase::UNITY_GAIN_FLOAT;
+ }
+ break;
+ }
+ }
+
+ // set floating point volume ramp
+ if (ramp != 0) {
+ // when the ramp completes, *pPrevVolume is set to *pSetVolume, so there
+ // is no computational mismatch; hence equality is checked here.
+ ALOGD_IF(*pPrevVolume != *pSetVolume, "previous float ramp hasn't finished,"
+ " prev:%f set_to:%f", *pPrevVolume, *pSetVolume);
+ const float inc = (newVolume - *pPrevVolume) / ramp; // could be inf, nan, subnormal
+ // could be inf, cannot be nan, subnormal
+ const float maxv = std::max(newVolume, *pPrevVolume);
+
+ if (isnormal(inc) // inc must be a normal number (no subnormals, infinite, nan)
+ && maxv + inc != maxv) { // inc must make forward progress
+ *pVolumeInc = inc;
+ // ramp is set now.
+ // Note: if newVolume is 0, then near the end of the ramp,
+ // it may be possible that the ramped volume may be subnormal or
+ // temporarily negative by a small amount or subnormal due to floating
+ // point inaccuracies.
+ } else {
+ ramp = 0; // ramp not allowed
+ }
+ }
+
+ // compute and check integer volume, no need to check negative values
+ // The integer volume is limited to "unity_gain" to avoid wrapping and other
+ // audio artifacts, so it never reaches the range limit of U4.28.
+ // We safely use signed 16 and 32 bit integers here.
+ const float scaledVolume = newVolume * AudioMixerBase::UNITY_GAIN_INT; // not neg, subnormal, nan
+ const int32_t intVolume = (scaledVolume >= (float)AudioMixerBase::UNITY_GAIN_INT) ?
+ AudioMixerBase::UNITY_GAIN_INT : (int32_t)scaledVolume;
+
+ // set integer volume ramp
+ if (ramp != 0) {
+ // integer volume is U4.12 (to use 16 bit multiplies), but ramping uses U4.28.
+ // when the ramp completes, *pIntPrevVolume is set to *pIntSetVolume << 16, so there
+ // is no computational mismatch; hence equality is checked here.
+ ALOGD_IF(*pIntPrevVolume != *pIntSetVolume << 16, "previous int ramp hasn't finished,"
+ " prev:%d set_to:%d", *pIntPrevVolume, *pIntSetVolume << 16);
+ const int32_t inc = ((intVolume << 16) - *pIntPrevVolume) / ramp;
+
+ if (inc != 0) { // inc must make forward progress
+ *pIntVolumeInc = inc;
+ } else {
+ ramp = 0; // ramp not allowed
+ }
+ }
+
+ // if no ramp, or ramp not allowed, then clear float and integer increments
+ if (ramp == 0) {
+ *pVolumeInc = 0;
+ *pPrevVolume = newVolume;
+ *pIntVolumeInc = 0;
+ *pIntPrevVolume = intVolume << 16;
+ }
+ *pSetVolume = newVolume;
+ *pIntSetVolume = intVolume;
+ return true;
+}
+
+void AudioMixerBase::setParameter(int name, int target, int param, void *value)
+{
+ LOG_ALWAYS_FATAL_IF(!exists(name), "invalid name: %d", name);
+ const std::shared_ptr<TrackBase> &track = mTracks[name];
+
+ int valueInt = static_cast<int>(reinterpret_cast<uintptr_t>(value));
+ int32_t *valueBuf = reinterpret_cast<int32_t*>(value);
+
+ switch (target) {
+
+ case TRACK:
+ switch (param) {
+ case CHANNEL_MASK: {
+ const audio_channel_mask_t trackChannelMask =
+ static_cast<audio_channel_mask_t>(valueInt);
+ if (setChannelMasks(name, trackChannelMask, track->mMixerChannelMask)) {
+ ALOGV("setParameter(TRACK, CHANNEL_MASK, %x)", trackChannelMask);
+ invalidate();
+ }
+ } break;
+ case MAIN_BUFFER:
+ if (track->mainBuffer != valueBuf) {
+ track->mainBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, MAIN_BUFFER, %p)", valueBuf);
+ invalidate();
+ }
+ break;
+ case AUX_BUFFER:
+ if (track->auxBuffer != valueBuf) {
+ track->auxBuffer = valueBuf;
+ ALOGV("setParameter(TRACK, AUX_BUFFER, %p)", valueBuf);
+ invalidate();
+ }
+ break;
+ case FORMAT: {
+ audio_format_t format = static_cast<audio_format_t>(valueInt);
+ if (track->mFormat != format) {
+ ALOG_ASSERT(audio_is_linear_pcm(format), "Invalid format %#x", format);
+ track->mFormat = format;
+ ALOGV("setParameter(TRACK, FORMAT, %#x)", format);
+ invalidate();
+ }
+ } break;
+ case MIXER_FORMAT: {
+ audio_format_t format = static_cast<audio_format_t>(valueInt);
+ if (track->mMixerFormat != format) {
+ track->mMixerFormat = format;
+ ALOGV("setParameter(TRACK, MIXER_FORMAT, %#x)", format);
+ }
+ } break;
+ case MIXER_CHANNEL_MASK: {
+ const audio_channel_mask_t mixerChannelMask =
+ static_cast<audio_channel_mask_t>(valueInt);
+ if (setChannelMasks(name, track->channelMask, mixerChannelMask)) {
+ ALOGV("setParameter(TRACK, MIXER_CHANNEL_MASK, %#x)", mixerChannelMask);
+ invalidate();
+ }
+ } break;
+ default:
+ LOG_ALWAYS_FATAL("setParameter track: bad param %d", param);
+ }
+ break;
+
+ case RESAMPLE:
+ switch (param) {
+ case SAMPLE_RATE:
+ ALOG_ASSERT(valueInt > 0, "bad sample rate %d", valueInt);
+ if (track->setResampler(uint32_t(valueInt), mSampleRate)) {
+ ALOGV("setParameter(RESAMPLE, SAMPLE_RATE, %u)",
+ uint32_t(valueInt));
+ invalidate();
+ }
+ break;
+ case RESET:
+ track->resetResampler();
+ invalidate();
+ break;
+ case REMOVE:
+ track->mResampler.reset(nullptr);
+ track->sampleRate = mSampleRate;
+ invalidate();
+ break;
+ default:
+ LOG_ALWAYS_FATAL("setParameter resample: bad param %d", param);
+ }
+ break;
+
+ case RAMP_VOLUME:
+ case VOLUME:
+ switch (param) {
+ case AUXLEVEL:
+ if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+ target == RAMP_VOLUME ? mFrameCount : 0,
+ &track->auxLevel, &track->prevAuxLevel, &track->auxInc,
+ &track->mAuxLevel, &track->mPrevAuxLevel, &track->mAuxInc)) {
+ ALOGV("setParameter(%s, AUXLEVEL: %04x)",
+ target == VOLUME ? "VOLUME" : "RAMP_VOLUME", track->auxLevel);
+ invalidate();
+ }
+ break;
+ default:
+ if ((unsigned)param >= VOLUME0 && (unsigned)param < VOLUME0 + MAX_NUM_VOLUMES) {
+ if (setVolumeRampVariables(*reinterpret_cast<float*>(value),
+ target == RAMP_VOLUME ? mFrameCount : 0,
+ &track->volume[param - VOLUME0],
+ &track->prevVolume[param - VOLUME0],
+ &track->volumeInc[param - VOLUME0],
+ &track->mVolume[param - VOLUME0],
+ &track->mPrevVolume[param - VOLUME0],
+ &track->mVolumeInc[param - VOLUME0])) {
+ ALOGV("setParameter(%s, VOLUME%d: %04x)",
+ target == VOLUME ? "VOLUME" : "RAMP_VOLUME", param - VOLUME0,
+ track->volume[param - VOLUME0]);
+ invalidate();
+ }
+ } else {
+ LOG_ALWAYS_FATAL("setParameter volume: bad param %d", param);
+ }
+ }
+ break;
+
+ default:
+ LOG_ALWAYS_FATAL("setParameter: bad target %d", target);
+ }
+}
+
+bool AudioMixerBase::TrackBase::setResampler(uint32_t trackSampleRate, uint32_t devSampleRate)
+{
+ if (trackSampleRate != devSampleRate || mResampler.get() != nullptr) {
+ if (sampleRate != trackSampleRate) {
+ sampleRate = trackSampleRate;
+ if (mResampler.get() == nullptr) {
+ ALOGV("Creating resampler from track %d Hz to device %d Hz",
+ trackSampleRate, devSampleRate);
+ AudioResampler::src_quality quality;
+ // force lowest quality level resampler if use case isn't music or video
+ // FIXME this is flawed for dynamic sample rates, as we choose the resampler
+ // quality level based on the initial ratio, but that could change later.
+ // Should have a way to distinguish tracks with static ratios vs. dynamic ratios.
+ if (isMusicRate(trackSampleRate)) {
+ quality = AudioResampler::DEFAULT_QUALITY;
+ } else {
+ quality = AudioResampler::DYN_LOW_QUALITY;
+ }
+
+ // TODO: Remove MONO_HACK. Resampler sees #channels after the downmixer
+ // but if none exists, it is the channel count (1 for mono).
+ const int resamplerChannelCount = getOutputChannelCount();
+ ALOGVV("Creating resampler:"
+ " format(%#x) channels(%d) devSampleRate(%u) quality(%d)\n",
+ mMixerInFormat, resamplerChannelCount, devSampleRate, quality);
+ mResampler.reset(AudioResampler::create(
+ mMixerInFormat,
+ resamplerChannelCount,
+ devSampleRate, quality));
+ }
+ return true;
+ }
+ }
+ return false;
+}
+
+/* Checks to see if the volume ramp has completed and clears the increment
+ * variables appropriately.
+ *
+ * FIXME: There is code to handle int/float ramp variable switchover should it not
+ * complete within a mixer buffer processing call, but it is preferred to avoid switchover
+ * due to precision issues. The switchover code is included for legacy code purposes
+ * and can be removed once the integer volume is removed.
+ *
+ * It is not sufficient to clear only the volumeInc integer variable because
+ * if one channel requires ramping, all channels are ramped.
+ *
+ * There is a bit of duplicated code here, but it keeps backward compatibility.
+ */
+void AudioMixerBase::TrackBase::adjustVolumeRamp(bool aux, bool useFloat)
+{
+ if (useFloat) {
+ for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
+ if ((mVolumeInc[i] > 0 && mPrevVolume[i] + mVolumeInc[i] >= mVolume[i]) ||
+ (mVolumeInc[i] < 0 && mPrevVolume[i] + mVolumeInc[i] <= mVolume[i])) {
+ volumeInc[i] = 0;
+ prevVolume[i] = volume[i] << 16;
+ mVolumeInc[i] = 0.;
+ mPrevVolume[i] = mVolume[i];
+ } else {
+ //ALOGV("ramp: %f %f %f", mVolume[i], mPrevVolume[i], mVolumeInc[i]);
+ prevVolume[i] = u4_28_from_float(mPrevVolume[i]);
+ }
+ }
+ } else {
+ for (uint32_t i = 0; i < MAX_NUM_VOLUMES; i++) {
+ if (((volumeInc[i]>0) && (((prevVolume[i]+volumeInc[i])>>16) >= volume[i])) ||
+ ((volumeInc[i]<0) && (((prevVolume[i]+volumeInc[i])>>16) <= volume[i]))) {
+ volumeInc[i] = 0;
+ prevVolume[i] = volume[i] << 16;
+ mVolumeInc[i] = 0.;
+ mPrevVolume[i] = mVolume[i];
+ } else {
+ //ALOGV("ramp: %d %d %d", volume[i] << 16, prevVolume[i], volumeInc[i]);
+ mPrevVolume[i] = float_from_u4_28(prevVolume[i]);
+ }
+ }
+ }
+
+ if (aux) {
+#ifdef FLOAT_AUX
+ if (useFloat) {
+ if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) ||
+ (mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
+ auxInc = 0;
+ prevAuxLevel = auxLevel << 16;
+ mAuxInc = 0.f;
+ mPrevAuxLevel = mAuxLevel;
+ }
+ } else
+#endif
+ if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) ||
+ (auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
+ auxInc = 0;
+ prevAuxLevel = auxLevel << 16;
+ mAuxInc = 0.f;
+ mPrevAuxLevel = mAuxLevel;
+ }
+ }
+}
+
+void AudioMixerBase::TrackBase::recreateResampler(uint32_t devSampleRate)
+{
+ if (mResampler.get() != nullptr) {
+ const uint32_t resetToSampleRate = sampleRate;
+ mResampler.reset(nullptr);
+ sampleRate = devSampleRate; // without resampler, track rate is device sample rate.
+ // recreate the resampler with updated format, channels, saved sampleRate.
+ setResampler(resetToSampleRate /*trackSampleRate*/, devSampleRate);
+ }
+}
+
+size_t AudioMixerBase::getUnreleasedFrames(int name) const
+{
+ const auto it = mTracks.find(name);
+ if (it != mTracks.end()) {
+ return it->second->getUnreleasedFrames();
+ }
+ return 0;
+}
+
+std::string AudioMixerBase::trackNames() const
+{
+ std::stringstream ss;
+ for (const auto &pair : mTracks) {
+ ss << pair.first << " ";
+ }
+ return ss.str();
+}
+
+void AudioMixerBase::process__validate()
+{
+ // TODO: fix all16BitsStereNoResample logic to
+ // either properly handle muted tracks (it should ignore them)
+ // or remove altogether as an obsolete optimization.
+ bool all16BitsStereoNoResample = true;
+ bool resampling = false;
+ bool volumeRamp = false;
+
+ mEnabled.clear();
+ mGroups.clear();
+ for (const auto &pair : mTracks) {
+ const int name = pair.first;
+ const std::shared_ptr<TrackBase> &t = pair.second;
+ if (!t->enabled) continue;
+
+ mEnabled.emplace_back(name); // we add to mEnabled in order of name.
+ mGroups[t->mainBuffer].emplace_back(name); // mGroups also in order of name.
+
+ uint32_t n = 0;
+ // FIXME can overflow (mask is only 3 bits)
+ n |= NEEDS_CHANNEL_1 + t->channelCount - 1;
+ if (t->doesResample()) {
+ n |= NEEDS_RESAMPLE;
+ }
+ if (t->auxLevel != 0 && t->auxBuffer != NULL) {
+ n |= NEEDS_AUX;
+ }
+
+ if (t->volumeInc[0]|t->volumeInc[1]) {
+ volumeRamp = true;
+ } else if (!t->doesResample() && t->volumeRL == 0) {
+ n |= NEEDS_MUTE;
+ }
+ t->needs = n;
+
+ if (n & NEEDS_MUTE) {
+ t->hook = &TrackBase::track__nop;
+ } else {
+ if (n & NEEDS_AUX) {
+ all16BitsStereoNoResample = false;
+ }
+ if (n & NEEDS_RESAMPLE) {
+ all16BitsStereoNoResample = false;
+ resampling = true;
+ t->hook = TrackBase::getTrackHook(TRACKTYPE_RESAMPLE, t->mMixerChannelCount,
+ t->mMixerInFormat, t->mMixerFormat);
+ ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+ "Track %d needs downmix + resample", name);
+ } else {
+ if ((n & NEEDS_CHANNEL_COUNT__MASK) == NEEDS_CHANNEL_1){
+ t->hook = TrackBase::getTrackHook(
+ (t->mMixerChannelMask == AUDIO_CHANNEL_OUT_STEREO // TODO: MONO_HACK
+ && t->channelMask == AUDIO_CHANNEL_OUT_MONO)
+ ? TRACKTYPE_NORESAMPLEMONO : TRACKTYPE_NORESAMPLE,
+ t->mMixerChannelCount,
+ t->mMixerInFormat, t->mMixerFormat);
+ all16BitsStereoNoResample = false;
+ }
+ if ((n & NEEDS_CHANNEL_COUNT__MASK) >= NEEDS_CHANNEL_2){
+ t->hook = TrackBase::getTrackHook(TRACKTYPE_NORESAMPLE, t->mMixerChannelCount,
+ t->mMixerInFormat, t->mMixerFormat);
+ ALOGV_IF((n & NEEDS_CHANNEL_COUNT__MASK) > NEEDS_CHANNEL_2,
+ "Track %d needs downmix", name);
+ }
+ }
+ }
+ }
+
+ // select the processing hooks
+ mHook = &AudioMixerBase::process__nop;
+ if (mEnabled.size() > 0) {
+ if (resampling) {
+ if (mOutputTemp.get() == nullptr) {
+ mOutputTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
+ }
+ if (mResampleTemp.get() == nullptr) {
+ mResampleTemp.reset(new int32_t[MAX_NUM_CHANNELS * mFrameCount]);
+ }
+ mHook = &AudioMixerBase::process__genericResampling;
+ } else {
+ // we keep temp arrays around.
+ mHook = &AudioMixerBase::process__genericNoResampling;
+ if (all16BitsStereoNoResample && !volumeRamp) {
+ if (mEnabled.size() == 1) {
+ const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+ if ((t->needs & NEEDS_MUTE) == 0) {
+ // The check prevents a muted track from acquiring a process hook.
+ //
+ // This is dangerous if the track is MONO as that requires
+ // special case handling due to implicit channel duplication.
+ // Stereo or Multichannel should actually be fine here.
+ mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+ t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
+ }
+ }
+ }
+ }
+ }
+
+ ALOGV("mixer configuration change: %zu "
+ "all16BitsStereoNoResample=%d, resampling=%d, volumeRamp=%d",
+ mEnabled.size(), all16BitsStereoNoResample, resampling, volumeRamp);
+
+ process();
+
+ // Now that the volume ramp has been done, set optimal state and
+ // track hooks for subsequent mixer process
+ if (mEnabled.size() > 0) {
+ bool allMuted = true;
+
+ for (const int name : mEnabled) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ if (!t->doesResample() && t->volumeRL == 0) {
+ t->needs |= NEEDS_MUTE;
+ t->hook = &TrackBase::track__nop;
+ } else {
+ allMuted = false;
+ }
+ }
+ if (allMuted) {
+ mHook = &AudioMixerBase::process__nop;
+ } else if (all16BitsStereoNoResample) {
+ if (mEnabled.size() == 1) {
+ //const int i = 31 - __builtin_clz(enabledTracks);
+ const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+ // Muted single tracks handled by allMuted above.
+ mHook = getProcessHook(PROCESSTYPE_NORESAMPLEONETRACK,
+ t->mMixerChannelCount, t->mMixerInFormat, t->mMixerFormat);
+ }
+ }
+ }
+}
+
+void AudioMixerBase::TrackBase::track__genericResample(
+ int32_t* out, size_t outFrameCount, int32_t* temp, int32_t* aux)
+{
+ ALOGVV("track__genericResample\n");
+ mResampler->setSampleRate(sampleRate);
+
+ // ramp gain - resample to temp buffer and scale/mix in 2nd step
+ if (aux != NULL) {
+ // always resample with unity gain when sending to auxiliary buffer to be able
+ // to apply send level after resampling
+ mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+ memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(int32_t));
+ mResampler->resample(temp, outFrameCount, bufferProvider);
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+ volumeRampStereo(out, outFrameCount, temp, aux);
+ } else {
+ volumeStereo(out, outFrameCount, temp, aux);
+ }
+ } else {
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+ mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+ memset(temp, 0, outFrameCount * MAX_NUM_CHANNELS * sizeof(int32_t));
+ mResampler->resample(temp, outFrameCount, bufferProvider);
+ volumeRampStereo(out, outFrameCount, temp, aux);
+ }
+
+ // constant gain
+ else {
+ mResampler->setVolume(mVolume[0], mVolume[1]);
+ mResampler->resample(out, outFrameCount, bufferProvider);
+ }
+ }
+}
+
+void AudioMixerBase::TrackBase::track__nop(int32_t* out __unused,
+ size_t outFrameCount __unused, int32_t* temp __unused, int32_t* aux __unused)
+{
+}
+
+void AudioMixerBase::TrackBase::volumeRampStereo(
+ int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+{
+ int32_t vl = prevVolume[0];
+ int32_t vr = prevVolume[1];
+ const int32_t vlInc = volumeInc[0];
+ const int32_t vrInc = volumeInc[1];
+
+ //ALOGD("[0] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ // ramp volume
+ if (CC_UNLIKELY(aux != NULL)) {
+ int32_t va = prevAuxLevel;
+ const int32_t vaInc = auxInc;
+ int32_t l;
+ int32_t r;
+
+ do {
+ l = (*temp++ >> 12);
+ r = (*temp++ >> 12);
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * r;
+ *aux++ += (va >> 17) * (l + r);
+ vl += vlInc;
+ vr += vrInc;
+ va += vaInc;
+ } while (--frameCount);
+ prevAuxLevel = va;
+ } else {
+ do {
+ *out++ += (vl >> 16) * (*temp++ >> 12);
+ *out++ += (vr >> 16) * (*temp++ >> 12);
+ vl += vlInc;
+ vr += vrInc;
+ } while (--frameCount);
+ }
+ prevVolume[0] = vl;
+ prevVolume[1] = vr;
+ adjustVolumeRamp(aux != NULL);
+}
+
+void AudioMixerBase::TrackBase::volumeStereo(
+ int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux)
+{
+ const int16_t vl = volume[0];
+ const int16_t vr = volume[1];
+
+ if (CC_UNLIKELY(aux != NULL)) {
+ const int16_t va = auxLevel;
+ do {
+ int16_t l = (int16_t)(*temp++ >> 12);
+ int16_t r = (int16_t)(*temp++ >> 12);
+ out[0] = mulAdd(l, vl, out[0]);
+ int16_t a = (int16_t)(((int32_t)l + r) >> 1);
+ out[1] = mulAdd(r, vr, out[1]);
+ out += 2;
+ aux[0] = mulAdd(a, va, aux[0]);
+ aux++;
+ } while (--frameCount);
+ } else {
+ do {
+ int16_t l = (int16_t)(*temp++ >> 12);
+ int16_t r = (int16_t)(*temp++ >> 12);
+ out[0] = mulAdd(l, vl, out[0]);
+ out[1] = mulAdd(r, vr, out[1]);
+ out += 2;
+ } while (--frameCount);
+ }
+}
+
+void AudioMixerBase::TrackBase::track__16BitsStereo(
+ int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
+{
+ ALOGVV("track__16BitsStereo\n");
+ const int16_t *in = static_cast<const int16_t *>(mIn);
+
+ if (CC_UNLIKELY(aux != NULL)) {
+ int32_t l;
+ int32_t r;
+ // ramp gain
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+ int32_t vl = prevVolume[0];
+ int32_t vr = prevVolume[1];
+ int32_t va = prevAuxLevel;
+ const int32_t vlInc = volumeInc[0];
+ const int32_t vrInc = volumeInc[1];
+ const int32_t vaInc = auxInc;
+ // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ do {
+ l = (int32_t)*in++;
+ r = (int32_t)*in++;
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * r;
+ *aux++ += (va >> 17) * (l + r);
+ vl += vlInc;
+ vr += vrInc;
+ va += vaInc;
+ } while (--frameCount);
+
+ prevVolume[0] = vl;
+ prevVolume[1] = vr;
+ prevAuxLevel = va;
+ adjustVolumeRamp(true);
+ }
+
+ // constant gain
+ else {
+ const uint32_t vrl = volumeRL;
+ const int16_t va = (int16_t)auxLevel;
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ int16_t a = (int16_t)(((int32_t)in[0] + in[1]) >> 1);
+ in += 2;
+ out[0] = mulAddRL(1, rl, vrl, out[0]);
+ out[1] = mulAddRL(0, rl, vrl, out[1]);
+ out += 2;
+ aux[0] = mulAdd(a, va, aux[0]);
+ aux++;
+ } while (--frameCount);
+ }
+ } else {
+ // ramp gain
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+ int32_t vl = prevVolume[0];
+ int32_t vr = prevVolume[1];
+ const int32_t vlInc = volumeInc[0];
+ const int32_t vrInc = volumeInc[1];
+
+ // ALOGD("[1] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ do {
+ *out++ += (vl >> 16) * (int32_t) *in++;
+ *out++ += (vr >> 16) * (int32_t) *in++;
+ vl += vlInc;
+ vr += vrInc;
+ } while (--frameCount);
+
+ prevVolume[0] = vl;
+ prevVolume[1] = vr;
+ adjustVolumeRamp(false);
+ }
+
+ // constant gain
+ else {
+ const uint32_t vrl = volumeRL;
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ out[0] = mulAddRL(1, rl, vrl, out[0]);
+ out[1] = mulAddRL(0, rl, vrl, out[1]);
+ out += 2;
+ } while (--frameCount);
+ }
+ }
+ mIn = in;
+}
+
+void AudioMixerBase::TrackBase::track__16BitsMono(
+ int32_t* out, size_t frameCount, int32_t* temp __unused, int32_t* aux)
+{
+ ALOGVV("track__16BitsMono\n");
+ const int16_t *in = static_cast<int16_t const *>(mIn);
+
+ if (CC_UNLIKELY(aux != NULL)) {
+ // ramp gain
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1]|auxInc)) {
+ int32_t vl = prevVolume[0];
+ int32_t vr = prevVolume[1];
+ int32_t va = prevAuxLevel;
+ const int32_t vlInc = volumeInc[0];
+ const int32_t vrInc = volumeInc[1];
+ const int32_t vaInc = auxInc;
+
+ // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ do {
+ int32_t l = *in++;
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * l;
+ *aux++ += (va >> 16) * l;
+ vl += vlInc;
+ vr += vrInc;
+ va += vaInc;
+ } while (--frameCount);
+
+ prevVolume[0] = vl;
+ prevVolume[1] = vr;
+ prevAuxLevel = va;
+ adjustVolumeRamp(true);
+ }
+ // constant gain
+ else {
+ const int16_t vl = volume[0];
+ const int16_t vr = volume[1];
+ const int16_t va = (int16_t)auxLevel;
+ do {
+ int16_t l = *in++;
+ out[0] = mulAdd(l, vl, out[0]);
+ out[1] = mulAdd(l, vr, out[1]);
+ out += 2;
+ aux[0] = mulAdd(l, va, aux[0]);
+ aux++;
+ } while (--frameCount);
+ }
+ } else {
+ // ramp gain
+ if (CC_UNLIKELY(volumeInc[0]|volumeInc[1])) {
+ int32_t vl = prevVolume[0];
+ int32_t vr = prevVolume[1];
+ const int32_t vlInc = volumeInc[0];
+ const int32_t vrInc = volumeInc[1];
+
+ // ALOGD("[2] %p: inc=%f, v0=%f, v1=%d, final=%f, count=%d",
+ // t, vlInc/65536.0f, vl/65536.0f, volume[0],
+ // (vl + vlInc*frameCount)/65536.0f, frameCount);
+
+ do {
+ int32_t l = *in++;
+ *out++ += (vl >> 16) * l;
+ *out++ += (vr >> 16) * l;
+ vl += vlInc;
+ vr += vrInc;
+ } while (--frameCount);
+
+ prevVolume[0] = vl;
+ prevVolume[1] = vr;
+ adjustVolumeRamp(false);
+ }
+ // constant gain
+ else {
+ const int16_t vl = volume[0];
+ const int16_t vr = volume[1];
+ do {
+ int16_t l = *in++;
+ out[0] = mulAdd(l, vl, out[0]);
+ out[1] = mulAdd(l, vr, out[1]);
+ out += 2;
+ } while (--frameCount);
+ }
+ }
+ mIn = in;
+}
+
+// no-op case
+void AudioMixerBase::process__nop()
+{
+ ALOGVV("process__nop\n");
+
+ for (const auto &pair : mGroups) {
+ // process by group of tracks with same output buffer to
+ // avoid multiple memset() on same buffer
+ const auto &group = pair.second;
+
+ const std::shared_ptr<TrackBase> &t = mTracks[group[0]];
+ memset(t->mainBuffer, 0,
+ mFrameCount * audio_bytes_per_frame(t->getMixerChannelCount(), t->mMixerFormat));
+
+ // now consume data
+ for (const int name : group) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ size_t outFrames = mFrameCount;
+ while (outFrames) {
+ t->buffer.frameCount = outFrames;
+ t->bufferProvider->getNextBuffer(&t->buffer);
+ if (t->buffer.raw == NULL) break;
+ outFrames -= t->buffer.frameCount;
+ t->bufferProvider->releaseBuffer(&t->buffer);
+ }
+ }
+ }
+}
+
+// generic code without resampling
+void AudioMixerBase::process__genericNoResampling()
+{
+ ALOGVV("process__genericNoResampling\n");
+ int32_t outTemp[BLOCKSIZE * MAX_NUM_CHANNELS] __attribute__((aligned(32)));
+
+ for (const auto &pair : mGroups) {
+ // process by group of tracks with same output main buffer to
+ // avoid multiple memset() on same buffer
+ const auto &group = pair.second;
+
+ // acquire buffer
+ for (const int name : group) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ t->buffer.frameCount = mFrameCount;
+ t->bufferProvider->getNextBuffer(&t->buffer);
+ t->frameCount = t->buffer.frameCount;
+ t->mIn = t->buffer.raw;
+ }
+
+ int32_t *out = (int *)pair.first;
+ size_t numFrames = 0;
+ do {
+ const size_t frameCount = std::min((size_t)BLOCKSIZE, mFrameCount - numFrames);
+ memset(outTemp, 0, sizeof(outTemp));
+ for (const int name : group) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ int32_t *aux = NULL;
+ if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
+ aux = t->auxBuffer + numFrames;
+ }
+ for (int outFrames = frameCount; outFrames > 0; ) {
+ // t->in == nullptr can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (t->mIn == nullptr) {
+ break;
+ }
+ size_t inFrames = (t->frameCount > outFrames)?outFrames:t->frameCount;
+ if (inFrames > 0) {
+ (t.get()->*t->hook)(
+ outTemp + (frameCount - outFrames) * t->mMixerChannelCount,
+ inFrames, mResampleTemp.get() /* naked ptr */, aux);
+ t->frameCount -= inFrames;
+ outFrames -= inFrames;
+ if (CC_UNLIKELY(aux != NULL)) {
+ aux += inFrames;
+ }
+ }
+ if (t->frameCount == 0 && outFrames) {
+ t->bufferProvider->releaseBuffer(&t->buffer);
+ t->buffer.frameCount = (mFrameCount - numFrames) -
+ (frameCount - outFrames);
+ t->bufferProvider->getNextBuffer(&t->buffer);
+ t->mIn = t->buffer.raw;
+ if (t->mIn == nullptr) {
+ break;
+ }
+ t->frameCount = t->buffer.frameCount;
+ }
+ }
+ }
+
+ const std::shared_ptr<TrackBase> &t1 = mTracks[group[0]];
+ convertMixerFormat(out, t1->mMixerFormat, outTemp, t1->mMixerInFormat,
+ frameCount * t1->mMixerChannelCount);
+ // TODO: fix ugly casting due to choice of out pointer type
+ out = reinterpret_cast<int32_t*>((uint8_t*)out
+ + frameCount * t1->mMixerChannelCount
+ * audio_bytes_per_sample(t1->mMixerFormat));
+ numFrames += frameCount;
+ } while (numFrames < mFrameCount);
+
+ // release each track's buffer
+ for (const int name : group) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ t->bufferProvider->releaseBuffer(&t->buffer);
+ }
+ }
+}
+
+// generic code with resampling
+void AudioMixerBase::process__genericResampling()
+{
+ ALOGVV("process__genericResampling\n");
+ int32_t * const outTemp = mOutputTemp.get(); // naked ptr
+ size_t numFrames = mFrameCount;
+
+ for (const auto &pair : mGroups) {
+ const auto &group = pair.second;
+ const std::shared_ptr<TrackBase> &t1 = mTracks[group[0]];
+
+ // clear temp buffer
+ memset(outTemp, 0, sizeof(*outTemp) * t1->mMixerChannelCount * mFrameCount);
+ for (const int name : group) {
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+ int32_t *aux = NULL;
+ if (CC_UNLIKELY(t->needs & NEEDS_AUX)) {
+ aux = t->auxBuffer;
+ }
+
+ // this is a little goofy, on the resampling case we don't
+ // acquire/release the buffers because it's done by
+ // the resampler.
+ if (t->needs & NEEDS_RESAMPLE) {
+ (t.get()->*t->hook)(outTemp, numFrames, mResampleTemp.get() /* naked ptr */, aux);
+ } else {
+
+ size_t outFrames = 0;
+
+ while (outFrames < numFrames) {
+ t->buffer.frameCount = numFrames - outFrames;
+ t->bufferProvider->getNextBuffer(&t->buffer);
+ t->mIn = t->buffer.raw;
+ // t->mIn == nullptr can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (t->mIn == nullptr) break;
+
+ (t.get()->*t->hook)(
+ outTemp + outFrames * t->mMixerChannelCount, t->buffer.frameCount,
+ mResampleTemp.get() /* naked ptr */,
+ aux != nullptr ? aux + outFrames : nullptr);
+ outFrames += t->buffer.frameCount;
+
+ t->bufferProvider->releaseBuffer(&t->buffer);
+ }
+ }
+ }
+ convertMixerFormat(t1->mainBuffer, t1->mMixerFormat,
+ outTemp, t1->mMixerInFormat, numFrames * t1->mMixerChannelCount);
+ }
+}
+
+// one track, 16 bits stereo without resampling is the most common case
+void AudioMixerBase::process__oneTrack16BitsStereoNoResampling()
+{
+ ALOGVV("process__oneTrack16BitsStereoNoResampling\n");
+ LOG_ALWAYS_FATAL_IF(mEnabled.size() != 0,
+ "%zu != 1 tracks enabled", mEnabled.size());
+ const int name = mEnabled[0];
+ const std::shared_ptr<TrackBase> &t = mTracks[name];
+
+ AudioBufferProvider::Buffer& b(t->buffer);
+
+ int32_t* out = t->mainBuffer;
+ float *fout = reinterpret_cast<float*>(out);
+ size_t numFrames = mFrameCount;
+
+ const int16_t vl = t->volume[0];
+ const int16_t vr = t->volume[1];
+ const uint32_t vrl = t->volumeRL;
+ while (numFrames) {
+ b.frameCount = numFrames;
+ t->bufferProvider->getNextBuffer(&b);
+ const int16_t *in = b.i16;
+
+ // in == NULL can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (in == NULL || (((uintptr_t)in) & 3)) {
+ if ( AUDIO_FORMAT_PCM_FLOAT == t->mMixerFormat ) {
+ memset((char*)fout, 0, numFrames
+ * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
+ } else {
+ memset((char*)out, 0, numFrames
+ * t->mMixerChannelCount * audio_bytes_per_sample(t->mMixerFormat));
+ }
+ ALOGE_IF((((uintptr_t)in) & 3),
+ "process__oneTrack16BitsStereoNoResampling: misaligned buffer"
+ " %p track %d, channels %d, needs %08x, volume %08x vfl %f vfr %f",
+ in, name, t->channelCount, t->needs, vrl, t->mVolume[0], t->mVolume[1]);
+ return;
+ }
+ size_t outFrames = b.frameCount;
+
+ switch (t->mMixerFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ int32_t l = mulRL(1, rl, vrl);
+ int32_t r = mulRL(0, rl, vrl);
+ *fout++ = float_from_q4_27(l);
+ *fout++ = float_from_q4_27(r);
+ // Note: In case of later int16_t sink output,
+ // conversion and clamping is done by memcpy_to_i16_from_float().
+ } while (--outFrames);
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ if (CC_UNLIKELY(uint32_t(vl) > UNITY_GAIN_INT || uint32_t(vr) > UNITY_GAIN_INT)) {
+ // volume is boosted, so we might need to clamp even though
+ // we process only one track.
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ int32_t l = mulRL(1, rl, vrl) >> 12;
+ int32_t r = mulRL(0, rl, vrl) >> 12;
+ // clamping...
+ l = clamp16(l);
+ r = clamp16(r);
+ *out++ = (r<<16) | (l & 0xFFFF);
+ } while (--outFrames);
+ } else {
+ do {
+ uint32_t rl = *reinterpret_cast<const uint32_t *>(in);
+ in += 2;
+ int32_t l = mulRL(1, rl, vrl) >> 12;
+ int32_t r = mulRL(0, rl, vrl) >> 12;
+ *out++ = (r<<16) | (l & 0xFFFF);
+ } while (--outFrames);
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixer format: %d", t->mMixerFormat);
+ }
+ numFrames -= b.frameCount;
+ t->bufferProvider->releaseBuffer(&b);
+ }
+}
+
+/* TODO: consider whether this level of optimization is necessary.
+ * Perhaps just stick with a single for loop.
+ */
+
+// Needs to derive a compile time constant (constexpr). Could be targeted to go
+// to a MONOVOL mixtype based on MAX_NUM_VOLUMES, but that's an unnecessary complication.
+#define MIXTYPE_MONOVOL(mixtype) ((mixtype) == MIXTYPE_MULTI ? MIXTYPE_MULTI_MONOVOL : \
+ (mixtype) == MIXTYPE_MULTI_SAVEONLY ? MIXTYPE_MULTI_SAVEONLY_MONOVOL : (mixtype))
+
+/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE,
+ typename TO, typename TI, typename TV, typename TA, typename TAV>
+static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
+ const TI* in, TA* aux, TV *vol, const TV *volinc, TAV *vola, TAV volainc)
+{
+ switch (channels) {
+ case 1:
+ volumeRampMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 2:
+ volumeRampMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 3:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 4:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 5:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 6:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 7:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ case 8:
+ volumeRampMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out,
+ frameCount, in, aux, vol, volinc, vola, volainc);
+ break;
+ }
+}
+
+/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE,
+ typename TO, typename TI, typename TV, typename TA, typename TAV>
+static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
+ const TI* in, TA* aux, const TV *vol, TAV vola)
+{
+ switch (channels) {
+ case 1:
+ volumeMulti<MIXTYPE, 1>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 2:
+ volumeMulti<MIXTYPE, 2>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 3:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 3>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 4:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 4>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 5:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 5>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 6:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 6>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 7:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 7>(out, frameCount, in, aux, vol, vola);
+ break;
+ case 8:
+ volumeMulti<MIXTYPE_MONOVOL(MIXTYPE), 8>(out, frameCount, in, aux, vol, vola);
+ break;
+ }
+}
+
+/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * USEFLOATVOL (set to true if float volume is used)
+ * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+ typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::volumeMix(TO *out, size_t outFrames,
+ const TI *in, TA *aux, bool ramp)
+{
+ if (USEFLOATVOL) {
+ if (ramp) {
+ volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+ mPrevVolume, mVolumeInc,
+#ifdef FLOAT_AUX
+ &mPrevAuxLevel, mAuxInc
+#else
+ &prevAuxLevel, auxInc
+#endif
+ );
+ if (ADJUSTVOL) {
+ adjustVolumeRamp(aux != NULL, true);
+ }
+ } else {
+ volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+ mVolume,
+#ifdef FLOAT_AUX
+ mAuxLevel
+#else
+ auxLevel
+#endif
+ );
+ }
+ } else {
+ if (ramp) {
+ volumeRampMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+ prevVolume, volumeInc, &prevAuxLevel, auxInc);
+ if (ADJUSTVOL) {
+ adjustVolumeRamp(aux != NULL);
+ }
+ } else {
+ volumeMulti<MIXTYPE>(mMixerChannelCount, out, outFrames, in, aux,
+ volume, auxLevel);
+ }
+ }
+}
+
+/* This process hook is called when there is a single track without
+ * aux buffer, volume ramp, or resampling.
+ * TODO: Update the hook selection: this can properly handle aux and ramp.
+ *
+ * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27)
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::process__noResampleOneTrack()
+{
+ ALOGVV("process__noResampleOneTrack\n");
+ LOG_ALWAYS_FATAL_IF(mEnabled.size() != 1,
+ "%zu != 1 tracks enabled", mEnabled.size());
+ const std::shared_ptr<TrackBase> &t = mTracks[mEnabled[0]];
+ const uint32_t channels = t->mMixerChannelCount;
+ TO* out = reinterpret_cast<TO*>(t->mainBuffer);
+ TA* aux = reinterpret_cast<TA*>(t->auxBuffer);
+ const bool ramp = t->needsRamp();
+
+ for (size_t numFrames = mFrameCount; numFrames > 0; ) {
+ AudioBufferProvider::Buffer& b(t->buffer);
+ // get input buffer
+ b.frameCount = numFrames;
+ t->bufferProvider->getNextBuffer(&b);
+ const TI *in = reinterpret_cast<TI*>(b.raw);
+
+ // in == NULL can happen if the track was flushed just after having
+ // been enabled for mixing.
+ if (in == NULL || (((uintptr_t)in) & 3)) {
+ memset(out, 0, numFrames
+ * channels * audio_bytes_per_sample(t->mMixerFormat));
+ ALOGE_IF((((uintptr_t)in) & 3), "process__noResampleOneTrack: bus error: "
+ "buffer %p track %p, channels %d, needs %#x",
+ in, &t, t->channelCount, t->needs);
+ return;
+ }
+
+ const size_t outFrames = b.frameCount;
+ t->volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, false /* ADJUSTVOL */> (
+ out, outFrames, in, aux, ramp);
+
+ out += outFrames * channels;
+ if (aux != NULL) {
+ aux += outFrames;
+ }
+ numFrames -= b.frameCount;
+
+ // release buffer
+ t->bufferProvider->releaseBuffer(&b);
+ }
+ if (ramp) {
+ t->adjustVolumeRamp(aux != NULL, is_same<TI, float>::value);
+ }
+}
+
+/* This track hook is called to do resampling then mixing,
+ * pulling from the track's upstream AudioBufferProvider.
+ *
+ * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::track__Resample(TO* out, size_t outFrameCount, TO* temp, TA* aux)
+{
+ ALOGVV("track__Resample\n");
+ mResampler->setSampleRate(sampleRate);
+ const bool ramp = needsRamp();
+ if (ramp || aux != NULL) {
+ // if ramp: resample with unity gain to temp buffer and scale/mix in 2nd step.
+ // if aux != NULL: resample with unity gain to temp buffer then apply send level.
+
+ mResampler->setVolume(UNITY_GAIN_FLOAT, UNITY_GAIN_FLOAT);
+ memset(temp, 0, outFrameCount * mMixerChannelCount * sizeof(TO));
+ mResampler->resample((int32_t*)temp, outFrameCount, bufferProvider);
+
+ volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
+ out, outFrameCount, temp, aux, ramp);
+
+ } else { // constant volume gain
+ mResampler->setVolume(mVolume[0], mVolume[1]);
+ mResampler->resample((int32_t*)out, outFrameCount, bufferProvider);
+ }
+}
+
+/* This track hook is called to mix a track, when no resampling is required.
+ * The input buffer should be present in in.
+ *
+ * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27) or float
+ */
+template <int MIXTYPE, typename TO, typename TI, typename TA>
+void AudioMixerBase::TrackBase::track__NoResample(
+ TO* out, size_t frameCount, TO* temp __unused, TA* aux)
+{
+ ALOGVV("track__NoResample\n");
+ const TI *in = static_cast<const TI *>(mIn);
+
+ volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL */, true /* ADJUSTVOL */>(
+ out, frameCount, in, aux, needsRamp());
+
+ // MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
+ // MIXTYPE_MULTI reads NCHAN input channels and places to NCHAN output channels.
+ in += (MIXTYPE == MIXTYPE_MONOEXPAND) ? frameCount : frameCount * mMixerChannelCount;
+ mIn = in;
+}
+
+/* The Mixer engine generates either int32_t (Q4_27) or float data.
+ * We use this function to convert the engine buffers
+ * to the desired mixer output format, either int16_t (Q.15) or float.
+ */
+/* static */
+void AudioMixerBase::convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+ void *in, audio_format_t mixerInFormat, size_t sampleCount)
+{
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy(out, in, sampleCount * sizeof(float)); // MEMCPY. TODO optimize out
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ memcpy_to_i16_from_float((int16_t*)out, (float*)in, sampleCount);
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ memcpy_to_float_from_q4_27((float*)out, (const int32_t*)in, sampleCount);
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ memcpy_to_i16_from_q4_27((int16_t*)out, (const int32_t*)in, sampleCount);
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+}
+
+/* Returns the proper track hook to use for mixing the track into the output buffer.
+ */
+/* static */
+AudioMixerBase::hook_t AudioMixerBase::TrackBase::getTrackHook(int trackType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat __unused)
+{
+ if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+ switch (trackType) {
+ case TRACKTYPE_NOP:
+ return &TrackBase::track__nop;
+ case TRACKTYPE_RESAMPLE:
+ return &TrackBase::track__genericResample;
+ case TRACKTYPE_NORESAMPLEMONO:
+ return &TrackBase::track__16BitsMono;
+ case TRACKTYPE_NORESAMPLE:
+ return &TrackBase::track__16BitsStereo;
+ default:
+ LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+ break;
+ }
+ }
+ LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
+ switch (trackType) {
+ case TRACKTYPE_NOP:
+ return &TrackBase::track__nop;
+ case TRACKTYPE_RESAMPLE:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
+ MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__Resample<
+ MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ case TRACKTYPE_NORESAMPLEMONO:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+ MIXTYPE_MONOEXPAND, float /*TO*/, float /*TI*/, TYPE_AUX>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+ MIXTYPE_MONOEXPAND, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ case TRACKTYPE_NORESAMPLE:
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+ MIXTYPE_MULTI, float /*TO*/, float /*TI*/, TYPE_AUX>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return (AudioMixerBase::hook_t) &TrackBase::track__NoResample<
+ MIXTYPE_MULTI, int32_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad trackType: %d", trackType);
+ break;
+ }
+ return NULL;
+}
+
+/* Returns the proper process hook for mixing tracks. Currently works only for
+ * PROCESSTYPE_NORESAMPLEONETRACK, a mix involving one track, no resampling.
+ *
+ * TODO: Due to the special mixing considerations of duplicating to
+ * a stereo output track, the input track cannot be MONO. This should be
+ * prevented by the caller.
+ */
+/* static */
+AudioMixerBase::process_hook_t AudioMixerBase::getProcessHook(
+ int processType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat)
+{
+ if (processType != PROCESSTYPE_NORESAMPLEONETRACK) { // Only NORESAMPLEONETRACK
+ LOG_ALWAYS_FATAL("bad processType: %d", processType);
+ return NULL;
+ }
+ if (!kUseNewMixer && channelCount == FCC_2 && mixerInFormat == AUDIO_FORMAT_PCM_16_BIT) {
+ return &AudioMixerBase::process__oneTrack16BitsStereoNoResampling;
+ }
+ LOG_ALWAYS_FATAL_IF(channelCount > MAX_NUM_CHANNELS);
+ switch (mixerInFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return &AudioMixerBase::process__noResampleOneTrack<
+ MIXTYPE_MULTI_SAVEONLY, float /*TO*/, float /*TI*/, TYPE_AUX>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return &AudioMixerBase::process__noResampleOneTrack<
+ MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, float /*TI*/, TYPE_AUX>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ switch (mixerOutFormat) {
+ case AUDIO_FORMAT_PCM_FLOAT:
+ return &AudioMixerBase::process__noResampleOneTrack<
+ MIXTYPE_MULTI_SAVEONLY, float /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+ case AUDIO_FORMAT_PCM_16_BIT:
+ return &AudioMixerBase::process__noResampleOneTrack<
+ MIXTYPE_MULTI_SAVEONLY, int16_t /*TO*/, int16_t /*TI*/, TYPE_AUX>;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
+ break;
+ }
+ break;
+ default:
+ LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
+ break;
+ }
+ return NULL;
+}
+
+// ----------------------------------------------------------------------------
+} // namespace android
diff --git a/media/libaudioprocessing/include/media/AudioMixerBase.h b/media/libaudioprocessing/include/media/AudioMixerBase.h
new file mode 100644
index 0000000..805b6d0
--- /dev/null
+++ b/media/libaudioprocessing/include/media/AudioMixerBase.h
@@ -0,0 +1,359 @@
+/*
+**
+** Copyright 2019, The Android Open Source Project
+**
+** Licensed under the Apache License, Version 2.0 (the "License");
+** you may not use this file except in compliance with the License.
+** You may obtain a copy of the License at
+**
+** http://www.apache.org/licenses/LICENSE-2.0
+**
+** Unless required by applicable law or agreed to in writing, software
+** distributed under the License is distributed on an "AS IS" BASIS,
+** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+** See the License for the specific language governing permissions and
+** limitations under the License.
+*/
+
+#ifndef ANDROID_AUDIO_MIXER_BASE_H
+#define ANDROID_AUDIO_MIXER_BASE_H
+
+#include <map>
+#include <memory>
+#include <string>
+#include <unordered_map>
+#include <vector>
+
+#include <media/AudioBufferProvider.h>
+#include <media/AudioResampler.h>
+#include <media/AudioResamplerPublic.h>
+#include <system/audio.h>
+#include <utils/Compat.h>
+
+// This must match frameworks/av/services/audioflinger/Configuration.h
+// when used with the Audio Framework.
+#define FLOAT_AUX
+
+namespace android {
+
+// ----------------------------------------------------------------------------
+
+// AudioMixerBase is functional on its own if only mixing and resampling
+// is needed.
+
+class AudioMixerBase
+{
+public:
+ // Do not change these unless underlying code changes.
+ // This mixer has a hard-coded upper limit of 8 channels for output.
+ static constexpr uint32_t MAX_NUM_CHANNELS = FCC_8;
+ static constexpr uint32_t MAX_NUM_VOLUMES = FCC_2; // stereo volume only
+
+ static const uint16_t UNITY_GAIN_INT = 0x1000;
+ static const CONSTEXPR float UNITY_GAIN_FLOAT = 1.0f;
+
+ enum { // names
+ // setParameter targets
+ TRACK = 0x3000,
+ RESAMPLE = 0x3001,
+ RAMP_VOLUME = 0x3002, // ramp to new volume
+ VOLUME = 0x3003, // don't ramp
+ TIMESTRETCH = 0x3004,
+
+ // set Parameter names
+ // for target TRACK
+ CHANNEL_MASK = 0x4000,
+ FORMAT = 0x4001,
+ MAIN_BUFFER = 0x4002,
+ AUX_BUFFER = 0x4003,
+ // 0x4004 reserved
+ MIXER_FORMAT = 0x4005, // AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ MIXER_CHANNEL_MASK = 0x4006, // Channel mask for mixer output
+ // for target RESAMPLE
+ SAMPLE_RATE = 0x4100, // Configure sample rate conversion on this track name;
+ // parameter 'value' is the new sample rate in Hz.
+ // Only creates a sample rate converter the first time that
+ // the track sample rate is different from the mix sample rate.
+ // If the new sample rate is the same as the mix sample rate,
+ // and a sample rate converter already exists,
+ // then the sample rate converter remains present but is a no-op.
+ RESET = 0x4101, // Reset sample rate converter without changing sample rate.
+ // This clears out the resampler's input buffer.
+ REMOVE = 0x4102, // Remove the sample rate converter on this track name;
+ // the track is restored to the mix sample rate.
+ // for target RAMP_VOLUME and VOLUME (8 channels max)
+ // FIXME use float for these 3 to improve the dynamic range
+ VOLUME0 = 0x4200,
+ VOLUME1 = 0x4201,
+ AUXLEVEL = 0x4210,
+ };
+
+ AudioMixerBase(size_t frameCount, uint32_t sampleRate)
+ : mSampleRate(sampleRate)
+ , mFrameCount(frameCount) {
+ }
+
+ virtual ~AudioMixerBase() {}
+
+ virtual bool isValidFormat(audio_format_t format) const;
+ virtual bool isValidChannelMask(audio_channel_mask_t channelMask) const;
+
+ // Create a new track in the mixer.
+ //
+ // \param name a unique user-provided integer associated with the track.
+ // If name already exists, the function will abort.
+ // \param channelMask output channel mask.
+ // \param format PCM format
+ // \param sessionId Session id for the track. Tracks with the same
+ // session id will be submixed together.
+ //
+ // \return OK on success.
+ // BAD_VALUE if the format does not satisfy isValidFormat()
+ // or the channelMask does not satisfy isValidChannelMask().
+ status_t create(
+ int name, audio_channel_mask_t channelMask, audio_format_t format, int sessionId);
+
+ bool exists(int name) const {
+ return mTracks.count(name) > 0;
+ }
+
+ // Free an allocated track by name.
+ void destroy(int name);
+
+ // Enable or disable an allocated track by name
+ void enable(int name);
+ void disable(int name);
+
+ virtual void setParameter(int name, int target, int param, void *value);
+
+ void process() {
+ preProcess();
+ (this->*mHook)();
+ postProcess();
+ }
+
+ size_t getUnreleasedFrames(int name) const;
+
+ std::string trackNames() const;
+
+ protected:
+ // Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
+ // original code will be used for stereo sinks, the new mixer for everything else.
+ static constexpr bool kUseNewMixer = true;
+
+ // Set kUseFloat to true to allow floating input into the mixer engine.
+ // If kUseNewMixer is false, this is ignored or may be overridden internally
+ static constexpr bool kUseFloat = true;
+
+#ifdef FLOAT_AUX
+ using TYPE_AUX = float;
+ static_assert(kUseNewMixer && kUseFloat,
+ "kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
+#else
+ using TYPE_AUX = int32_t; // q4.27
+#endif
+
+ /* For multi-format functions (calls template functions
+ * in AudioMixerOps.h). The template parameters are as follows:
+ *
+ * MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
+ * USEFLOATVOL (set to true if float volume is used)
+ * ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
+ * TO: int32_t (Q4.27) or float
+ * TI: int32_t (Q4.27) or int16_t (Q0.15) or float
+ * TA: int32_t (Q4.27)
+ */
+
+ enum {
+ // FIXME this representation permits up to 8 channels
+ NEEDS_CHANNEL_COUNT__MASK = 0x00000007,
+ };
+
+ enum {
+ NEEDS_CHANNEL_1 = 0x00000000, // mono
+ NEEDS_CHANNEL_2 = 0x00000001, // stereo
+
+ // sample format is not explicitly specified, and is assumed to be AUDIO_FORMAT_PCM_16_BIT
+
+ NEEDS_MUTE = 0x00000100,
+ NEEDS_RESAMPLE = 0x00001000,
+ NEEDS_AUX = 0x00010000,
+ };
+
+ // hook types
+ enum {
+ PROCESSTYPE_NORESAMPLEONETRACK, // others set elsewhere
+ };
+
+ enum {
+ TRACKTYPE_NOP,
+ TRACKTYPE_RESAMPLE,
+ TRACKTYPE_NORESAMPLE,
+ TRACKTYPE_NORESAMPLEMONO,
+ };
+
+ // process hook functionality
+ using process_hook_t = void(AudioMixerBase::*)();
+
+ struct TrackBase;
+ using hook_t = void(TrackBase::*)(
+ int32_t* output, size_t numOutFrames, int32_t* temp, int32_t* aux);
+
+ struct TrackBase {
+ TrackBase()
+ : bufferProvider(nullptr)
+ {
+ // TODO: move additional initialization here.
+ }
+ virtual ~TrackBase() {}
+
+ virtual uint32_t getOutputChannelCount() { return channelCount; }
+ virtual uint32_t getMixerChannelCount() { return mMixerChannelCount; }
+
+ bool needsRamp() { return (volumeInc[0] | volumeInc[1] | auxInc) != 0; }
+ bool setResampler(uint32_t trackSampleRate, uint32_t devSampleRate);
+ bool doesResample() const { return mResampler.get() != nullptr; }
+ void recreateResampler(uint32_t devSampleRate);
+ void resetResampler() { if (mResampler.get() != nullptr) mResampler->reset(); }
+ void adjustVolumeRamp(bool aux, bool useFloat = false);
+ size_t getUnreleasedFrames() const { return mResampler.get() != nullptr ?
+ mResampler->getUnreleasedFrames() : 0; };
+
+ static hook_t getTrackHook(int trackType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+
+ void track__nop(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+
+ template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
+ typename TO, typename TI, typename TA>
+ void volumeMix(TO *out, size_t outFrames, const TI *in, TA *aux, bool ramp);
+
+ uint32_t needs;
+
+ // TODO: Eventually remove legacy integer volume settings
+ union {
+ int16_t volume[MAX_NUM_VOLUMES]; // U4.12 fixed point (top bit should be zero)
+ int32_t volumeRL;
+ };
+
+ int32_t prevVolume[MAX_NUM_VOLUMES];
+ int32_t volumeInc[MAX_NUM_VOLUMES];
+ int32_t auxInc;
+ int32_t prevAuxLevel;
+ int16_t auxLevel; // 0 <= auxLevel <= MAX_GAIN_INT, but signed for mul performance
+
+ uint16_t frameCount;
+
+ uint8_t channelCount; // 1 or 2, redundant with (needs & NEEDS_CHANNEL_COUNT__MASK)
+ uint8_t unused_padding; // formerly format, was always 16
+ uint16_t enabled; // actually bool
+ audio_channel_mask_t channelMask;
+
+ // actual buffer provider used by the track hooks
+ AudioBufferProvider* bufferProvider;
+
+ mutable AudioBufferProvider::Buffer buffer; // 8 bytes
+
+ hook_t hook;
+ const void *mIn; // current location in buffer
+
+ std::unique_ptr<AudioResampler> mResampler;
+ uint32_t sampleRate;
+ int32_t* mainBuffer;
+ int32_t* auxBuffer;
+
+ int32_t sessionId;
+
+ audio_format_t mMixerFormat; // output mix format: AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ audio_format_t mFormat; // input track format
+ audio_format_t mMixerInFormat; // mix internal format AUDIO_FORMAT_PCM_(FLOAT|16_BIT)
+ // each track must be converted to this format.
+
+ float mVolume[MAX_NUM_VOLUMES]; // floating point set volume
+ float mPrevVolume[MAX_NUM_VOLUMES]; // floating point previous volume
+ float mVolumeInc[MAX_NUM_VOLUMES]; // floating point volume increment
+
+ float mAuxLevel; // floating point set aux level
+ float mPrevAuxLevel; // floating point prev aux level
+ float mAuxInc; // floating point aux increment
+
+ audio_channel_mask_t mMixerChannelMask;
+ uint32_t mMixerChannelCount;
+
+ protected:
+
+ // hooks
+ void track__genericResample(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+ void track__16BitsStereo(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+ void track__16BitsMono(int32_t* out, size_t numFrames, int32_t* temp, int32_t* aux);
+
+ void volumeRampStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+ void volumeStereo(int32_t* out, size_t frameCount, int32_t* temp, int32_t* aux);
+
+ // multi-format track hooks
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ void track__Resample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ void track__NoResample(TO* out, size_t frameCount, TO* temp __unused, TA* aux);
+ };
+
+ // preCreateTrack must create an instance of a proper TrackBase descendant.
+ // postCreateTrack is called after filling out fields of TrackBase. It can
+ // abort track creation by returning non-OK status. See the implementation
+ // of create() for details.
+ virtual std::shared_ptr<TrackBase> preCreateTrack();
+ virtual status_t postCreateTrack(TrackBase *track __unused) { return OK; }
+
+ // preProcess is called before the process hook, postProcess after,
+ // see the implementation of process() method.
+ virtual void preProcess() {}
+ virtual void postProcess() {}
+
+ virtual bool setChannelMasks(int name,
+ audio_channel_mask_t trackChannelMask, audio_channel_mask_t mixerChannelMask);
+
+ // Called when track info changes and a new process hook should be determined.
+ void invalidate() {
+ mHook = &AudioMixerBase::process__validate;
+ }
+
+ void process__validate();
+ void process__nop();
+ void process__genericNoResampling();
+ void process__genericResampling();
+ void process__oneTrack16BitsStereoNoResampling();
+
+ template <int MIXTYPE, typename TO, typename TI, typename TA>
+ void process__noResampleOneTrack();
+
+ static process_hook_t getProcessHook(int processType, uint32_t channelCount,
+ audio_format_t mixerInFormat, audio_format_t mixerOutFormat);
+
+ static void convertMixerFormat(void *out, audio_format_t mixerOutFormat,
+ void *in, audio_format_t mixerInFormat, size_t sampleCount);
+
+ // initialization constants
+ const uint32_t mSampleRate;
+ const size_t mFrameCount;
+
+ process_hook_t mHook = &AudioMixerBase::process__nop; // one of process__*, never nullptr
+
+ // the size of the type (int32_t) should be the largest of all types supported
+ // by the mixer.
+ std::unique_ptr<int32_t[]> mOutputTemp;
+ std::unique_ptr<int32_t[]> mResampleTemp;
+
+ // track names grouped by main buffer, in no particular order of main buffer.
+ // however names for a particular main buffer are in order (by construction).
+ std::unordered_map<void * /* mainBuffer */, std::vector<int /* name */>> mGroups;
+
+ // track names that are enabled, in increasing order (by construction).
+ std::vector<int /* name */> mEnabled;
+
+ // track smart pointers, by name, in increasing order of name.
+ std::map<int /* name */, std::shared_ptr<TrackBase>> mTracks;
+};
+
+} // namespace android
+
+#endif // ANDROID_AUDIO_MIXER_BASE_H
diff --git a/media/libmedia/include/media/RecordBufferConverter.h b/media/libaudioprocessing/include/media/RecordBufferConverter.h
similarity index 100%
rename from media/libmedia/include/media/RecordBufferConverter.h
rename to media/libaudioprocessing/include/media/RecordBufferConverter.h
diff --git a/media/libaudioprocessing/tests/Android.bp b/media/libaudioprocessing/tests/Android.bp
index d990111..20c2c2c 100644
--- a/media/libaudioprocessing/tests/Android.bp
+++ b/media/libaudioprocessing/tests/Android.bp
@@ -3,8 +3,13 @@
cc_defaults {
name: "libaudioprocessing_test_defaults",
- header_libs: ["libbase_headers"],
+ header_libs: [
+ "libbase_headers",
+ "libmedia_headers",
+ ],
+
shared_libs: [
+ "libaudioclient",
"libaudioprocessing",
"libaudioutils",
"libcutils",
diff --git a/media/libaudioprocessing/tests/fuzzer/Android.bp b/media/libaudioprocessing/tests/fuzzer/Android.bp
new file mode 100644
index 0000000..1df47b7
--- /dev/null
+++ b/media/libaudioprocessing/tests/fuzzer/Android.bp
@@ -0,0 +1,10 @@
+cc_fuzz {
+ name: "libaudioprocessing_resampler_fuzzer",
+ srcs: [
+ "libaudioprocessing_resampler_fuzzer.cpp",
+ ],
+ defaults: ["libaudioprocessing_test_defaults"],
+ static_libs: [
+ "libsndfile",
+ ],
+}
diff --git a/media/libaudioprocessing/tests/fuzzer/libaudioprocessing_resampler_fuzzer.cpp b/media/libaudioprocessing/tests/fuzzer/libaudioprocessing_resampler_fuzzer.cpp
new file mode 100644
index 0000000..938c610
--- /dev/null
+++ b/media/libaudioprocessing/tests/fuzzer/libaudioprocessing_resampler_fuzzer.cpp
@@ -0,0 +1,188 @@
+/*
+ * Copyright (C) 2019 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#include <android-base/macros.h>
+#include <audio_utils/primitives.h>
+#include <audio_utils/sndfile.h>
+#include <errno.h>
+#include <fcntl.h>
+#include <inttypes.h>
+#include <math.h>
+#include <media/AudioBufferProvider.h>
+#include <media/AudioResampler.h>
+#include <stddef.h>
+#include <stdint.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <sys/mman.h>
+#include <sys/stat.h>
+#include <time.h>
+#include <unistd.h>
+#include <utils/Vector.h>
+
+#include <memory>
+
+using namespace android;
+
+const int MAX_FRAMES = 10;
+const int MIN_FREQ = 1e3;
+const int MAX_FREQ = 100e3;
+
+const AudioResampler::src_quality qualities[] = {
+ AudioResampler::DEFAULT_QUALITY,
+ AudioResampler::LOW_QUALITY,
+ AudioResampler::MED_QUALITY,
+ AudioResampler::HIGH_QUALITY,
+ AudioResampler::VERY_HIGH_QUALITY,
+ AudioResampler::DYN_LOW_QUALITY,
+ AudioResampler::DYN_MED_QUALITY,
+ AudioResampler::DYN_HIGH_QUALITY,
+};
+
+class Provider : public AudioBufferProvider {
+ const void* mAddr; // base address
+ const size_t mNumFrames; // total frames
+ const size_t mFrameSize; // size of each frame in bytes
+ size_t mNextFrame; // index of next frame to provide
+ size_t mUnrel; // number of frames not yet released
+ public:
+ Provider(const void* addr, size_t frames, size_t frameSize)
+ : mAddr(addr),
+ mNumFrames(frames),
+ mFrameSize(frameSize),
+ mNextFrame(0),
+ mUnrel(0) {}
+ status_t getNextBuffer(Buffer* buffer) override {
+ if (buffer->frameCount > mNumFrames - mNextFrame) {
+ buffer->frameCount = mNumFrames - mNextFrame;
+ }
+ mUnrel = buffer->frameCount;
+ if (buffer->frameCount > 0) {
+ buffer->raw = (char*)mAddr + mFrameSize * mNextFrame;
+ return NO_ERROR;
+ } else {
+ buffer->raw = nullptr;
+ return NOT_ENOUGH_DATA;
+ }
+ }
+ virtual void releaseBuffer(Buffer* buffer) {
+ if (buffer->frameCount > mUnrel) {
+ mNextFrame += mUnrel;
+ mUnrel = 0;
+ } else {
+ mNextFrame += buffer->frameCount;
+ mUnrel -= buffer->frameCount;
+ }
+ buffer->frameCount = 0;
+ buffer->raw = nullptr;
+ }
+ void reset() { mNextFrame = 0; }
+};
+
+audio_format_t chooseFormat(AudioResampler::src_quality quality,
+ uint8_t input_byte) {
+ switch (quality) {
+ case AudioResampler::DYN_LOW_QUALITY:
+ case AudioResampler::DYN_MED_QUALITY:
+ case AudioResampler::DYN_HIGH_QUALITY:
+ if (input_byte % 2) {
+ return AUDIO_FORMAT_PCM_FLOAT;
+ }
+ FALLTHROUGH_INTENDED;
+ default:
+ return AUDIO_FORMAT_PCM_16_BIT;
+ }
+}
+
+int parseValue(const uint8_t* src, int index, void* dst, size_t size) {
+ memcpy(dst, &src[index], size);
+ return size;
+}
+
+bool validFreq(int freq) { return freq > MIN_FREQ && freq < MAX_FREQ; }
+
+extern "C" int LLVMFuzzerTestOneInput(const uint8_t* data, size_t size) {
+ int input_freq = 0;
+ int output_freq = 0;
+ int input_channels = 0;
+
+ float left_volume = 0;
+ float right_volume = 0;
+
+ size_t metadata_size = 2 + 3 * sizeof(int) + 2 * sizeof(float);
+ if (size < metadata_size) {
+ // not enough data to set options
+ return 0;
+ }
+
+ AudioResampler::src_quality quality = qualities[data[0] % 8];
+ audio_format_t format = chooseFormat(quality, data[1]);
+
+ int index = 2;
+
+ index += parseValue(data, index, &input_freq, sizeof(int));
+ index += parseValue(data, index, &output_freq, sizeof(int));
+ index += parseValue(data, index, &input_channels, sizeof(int));
+
+ index += parseValue(data, index, &left_volume, sizeof(float));
+ index += parseValue(data, index, &right_volume, sizeof(float));
+
+ if (!validFreq(input_freq) || !validFreq(output_freq)) {
+ // sampling frequencies must be reasonable
+ return 0;
+ }
+
+ if (input_channels < 1 ||
+ input_channels > (quality < AudioResampler::DYN_LOW_QUALITY ? 2 : 8)) {
+ // invalid number of input channels
+ return 0;
+ }
+
+ size_t single_channel_size =
+ format == AUDIO_FORMAT_PCM_FLOAT ? sizeof(float) : sizeof(int16_t);
+ size_t input_frame_size = single_channel_size * input_channels;
+ size_t input_size = size - metadata_size;
+ uint8_t input_data[input_size];
+ memcpy(input_data, &data[metadata_size], input_size);
+
+ size_t input_frames = input_size / input_frame_size;
+ if (input_frames > MAX_FRAMES) {
+ return 0;
+ }
+
+ Provider provider(input_data, input_frames, input_frame_size);
+
+ std::unique_ptr<AudioResampler> resampler(
+ AudioResampler::create(format, input_channels, output_freq, quality));
+
+ resampler->setSampleRate(input_freq);
+ resampler->setVolume(left_volume, right_volume);
+
+ // output is at least stereo samples
+ int output_channels = input_channels > 2 ? input_channels : 2;
+ size_t output_frame_size = output_channels * sizeof(int32_t);
+ size_t output_frames = (input_frames * output_freq) / input_freq;
+ size_t output_size = output_frames * output_frame_size;
+
+ uint8_t output_data[output_size];
+ for (size_t i = 0; i < output_frames; i++) {
+ memset(output_data, 0, output_size);
+ resampler->resample((int*)output_data, i, &provider);
+ }
+
+ return 0;
+}
diff --git a/media/libcpustats/Android.bp b/media/libcpustats/Android.bp
index 8fcd8a4..6e8ca1d 100644
--- a/media/libcpustats/Android.bp
+++ b/media/libcpustats/Android.bp
@@ -6,6 +6,14 @@
"ThreadCpuUsage.cpp",
],
+ local_include_dirs: [
+ "include",
+ ],
+
+ export_include_dirs: [
+ "include",
+ ],
+
cflags: [
"-Werror",
"-Wall",
diff --git a/media/libdatasource/Android.bp b/media/libdatasource/Android.bp
new file mode 100644
index 0000000..f191c21
--- /dev/null
+++ b/media/libdatasource/Android.bp
@@ -0,0 +1,63 @@
+cc_library {
+ name: "libdatasource",
+
+ srcs: [
+ "DataSourceFactory.cpp",
+ "DataURISource.cpp",
+ "FileSource.cpp",
+ "HTTPBase.cpp",
+ "MediaHTTP.cpp",
+ "NuCachedSource2.cpp",
+ ],
+
+ aidl: {
+ local_include_dirs: ["aidl"],
+ export_aidl_headers: true,
+ },
+
+ header_libs: [
+ "libstagefright_headers",
+ "media_ndk_headers",
+ "libmedia_headers",
+ ],
+
+ export_header_lib_headers: [
+ "libstagefright_headers",
+ "media_ndk_headers",
+ ],
+
+ shared_libs: [
+ "liblog",
+ "libcutils",
+ "libutils",
+ "libstagefright_foundation",
+ "libdl",
+ ],
+
+ static_libs: [
+ "libc_malloc_debug_backtrace", // for memory heap analysis
+ "libmedia_midiiowrapper",
+ ],
+
+ local_include_dirs: [
+ "include",
+ ],
+
+ export_include_dirs: [
+ "include",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wno-error=deprecated-declarations",
+ "-Wall",
+ ],
+
+ sanitize: {
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ cfi: true,
+ },
+}
diff --git a/media/libstagefright/DataSourceFactory.cpp b/media/libdatasource/DataSourceFactory.cpp
similarity index 72%
rename from media/libstagefright/DataSourceFactory.cpp
rename to media/libdatasource/DataSourceFactory.cpp
index 54bf0cc..bb6a08c 100644
--- a/media/libstagefright/DataSourceFactory.cpp
+++ b/media/libdatasource/DataSourceFactory.cpp
@@ -16,20 +16,33 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "DataSource"
-#include "include/HTTPBase.h"
-#include "include/NuCachedSource2.h"
+#include <datasource/DataSourceFactory.h>
+#include <datasource/DataURISource.h>
+#include <datasource/HTTPBase.h>
+#include <datasource/FileSource.h>
+#include <datasource/MediaHTTP.h>
+#include <datasource/NuCachedSource2.h>
#include <media/MediaHTTPConnection.h>
#include <media/MediaHTTPService.h>
-#include <media/stagefright/DataSourceFactory.h>
-#include <media/stagefright/DataURISource.h>
-#include <media/stagefright/FileSource.h>
-#include <media/stagefright/MediaHTTP.h>
#include <utils/String8.h>
namespace android {
// static
+sp<DataSourceFactory> DataSourceFactory::sInstance;
+// static
+Mutex DataSourceFactory::sInstanceLock;
+
+// static
+sp<DataSourceFactory> DataSourceFactory::getInstance() {
+ Mutex::Autolock l(sInstanceLock);
+ if (!sInstance) {
+ sInstance = new DataSourceFactory();
+ }
+ return sInstance;
+}
+
sp<DataSource> DataSourceFactory::CreateFromURI(
const sp<MediaHTTPService> &httpService,
const char *uri,
@@ -42,20 +55,16 @@
sp<DataSource> source;
if (!strncasecmp("file://", uri, 7)) {
- source = new FileSource(uri + 7);
+ source = CreateFileSource(uri + 7);
} else if (!strncasecmp("http://", uri, 7) || !strncasecmp("https://", uri, 8)) {
if (httpService == NULL) {
ALOGE("Invalid http service!");
return NULL;
}
- if (httpSource == NULL) {
- sp<MediaHTTPConnection> conn = httpService->makeHTTPConnection();
- if (conn == NULL) {
- ALOGE("Failed to make http connection from http service!");
- return NULL;
- }
- httpSource = new MediaHTTP(conn);
+ sp<HTTPBase> mediaHTTP = httpSource;
+ if (mediaHTTP == NULL) {
+ mediaHTTP = static_cast<HTTPBase *>(CreateMediaHTTP(httpService).get());
}
String8 cacheConfig;
@@ -69,24 +78,24 @@
&disconnectAtHighwatermark);
}
- if (httpSource->connect(uri, &nonCacheSpecificHeaders) != OK) {
+ if (mediaHTTP->connect(uri, &nonCacheSpecificHeaders) != OK) {
ALOGE("Failed to connect http source!");
return NULL;
}
if (contentType != NULL) {
- *contentType = httpSource->getMIMEType();
+ *contentType = mediaHTTP->getMIMEType();
}
source = NuCachedSource2::Create(
- httpSource,
+ mediaHTTP,
cacheConfig.isEmpty() ? NULL : cacheConfig.string(),
disconnectAtHighwatermark);
} else if (!strncasecmp("data:", uri, 5)) {
source = DataURISource::Create(uri);
} else {
// Assume it's a filename.
- source = new FileSource(uri);
+ source = CreateFileSource(uri);
}
if (source == NULL || source->initCheck() != OK) {
@@ -108,10 +117,15 @@
sp<MediaHTTPConnection> conn = httpService->makeHTTPConnection();
if (conn == NULL) {
+ ALOGE("Failed to make http connection from http service!");
return NULL;
} else {
return new MediaHTTP(conn);
}
}
+sp<DataSource> DataSourceFactory::CreateFileSource(const char *uri) {
+ return new FileSource(uri);
+}
+
} // namespace android
diff --git a/media/libstagefright/DataURISource.cpp b/media/libdatasource/DataURISource.cpp
similarity index 98%
rename from media/libstagefright/DataURISource.cpp
rename to media/libdatasource/DataURISource.cpp
index b975b38..216f3d0 100644
--- a/media/libstagefright/DataURISource.cpp
+++ b/media/libdatasource/DataURISource.cpp
@@ -13,7 +13,7 @@
* See the License for the specific language governing permissions and
* limitations under the License.
*/
-#include <media/stagefright/DataURISource.h>
+#include <datasource/DataURISource.h>
#include <media/stagefright/foundation/ABuffer.h>
#include <media/stagefright/foundation/AString.h>
diff --git a/media/libstagefright/ClearFileSource.cpp b/media/libdatasource/FileSource.cpp
similarity index 85%
rename from media/libstagefright/ClearFileSource.cpp
rename to media/libdatasource/FileSource.cpp
index e3a2cb7..bbf7dda 100644
--- a/media/libstagefright/ClearFileSource.cpp
+++ b/media/libdatasource/FileSource.cpp
@@ -15,12 +15,12 @@
*/
//#define LOG_NDEBUG 0
-#define LOG_TAG "ClearFileSource"
+#define LOG_TAG "FileSource"
#include <utils/Log.h>
+#include <datasource/FileSource.h>
#include <media/stagefright/foundation/ADebug.h>
-#include <media/stagefright/ClearFileSource.h>
-#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <sys/types.h>
#include <unistd.h>
#include <sys/types.h>
@@ -29,7 +29,7 @@
namespace android {
-ClearFileSource::ClearFileSource(const char *filename)
+FileSource::FileSource(const char *filename)
: mFd(-1),
mOffset(0),
mLength(-1),
@@ -48,7 +48,7 @@
}
}
-ClearFileSource::ClearFileSource(int fd, int64_t offset, int64_t length)
+FileSource::FileSource(int fd, int64_t offset, int64_t length)
: mFd(fd),
mOffset(offset),
mLength(length),
@@ -89,18 +89,18 @@
}
-ClearFileSource::~ClearFileSource() {
+FileSource::~FileSource() {
if (mFd >= 0) {
::close(mFd);
mFd = -1;
}
}
-status_t ClearFileSource::initCheck() const {
+status_t FileSource::initCheck() const {
return mFd >= 0 ? OK : NO_INIT;
}
-ssize_t ClearFileSource::readAt(off64_t offset, void *data, size_t size) {
+ssize_t FileSource::readAt(off64_t offset, void *data, size_t size) {
if (mFd < 0) {
return NO_INIT;
}
@@ -118,7 +118,7 @@
return readAt_l(offset, data, size);
}
-ssize_t ClearFileSource::readAt_l(off64_t offset, void *data, size_t size) {
+ssize_t FileSource::readAt_l(off64_t offset, void *data, size_t size) {
off64_t result = lseek64(mFd, offset + mOffset, SEEK_SET);
if (result == -1) {
ALOGE("seek to %lld failed", (long long)(offset + mOffset));
@@ -128,7 +128,7 @@
return ::read(mFd, data, size);
}
-status_t ClearFileSource::getSize(off64_t *size) {
+status_t FileSource::getSize(off64_t *size) {
Mutex::Autolock autoLock(mLock);
if (mFd < 0) {
diff --git a/media/libstagefright/HTTPBase.cpp b/media/libdatasource/HTTPBase.cpp
similarity index 98%
rename from media/libstagefright/HTTPBase.cpp
rename to media/libdatasource/HTTPBase.cpp
index d118e8c..ef29c48 100644
--- a/media/libstagefright/HTTPBase.cpp
+++ b/media/libdatasource/HTTPBase.cpp
@@ -18,7 +18,7 @@
#define LOG_TAG "HTTPBase"
#include <utils/Log.h>
-#include "include/HTTPBase.h"
+#include <datasource/HTTPBase.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
diff --git a/media/libstagefright/http/ClearMediaHTTP.cpp b/media/libdatasource/MediaHTTP.cpp
similarity index 82%
rename from media/libstagefright/http/ClearMediaHTTP.cpp
rename to media/libdatasource/MediaHTTP.cpp
index 9557c8a..58c1ce8 100644
--- a/media/libstagefright/http/ClearMediaHTTP.cpp
+++ b/media/libdatasource/MediaHTTP.cpp
@@ -15,30 +15,30 @@
*/
//#define LOG_NDEBUG 0
-#define LOG_TAG "ClearMediaHTTP"
+#define LOG_TAG "MediaHTTP"
#include <utils/Log.h>
-#include <media/stagefright/ClearMediaHTTP.h>
+#include <datasource/MediaHTTP.h>
#include <media/stagefright/foundation/ADebug.h>
#include <media/stagefright/foundation/ALooper.h>
-#include <media/stagefright/Utils.h>
+#include <media/stagefright/FoundationUtils.h>
#include <media/MediaHTTPConnection.h>
namespace android {
-ClearMediaHTTP::ClearMediaHTTP(const sp<MediaHTTPConnection> &conn)
+MediaHTTP::MediaHTTP(const sp<MediaHTTPConnection> &conn)
: mInitCheck((conn != NULL) ? OK : NO_INIT),
mHTTPConnection(conn),
mCachedSizeValid(false),
mCachedSize(0ll) {
}
-ClearMediaHTTP::~ClearMediaHTTP() {
+MediaHTTP::~MediaHTTP() {
}
-status_t ClearMediaHTTP::connect(
+status_t MediaHTTP::connect(
const char *uri,
const KeyedVector<String8, String8> *headers,
off64_t /* offset */) {
@@ -68,18 +68,18 @@
if (success) {
AString sanitized = uriDebugString(mLastURI);
- mName = String8::format("ClearMediaHTTP(%s)", sanitized.c_str());
+ mName = String8::format("MediaHTTP(%s)", sanitized.c_str());
}
return success ? OK : UNKNOWN_ERROR;
}
-void ClearMediaHTTP::close() {
+void MediaHTTP::close() {
disconnect();
}
-void ClearMediaHTTP::disconnect() {
- mName = String8("ClearMediaHTTP(<disconnected>)");
+void MediaHTTP::disconnect() {
+ mName = String8("MediaHTTP(<disconnected>)");
if (mInitCheck != OK) {
return;
}
@@ -87,11 +87,11 @@
mHTTPConnection->disconnect();
}
-status_t ClearMediaHTTP::initCheck() const {
+status_t MediaHTTP::initCheck() const {
return mInitCheck;
}
-ssize_t ClearMediaHTTP::readAt(off64_t offset, void *data, size_t size) {
+ssize_t MediaHTTP::readAt(off64_t offset, void *data, size_t size) {
if (mInitCheck != OK) {
return mInitCheck;
}
@@ -127,7 +127,7 @@
return numBytesRead;
}
-status_t ClearMediaHTTP::getSize(off64_t *size) {
+status_t MediaHTTP::getSize(off64_t *size) {
if (mInitCheck != OK) {
return mInitCheck;
}
@@ -145,16 +145,16 @@
return *size < 0 ? *size : static_cast<status_t>(OK);
}
-uint32_t ClearMediaHTTP::flags() {
+uint32_t MediaHTTP::flags() {
return kWantsPrefetching | kIsHTTPBasedSource;
}
-status_t ClearMediaHTTP::reconnectAtOffset(off64_t offset) {
+status_t MediaHTTP::reconnectAtOffset(off64_t offset) {
return connect(mLastURI.c_str(), &mLastHeaders, offset);
}
-String8 ClearMediaHTTP::getUri() {
+String8 MediaHTTP::getUri() {
if (mInitCheck != OK) {
return String8::empty();
}
@@ -166,7 +166,7 @@
return String8(mLastURI.c_str());
}
-String8 ClearMediaHTTP::getMIMEType() const {
+String8 MediaHTTP::getMIMEType() const {
if (mInitCheck != OK) {
return String8("application/octet-stream");
}
diff --git a/media/libstagefright/NuCachedSource2.cpp b/media/libdatasource/NuCachedSource2.cpp
similarity index 99%
rename from media/libstagefright/NuCachedSource2.cpp
rename to media/libdatasource/NuCachedSource2.cpp
index 522c81d..7f5ae61 100644
--- a/media/libstagefright/NuCachedSource2.cpp
+++ b/media/libdatasource/NuCachedSource2.cpp
@@ -20,8 +20,8 @@
#define LOG_TAG "NuCachedSource2"
#include <utils/Log.h>
-#include "include/NuCachedSource2.h"
-#include "include/HTTPBase.h"
+#include <datasource/NuCachedSource2.h>
+#include <datasource/HTTPBase.h>
#include <cutils/properties.h>
#include <media/stagefright/foundation/ADebug.h>
diff --git a/media/libstagefright/include/media/stagefright/DataSourceFactory.h b/media/libdatasource/include/datasource/DataSourceFactory.h
similarity index 66%
rename from media/libstagefright/include/media/stagefright/DataSourceFactory.h
rename to media/libdatasource/include/datasource/DataSourceFactory.h
index 2a1d491..194abe2 100644
--- a/media/libstagefright/include/media/stagefright/DataSourceFactory.h
+++ b/media/libdatasource/include/datasource/DataSourceFactory.h
@@ -18,7 +18,9 @@
#define DATA_SOURCE_FACTORY_H_
+#include <media/DataSource.h>
#include <sys/types.h>
+#include <utils/KeyedVector.h>
#include <utils/RefBase.h>
namespace android {
@@ -27,17 +29,27 @@
class String8;
struct HTTPBase;
-class DataSourceFactory {
+class DataSourceFactory : public RefBase {
public:
- static sp<DataSource> CreateFromURI(
+ static sp<DataSourceFactory> getInstance();
+ sp<DataSource> CreateFromURI(
const sp<MediaHTTPService> &httpService,
const char *uri,
const KeyedVector<String8, String8> *headers = NULL,
String8 *contentType = NULL,
HTTPBase *httpSource = NULL);
- static sp<DataSource> CreateMediaHTTP(const sp<MediaHTTPService> &httpService);
- static sp<DataSource> CreateFromFd(int fd, int64_t offset, int64_t length);
+ virtual sp<DataSource> CreateMediaHTTP(const sp<MediaHTTPService> &httpService);
+ sp<DataSource> CreateFromFd(int fd, int64_t offset, int64_t length);
+
+protected:
+ virtual sp<DataSource> CreateFileSource(const char *uri);
+ DataSourceFactory() {};
+ virtual ~DataSourceFactory() {};
+
+private:
+ static sp<DataSourceFactory> sInstance;
+ static Mutex sInstanceLock;
};
} // namespace android
diff --git a/media/libstagefright/include/media/stagefright/DataURISource.h b/media/libdatasource/include/datasource/DataURISource.h
similarity index 100%
rename from media/libstagefright/include/media/stagefright/DataURISource.h
rename to media/libdatasource/include/datasource/DataURISource.h
diff --git a/media/libstagefright/include/media/stagefright/ClearFileSource.h b/media/libdatasource/include/datasource/FileSource.h
similarity index 74%
rename from media/libstagefright/include/media/stagefright/ClearFileSource.h
rename to media/libdatasource/include/datasource/FileSource.h
index be83748..dee0c33 100644
--- a/media/libstagefright/include/media/stagefright/ClearFileSource.h
+++ b/media/libdatasource/include/datasource/FileSource.h
@@ -14,9 +14,9 @@
* limitations under the License.
*/
-#ifndef CLEAR_FILE_SOURCE_H_
+#ifndef FILE_SOURCE_H_
-#define CLEAR_FILE_SOURCE_H_
+#define FILE_SOURCE_H_
#include <stdio.h>
@@ -26,11 +26,11 @@
namespace android {
-class ClearFileSource : public DataSource {
+class FileSource : public DataSource {
public:
- ClearFileSource(const char *filename);
- // ClearFileSource takes ownership and will close the fd
- ClearFileSource(int fd, int64_t offset, int64_t length);
+ FileSource(const char *filename);
+ // FileSource takes ownership and will close the fd
+ FileSource(int fd, int64_t offset, int64_t length);
virtual status_t initCheck() const;
@@ -47,7 +47,7 @@
}
protected:
- virtual ~ClearFileSource();
+ virtual ~FileSource();
virtual ssize_t readAt_l(off64_t offset, void *data, size_t size);
int mFd;
@@ -58,11 +58,11 @@
private:
String8 mName;
- ClearFileSource(const ClearFileSource &);
- ClearFileSource &operator=(const ClearFileSource &);
+ FileSource(const FileSource &);
+ FileSource &operator=(const FileSource &);
};
} // namespace android
-#endif // CLEAR_FILE_SOURCE_H_
+#endif // FILE_SOURCE_H_
diff --git a/media/libstagefright/include/HTTPBase.h b/media/libdatasource/include/datasource/HTTPBase.h
similarity index 100%
rename from media/libstagefright/include/HTTPBase.h
rename to media/libdatasource/include/datasource/HTTPBase.h
diff --git a/media/libstagefright/include/media/stagefright/ClearMediaHTTP.h b/media/libdatasource/include/datasource/MediaHTTP.h
similarity index 83%
rename from media/libstagefright/include/media/stagefright/ClearMediaHTTP.h
rename to media/libdatasource/include/datasource/MediaHTTP.h
index 72907a9..a8d203b 100644
--- a/media/libstagefright/include/media/stagefright/ClearMediaHTTP.h
+++ b/media/libdatasource/include/datasource/MediaHTTP.h
@@ -14,20 +14,20 @@
* limitations under the License.
*/
-#ifndef CLEAR_MEDIA_HTTP_H_
+#ifndef MEDIA_HTTP_H_
-#define CLEAR_MEDIA_HTTP_H_
+#define MEDIA_HTTP_H_
#include <media/stagefright/foundation/AString.h>
-#include "include/HTTPBase.h"
+#include "HTTPBase.h"
namespace android {
struct MediaHTTPConnection;
-struct ClearMediaHTTP : public HTTPBase {
- ClearMediaHTTP(const sp<MediaHTTPConnection> &conn);
+struct MediaHTTP : public HTTPBase {
+ MediaHTTP(const sp<MediaHTTPConnection> &conn);
virtual status_t connect(
const char *uri,
@@ -49,7 +49,7 @@
virtual status_t reconnectAtOffset(off64_t offset);
protected:
- virtual ~ClearMediaHTTP();
+ virtual ~MediaHTTP();
virtual String8 getUri();
virtual String8 getMIMEType() const;
@@ -65,9 +65,9 @@
bool mCachedSizeValid;
off64_t mCachedSize;
- DISALLOW_EVIL_CONSTRUCTORS(ClearMediaHTTP);
+ DISALLOW_EVIL_CONSTRUCTORS(MediaHTTP);
};
} // namespace android
-#endif // CLEAR_MEDIA_HTTP_H_
+#endif // MEDIA_HTTP_H_
diff --git a/media/libstagefright/include/NuCachedSource2.h b/media/libdatasource/include/datasource/NuCachedSource2.h
similarity index 100%
rename from media/libstagefright/include/NuCachedSource2.h
rename to media/libdatasource/include/datasource/NuCachedSource2.h
diff --git a/media/libmedia/Android.bp b/media/libmedia/Android.bp
index 1d33590..b49df9e 100644
--- a/media/libmedia/Android.bp
+++ b/media/libmedia/Android.bp
@@ -1,10 +1,3 @@
-cc_defaults {
- name: "libmedia_defaults",
- include_dirs: [
- "bionic/libc/private",
- ],
-}
-
cc_library_headers {
name: "libmedia_headers",
vendor_available: true,
@@ -47,6 +40,15 @@
clang: true,
}
+filegroup {
+ name: "libmedia_omx_aidl",
+ srcs: [
+ "aidl/android/IGraphicBufferSource.aidl",
+ "aidl/android/IOMXBufferSource.aidl",
+ ],
+ path: "aidl",
+}
+
cc_library_shared {
name: "libmedia_omx",
vendor_available: true,
@@ -56,13 +58,10 @@
double_loadable: true,
srcs: [
- "aidl/android/IGraphicBufferSource.aidl",
- "aidl/android/IOMXBufferSource.aidl",
+ ":libmedia_omx_aidl",
- "IMediaCodecList.cpp",
"IOMX.cpp",
"MediaCodecBuffer.cpp",
- "MediaCodecInfo.cpp",
"OMXBuffer.cpp",
"omx/1.0/WGraphicBufferSource.cpp",
"omx/1.0/WOmxBufferSource.cpp",
@@ -74,7 +73,7 @@
local_include_dirs: ["aidl"],
export_aidl_headers: true,
},
-
+
local_include_dirs: [
"include",
],
@@ -85,7 +84,6 @@
"libbinder",
"libcutils",
"libhidlbase",
- "libhidltransport",
"liblog",
"libstagefright_foundation",
"libui",
@@ -146,7 +144,6 @@
"libcutils",
"libgui",
"libhidlbase",
- "libhidltransport",
"liblog",
"libmedia_omx",
"libstagefright_foundation",
@@ -200,6 +197,7 @@
],
header_libs: [
+ "libmedia_headers",
"media_ndk_headers",
],
@@ -218,11 +216,52 @@
},
}
+cc_library_shared {
+ name: "libmedia_codeclist",
+
+ srcs: [
+ "IMediaCodecList.cpp",
+ "MediaCodecInfo.cpp",
+ ],
+
+ local_include_dirs: [
+ "include",
+ ],
+
+ shared_libs: [
+ "android.hardware.media.omx@1.0",
+ "libbinder",
+ "liblog",
+ "libstagefright_foundation",
+ "libutils",
+ ],
+
+ include_dirs: [
+ "system/libhidl/transport/token/1.0/utils/include",
+ ],
+
+ export_include_dirs: [
+ "include",
+ ],
+
+ cflags: [
+ "-Werror",
+ "-Wno-error=deprecated-declarations",
+ "-Wall",
+ ],
+
+ sanitize: {
+ misc_undefined: [
+ "unsigned-integer-overflow",
+ "signed-integer-overflow",
+ ],
+ cfi: true,
+ },
+}
+
cc_library {
name: "libmedia",
- defaults: [ "libmedia_defaults" ],
-
srcs: [
"IDataSource.cpp",
"BufferingSettings.cpp",
@@ -247,8 +286,6 @@
"mediarecorder.cpp",
"IMediaMetadataRetriever.cpp",
"mediametadataretriever.cpp",
- "MidiDeviceInfo.cpp",
- "JetPlayer.cpp",
"MediaScanner.cpp",
"MediaScannerClient.cpp",
"CharacterEncodingDetector.cpp",
@@ -256,7 +293,6 @@
"MediaProfiles.cpp",
"MediaResource.cpp",
"MediaResourcePolicy.cpp",
- "Visualizer.cpp",
"StringArray.cpp",
"NdkMediaFormatPriv.cpp",
"NdkMediaErrorPriv.cpp",
@@ -268,6 +304,7 @@
},
header_libs: [
+ "bionic_libc_platform_headers",
"libstagefright_headers",
"media_ndk_headers",
],
@@ -291,8 +328,8 @@
"libstagefright_foundation",
"libgui",
"libdl",
- "libaudioutils",
"libaudioclient",
+ "libmedia_codeclist",
"libmedia_omx",
],
@@ -306,7 +343,6 @@
static_libs: [
"libc_malloc_debug_backtrace", // for memory heap analysis
- "libmedia_midiiowrapper",
],
export_include_dirs: [
@@ -329,66 +365,3 @@
cfi: true,
},
}
-
-cc_library_static {
- name: "libmedia_player2_util",
-
- defaults: [ "libmedia_defaults" ],
-
- srcs: [
- "AudioParameter.cpp",
- "BufferingSettings.cpp",
- "DataSourceDesc.cpp",
- "MediaCodecBuffer.cpp",
- "Metadata.cpp",
- "NdkWrapper.cpp",
- ],
-
- shared_libs: [
- "libbinder",
- "libcutils",
- "liblog",
- "libmediandk",
- "libnativewindow",
- "libmediandk_utils",
- "libstagefright_foundation",
- "libui",
- "libutils",
- ],
-
- export_shared_lib_headers: [
- "libbinder",
- "libmediandk",
- ],
-
- header_libs: [
- "media_plugin_headers",
- ],
-
- include_dirs: [
- "frameworks/av/media/ndk",
- ],
-
- static_libs: [
- "libstagefright_rtsp",
- "libstagefright_timedtext",
- ],
-
- export_include_dirs: [
- "include",
- ],
-
- cflags: [
- "-Werror",
- "-Wno-error=deprecated-declarations",
- "-Wall",
- ],
-
- sanitize: {
- misc_undefined: [
- "unsigned-integer-overflow",
- "signed-integer-overflow",
- ],
- cfi: true,
- },
-}
diff --git a/media/libmedia/AudioParameter.cpp b/media/libmedia/AudioParameter.cpp
index 1c95e27..9f34035 100644
--- a/media/libmedia/AudioParameter.cpp
+++ b/media/libmedia/AudioParameter.cpp
@@ -40,6 +40,8 @@
AUDIO_PARAMETER_KEY_AUDIO_LANGUAGE_PREFERRED;
const char * const AudioParameter::keyMonoOutput = AUDIO_PARAMETER_MONO_OUTPUT;
const char * const AudioParameter::keyStreamHwAvSync = AUDIO_PARAMETER_STREAM_HW_AV_SYNC;
+const char * const AudioParameter::keyDeviceConnect = AUDIO_PARAMETER_DEVICE_CONNECT;
+const char * const AudioParameter::keyDeviceDisconnect = AUDIO_PARAMETER_DEVICE_DISCONNECT;
const char * const AudioParameter::keyStreamConnect = AUDIO_PARAMETER_DEVICE_CONNECT;
const char * const AudioParameter::keyStreamDisconnect = AUDIO_PARAMETER_DEVICE_DISCONNECT;
const char * const AudioParameter::keyStreamSupportedFormats = AUDIO_PARAMETER_STREAM_SUP_FORMATS;
diff --git a/media/libmedia/DataSourceDesc.cpp b/media/libmedia/DataSourceDesc.cpp
deleted file mode 100644
index b7ccbce..0000000
--- a/media/libmedia/DataSourceDesc.cpp
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "DataSourceDesc"
-
-#include <media/DataSource.h>
-#include <media/DataSourceDesc.h>
-#include <media/MediaHTTPService.h>
-
-namespace android {
-
-static const int64_t kLongMax = 0x7ffffffffffffffL;
-
-DataSourceDesc::DataSourceDesc()
- : mType(TYPE_NONE),
- mFDOffset(0),
- mFDLength(kLongMax),
- mId(0),
- mStartPositionMs(0),
- mEndPositionMs(0) {
-}
-
-} // namespace android
diff --git a/media/libmedia/JetPlayer.cpp b/media/libmedia/JetPlayer.cpp
deleted file mode 100644
index 0d3c1ba..0000000
--- a/media/libmedia/JetPlayer.cpp
+++ /dev/null
@@ -1,471 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "JetPlayer-C"
-
-#include <utils/Log.h>
-#include <media/JetPlayer.h>
-
-
-namespace android
-{
-
-static const int MIX_NUM_BUFFERS = 4;
-static const S_EAS_LIB_CONFIG* pLibConfig = NULL;
-
-//-------------------------------------------------------------------------------------------------
-JetPlayer::JetPlayer(void *javaJetPlayer, int maxTracks, int trackBufferSize) :
- mEventCallback(NULL),
- mJavaJetPlayerRef(javaJetPlayer),
- mTid(-1),
- mRender(false),
- mPaused(false),
- mMaxTracks(maxTracks),
- mEasData(NULL),
- mIoWrapper(NULL),
- mTrackBufferSize(trackBufferSize)
-{
- ALOGV("JetPlayer constructor");
- mPreviousJetStatus.currentUserID = -1;
- mPreviousJetStatus.segmentRepeatCount = -1;
- mPreviousJetStatus.numQueuedSegments = -1;
- mPreviousJetStatus.paused = true;
-}
-
-//-------------------------------------------------------------------------------------------------
-JetPlayer::~JetPlayer()
-{
- ALOGV("~JetPlayer");
- release();
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::init()
-{
- //Mutex::Autolock lock(&mMutex);
-
- EAS_RESULT result;
-
- // retrieve the EAS library settings
- if (pLibConfig == NULL)
- pLibConfig = EAS_Config();
- if (pLibConfig == NULL) {
- ALOGE("JetPlayer::init(): EAS library configuration could not be retrieved, aborting.");
- return EAS_FAILURE;
- }
-
- // init the EAS library
- result = EAS_Init(&mEasData);
- if (result != EAS_SUCCESS) {
- ALOGE("JetPlayer::init(): Error initializing Sonivox EAS library, aborting.");
- mState = EAS_STATE_ERROR;
- return result;
- }
- // init the JET library with the default app event controller range
- result = JET_Init(mEasData, NULL, sizeof(S_JET_CONFIG));
- if (result != EAS_SUCCESS) {
- ALOGE("JetPlayer::init(): Error initializing JET library, aborting.");
- mState = EAS_STATE_ERROR;
- return result;
- }
-
- // create the output AudioTrack
- mAudioTrack = new AudioTrack();
- status_t status = mAudioTrack->set(AUDIO_STREAM_MUSIC, //TODO parameterize this
- pLibConfig->sampleRate,
- AUDIO_FORMAT_PCM_16_BIT,
- audio_channel_out_mask_from_count(pLibConfig->numChannels),
- (size_t) mTrackBufferSize,
- AUDIO_OUTPUT_FLAG_NONE);
- if (status != OK) {
- ALOGE("JetPlayer::init(): Error initializing JET library; AudioTrack error %d", status);
- mAudioTrack.clear();
- mState = EAS_STATE_ERROR;
- return EAS_FAILURE;
- }
-
- // create render and playback thread
- {
- Mutex::Autolock l(mMutex);
- ALOGV("JetPlayer::init(): trying to start render thread");
- mThread = new JetPlayerThread(this);
- mThread->run("jetRenderThread", ANDROID_PRIORITY_AUDIO);
- mCondition.wait(mMutex);
- }
- if (mTid > 0) {
- // render thread started, we're ready
- ALOGV("JetPlayer::init(): render thread(%d) successfully started.", mTid);
- mState = EAS_STATE_READY;
- } else {
- ALOGE("JetPlayer::init(): failed to start render thread.");
- mState = EAS_STATE_ERROR;
- return EAS_FAILURE;
- }
-
- return EAS_SUCCESS;
-}
-
-void JetPlayer::setEventCallback(jetevent_callback eventCallback)
-{
- Mutex::Autolock l(mMutex);
- mEventCallback = eventCallback;
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::release()
-{
- ALOGV("JetPlayer::release()");
- Mutex::Autolock lock(mMutex);
- mPaused = true;
- mRender = false;
- if (mEasData) {
- JET_Pause(mEasData);
- JET_CloseFile(mEasData);
- JET_Shutdown(mEasData);
- EAS_Shutdown(mEasData);
- }
- delete mIoWrapper;
- mIoWrapper = NULL;
- if (mAudioTrack != 0) {
- mAudioTrack->stop();
- mAudioTrack->flush();
- mAudioTrack.clear();
- }
- if (mAudioBuffer) {
- delete mAudioBuffer;
- mAudioBuffer = NULL;
- }
- mEasData = NULL;
-
- return EAS_SUCCESS;
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::render() {
- EAS_RESULT result = EAS_FAILURE;
- EAS_I32 count;
- int temp;
- bool audioStarted = false;
-
- ALOGV("JetPlayer::render(): entering");
-
- // allocate render buffer
- mAudioBuffer =
- new EAS_PCM[pLibConfig->mixBufferSize * pLibConfig->numChannels * MIX_NUM_BUFFERS];
-
- // signal main thread that we started
- {
- Mutex::Autolock l(mMutex);
- mTid = gettid();
- ALOGV("JetPlayer::render(): render thread(%d) signal", mTid);
- mCondition.signal();
- }
-
- while (1) {
-
- mMutex.lock(); // [[[[[[[[ LOCK ---------------------------------------
-
- if (mEasData == NULL) {
- mMutex.unlock();
- ALOGV("JetPlayer::render(): NULL EAS data, exiting render.");
- goto threadExit;
- }
-
- // nothing to render, wait for client thread to wake us up
- while (!mRender)
- {
- ALOGV("JetPlayer::render(): signal wait");
- if (audioStarted) {
- mAudioTrack->pause();
- // we have to restart the playback once we start rendering again
- audioStarted = false;
- }
- mCondition.wait(mMutex);
- ALOGV("JetPlayer::render(): signal rx'd");
- }
-
- // render midi data into the input buffer
- int num_output = 0;
- EAS_PCM* p = mAudioBuffer;
- for (int i = 0; i < MIX_NUM_BUFFERS; i++) {
- result = EAS_Render(mEasData, p, pLibConfig->mixBufferSize, &count);
- if (result != EAS_SUCCESS) {
- ALOGE("JetPlayer::render(): EAS_Render returned error %ld", result);
- }
- p += count * pLibConfig->numChannels;
- num_output += count * pLibConfig->numChannels * sizeof(EAS_PCM);
-
- // send events that were generated (if any) to the event callback
- fireEventsFromJetQueue();
- }
-
- // update playback state
- //ALOGV("JetPlayer::render(): updating state");
- JET_Status(mEasData, &mJetStatus);
- fireUpdateOnStatusChange();
- mPaused = mJetStatus.paused;
-
- mMutex.unlock(); // UNLOCK ]]]]]]]] -----------------------------------
-
- // check audio output track
- if (mAudioTrack == NULL) {
- ALOGE("JetPlayer::render(): output AudioTrack was not created");
- goto threadExit;
- }
-
- // Write data to the audio hardware
- //ALOGV("JetPlayer::render(): writing to audio output");
- if ((temp = mAudioTrack->write(mAudioBuffer, num_output)) < 0) {
- ALOGE("JetPlayer::render(): Error in writing:%d",temp);
- return temp;
- }
-
- // start audio output if necessary
- if (!audioStarted) {
- ALOGV("JetPlayer::render(): starting audio playback");
- mAudioTrack->start();
- audioStarted = true;
- }
-
- }//while (1)
-
-threadExit:
- if (mAudioTrack != NULL) {
- mAudioTrack->stop();
- mAudioTrack->flush();
- }
- delete [] mAudioBuffer;
- mAudioBuffer = NULL;
- mMutex.lock();
- mTid = -1;
- mCondition.signal();
- mMutex.unlock();
- return result;
-}
-
-
-//-------------------------------------------------------------------------------------------------
-// fire up an update if any of the status fields has changed
-// precondition: mMutex locked
-void JetPlayer::fireUpdateOnStatusChange()
-{
- if ( (mJetStatus.currentUserID != mPreviousJetStatus.currentUserID)
- ||(mJetStatus.segmentRepeatCount != mPreviousJetStatus.segmentRepeatCount) ) {
- if (mEventCallback) {
- mEventCallback(
- JetPlayer::JET_USERID_UPDATE,
- mJetStatus.currentUserID,
- mJetStatus.segmentRepeatCount,
- mJavaJetPlayerRef);
- }
- mPreviousJetStatus.currentUserID = mJetStatus.currentUserID;
- mPreviousJetStatus.segmentRepeatCount = mJetStatus.segmentRepeatCount;
- }
-
- if (mJetStatus.numQueuedSegments != mPreviousJetStatus.numQueuedSegments) {
- if (mEventCallback) {
- mEventCallback(
- JetPlayer::JET_NUMQUEUEDSEGMENT_UPDATE,
- mJetStatus.numQueuedSegments,
- -1,
- mJavaJetPlayerRef);
- }
- mPreviousJetStatus.numQueuedSegments = mJetStatus.numQueuedSegments;
- }
-
- if (mJetStatus.paused != mPreviousJetStatus.paused) {
- if (mEventCallback) {
- mEventCallback(JetPlayer::JET_PAUSE_UPDATE,
- mJetStatus.paused,
- -1,
- mJavaJetPlayerRef);
- }
- mPreviousJetStatus.paused = mJetStatus.paused;
- }
-
-}
-
-
-//-------------------------------------------------------------------------------------------------
-// fire up all the JET events in the JET engine queue (until the queue is empty)
-// precondition: mMutex locked
-void JetPlayer::fireEventsFromJetQueue()
-{
- if (!mEventCallback) {
- // no callback, just empty the event queue
- while (JET_GetEvent(mEasData, NULL, NULL)) { }
- return;
- }
-
- EAS_U32 rawEvent;
- while (JET_GetEvent(mEasData, &rawEvent, NULL)) {
- mEventCallback(
- JetPlayer::JET_EVENT,
- rawEvent,
- -1,
- mJavaJetPlayerRef);
- }
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::loadFromFile(const char* path)
-{
- ALOGV("JetPlayer::loadFromFile(): path=%s", path);
-
- Mutex::Autolock lock(mMutex);
-
- delete mIoWrapper;
- mIoWrapper = new MidiIoWrapper(path);
-
- EAS_RESULT result = JET_OpenFile(mEasData, mIoWrapper->getLocator());
- if (result != EAS_SUCCESS)
- mState = EAS_STATE_ERROR;
- else
- mState = EAS_STATE_OPEN;
- return( result );
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::loadFromFD(const int fd, const long long offset, const long long length)
-{
- ALOGV("JetPlayer::loadFromFD(): fd=%d offset=%lld length=%lld", fd, offset, length);
-
- Mutex::Autolock lock(mMutex);
-
- delete mIoWrapper;
- mIoWrapper = new MidiIoWrapper(fd, offset, length);
-
- EAS_RESULT result = JET_OpenFile(mEasData, mIoWrapper->getLocator());
- if (result != EAS_SUCCESS)
- mState = EAS_STATE_ERROR;
- else
- mState = EAS_STATE_OPEN;
- return( result );
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::closeFile()
-{
- Mutex::Autolock lock(mMutex);
- return JET_CloseFile(mEasData);
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::play()
-{
- ALOGV("JetPlayer::play(): entering");
- Mutex::Autolock lock(mMutex);
-
- EAS_RESULT result = JET_Play(mEasData);
-
- mPaused = false;
- mRender = true;
-
- JET_Status(mEasData, &mJetStatus);
- this->dumpJetStatus(&mJetStatus);
-
- fireUpdateOnStatusChange();
-
- // wake up render thread
- ALOGV("JetPlayer::play(): wakeup render thread");
- mCondition.signal();
-
- return result;
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::pause()
-{
- Mutex::Autolock lock(mMutex);
- mPaused = true;
- EAS_RESULT result = JET_Pause(mEasData);
-
- mRender = false;
-
- JET_Status(mEasData, &mJetStatus);
- this->dumpJetStatus(&mJetStatus);
- fireUpdateOnStatusChange();
-
-
- return result;
-}
-
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::queueSegment(int segmentNum, int libNum, int repeatCount, int transpose,
- EAS_U32 muteFlags, EAS_U8 userID)
-{
- ALOGV("JetPlayer::queueSegment segmentNum=%d, libNum=%d, repeatCount=%d, transpose=%d",
- segmentNum, libNum, repeatCount, transpose);
- Mutex::Autolock lock(mMutex);
- return JET_QueueSegment(mEasData, segmentNum, libNum, repeatCount, transpose, muteFlags,
- userID);
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::setMuteFlags(EAS_U32 muteFlags, bool sync)
-{
- Mutex::Autolock lock(mMutex);
- return JET_SetMuteFlags(mEasData, muteFlags, sync);
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::setMuteFlag(int trackNum, bool muteFlag, bool sync)
-{
- Mutex::Autolock lock(mMutex);
- return JET_SetMuteFlag(mEasData, trackNum, muteFlag, sync);
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::triggerClip(int clipId)
-{
- ALOGV("JetPlayer::triggerClip clipId=%d", clipId);
- Mutex::Autolock lock(mMutex);
- return JET_TriggerClip(mEasData, clipId);
-}
-
-//-------------------------------------------------------------------------------------------------
-int JetPlayer::clearQueue()
-{
- ALOGV("JetPlayer::clearQueue");
- Mutex::Autolock lock(mMutex);
- return JET_Clear_Queue(mEasData);
-}
-
-//-------------------------------------------------------------------------------------------------
-void JetPlayer::dump()
-{
-}
-
-void JetPlayer::dumpJetStatus(S_JET_STATUS* pJetStatus)
-{
- if (pJetStatus!=NULL)
- ALOGV(">> current JET player status: userID=%d segmentRepeatCount=%d numQueuedSegments=%d "
- "paused=%d",
- pJetStatus->currentUserID, pJetStatus->segmentRepeatCount,
- pJetStatus->numQueuedSegments, pJetStatus->paused);
- else
- ALOGE(">> JET player status is NULL");
-}
-
-
-} // end namespace android
diff --git a/media/libmedia/MediaUtils.cpp b/media/libmedia/MediaUtils.cpp
index 31972fa..2efb30e 100644
--- a/media/libmedia/MediaUtils.cpp
+++ b/media/libmedia/MediaUtils.cpp
@@ -22,7 +22,7 @@
#include <sys/resource.h>
#include <unistd.h>
-#include <bionic_malloc.h>
+#include <bionic/malloc.h>
#include "MediaUtils.h"
diff --git a/media/libmedia/MidiDeviceInfo.cpp b/media/libmedia/MidiDeviceInfo.cpp
deleted file mode 100644
index 7588e00..0000000
--- a/media/libmedia/MidiDeviceInfo.cpp
+++ /dev/null
@@ -1,138 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "MidiDeviceInfo"
-
-#include <media/MidiDeviceInfo.h>
-
-#include <binder/Parcel.h>
-#include <log/log.h>
-#include <utils/Errors.h>
-#include <utils/String16.h>
-
-namespace android {
-namespace media {
-namespace midi {
-
-// The constant values need to be kept in sync with MidiDeviceInfo.java.
-// static
-const char* const MidiDeviceInfo::PROPERTY_NAME = "name";
-const char* const MidiDeviceInfo::PROPERTY_MANUFACTURER = "manufacturer";
-const char* const MidiDeviceInfo::PROPERTY_PRODUCT = "product";
-const char* const MidiDeviceInfo::PROPERTY_VERSION = "version";
-const char* const MidiDeviceInfo::PROPERTY_SERIAL_NUMBER = "serial_number";
-const char* const MidiDeviceInfo::PROPERTY_ALSA_CARD = "alsa_card";
-const char* const MidiDeviceInfo::PROPERTY_ALSA_DEVICE = "alsa_device";
-
-String16 MidiDeviceInfo::getProperty(const char* propertyName) {
- String16 value;
- if (mProperties.getString(String16(propertyName), &value)) {
- return value;
- } else {
- return String16();
- }
-}
-
-#define RETURN_IF_FAILED(calledOnce) \
- { \
- status_t returnStatus = calledOnce; \
- if (returnStatus) { \
- ALOGE("Failed at %s:%d (%s)", __FILE__, __LINE__, __func__); \
- return returnStatus; \
- } \
- }
-
-status_t MidiDeviceInfo::writeToParcel(Parcel* parcel) const {
- // Needs to be kept in sync with code in MidiDeviceInfo.java
- RETURN_IF_FAILED(parcel->writeInt32(mType));
- RETURN_IF_FAILED(parcel->writeInt32(mId));
- RETURN_IF_FAILED(parcel->writeInt32((int32_t)mInputPortNames.size()));
- RETURN_IF_FAILED(parcel->writeInt32((int32_t)mOutputPortNames.size()));
- RETURN_IF_FAILED(writeStringVector(parcel, mInputPortNames));
- RETURN_IF_FAILED(writeStringVector(parcel, mOutputPortNames));
- RETURN_IF_FAILED(parcel->writeInt32(mIsPrivate ? 1 : 0));
- RETURN_IF_FAILED(mProperties.writeToParcel(parcel));
- // This corresponds to "extra" properties written by Java code
- RETURN_IF_FAILED(mProperties.writeToParcel(parcel));
- return OK;
-}
-
-status_t MidiDeviceInfo::readFromParcel(const Parcel* parcel) {
- // Needs to be kept in sync with code in MidiDeviceInfo.java
- RETURN_IF_FAILED(parcel->readInt32(&mType));
- RETURN_IF_FAILED(parcel->readInt32(&mId));
- int32_t inputPortCount;
- RETURN_IF_FAILED(parcel->readInt32(&inputPortCount));
- int32_t outputPortCount;
- RETURN_IF_FAILED(parcel->readInt32(&outputPortCount));
- RETURN_IF_FAILED(readStringVector(parcel, &mInputPortNames, inputPortCount));
- RETURN_IF_FAILED(readStringVector(parcel, &mOutputPortNames, outputPortCount));
- int32_t isPrivate;
- RETURN_IF_FAILED(parcel->readInt32(&isPrivate));
- mIsPrivate = isPrivate == 1;
- RETURN_IF_FAILED(mProperties.readFromParcel(parcel));
- // Ignore "extra" properties as they may contain Java Parcelables
- return OK;
-}
-
-status_t MidiDeviceInfo::readStringVector(
- const Parcel* parcel, Vector<String16> *vectorPtr, size_t defaultLength) {
- std::unique_ptr<std::vector<std::unique_ptr<String16>>> v;
- status_t result = parcel->readString16Vector(&v);
- if (result != OK) return result;
- vectorPtr->clear();
- if (v.get() != nullptr) {
- for (const auto& iter : *v) {
- if (iter.get() != nullptr) {
- vectorPtr->push_back(*iter);
- } else {
- vectorPtr->push_back(String16());
- }
- }
- } else {
- vectorPtr->resize(defaultLength);
- }
- return OK;
-}
-
-status_t MidiDeviceInfo::writeStringVector(Parcel* parcel, const Vector<String16>& vector) const {
- std::vector<String16> v;
- for (size_t i = 0; i < vector.size(); ++i) {
- v.push_back(vector[i]);
- }
- return parcel->writeString16Vector(v);
-}
-
-// Vector does not define operator==
-static inline bool areVectorsEqual(const Vector<String16>& lhs, const Vector<String16>& rhs) {
- if (lhs.size() != rhs.size()) return false;
- for (size_t i = 0; i < lhs.size(); ++i) {
- if (lhs[i] != rhs[i]) return false;
- }
- return true;
-}
-
-bool operator==(const MidiDeviceInfo& lhs, const MidiDeviceInfo& rhs) {
- return (lhs.mType == rhs.mType && lhs.mId == rhs.mId &&
- areVectorsEqual(lhs.mInputPortNames, rhs.mInputPortNames) &&
- areVectorsEqual(lhs.mOutputPortNames, rhs.mOutputPortNames) &&
- lhs.mProperties == rhs.mProperties &&
- lhs.mIsPrivate == rhs.mIsPrivate);
-}
-
-} // namespace midi
-} // namespace media
-} // namespace android
diff --git a/media/libmedia/MidiIoWrapper.cpp b/media/libmedia/MidiIoWrapper.cpp
index d8ef9cf..6d46363 100644
--- a/media/libmedia/MidiIoWrapper.cpp
+++ b/media/libmedia/MidiIoWrapper.cpp
@@ -17,7 +17,6 @@
//#define LOG_NDEBUG 0
#define LOG_TAG "MidiIoWrapper"
#include <utils/Log.h>
-#include <utils/RefBase.h>
#include <sys/stat.h>
#include <fcntl.h>
@@ -50,7 +49,7 @@
mDataSource = nullptr;
}
-class DataSourceUnwrapper : public DataSourceBase {
+class DataSourceUnwrapper {
public:
explicit DataSourceUnwrapper(CDataSource *csource) {
diff --git a/media/libmedia/NdkWrapper.cpp b/media/libmedia/NdkWrapper.cpp
deleted file mode 100644
index c150407..0000000
--- a/media/libmedia/NdkWrapper.cpp
+++ /dev/null
@@ -1,1290 +0,0 @@
-/*
- * Copyright 2017, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "NdkWrapper"
-
-#include <media/NdkWrapper.h>
-
-#include <android/native_window.h>
-#include <log/log.h>
-#include <media/NdkMediaCodec.h>
-#include <media/NdkMediaCrypto.h>
-#include <media/NdkMediaDrm.h>
-#include <media/NdkMediaFormat.h>
-#include <media/NdkMediaExtractor.h>
-#include <media/stagefright/MetaData.h>
-#include <media/stagefright/foundation/ABuffer.h>
-#include <media/stagefright/foundation/AMessage.h>
-#include <utils/Errors.h>
-
-#include "NdkMediaDataSourceCallbacksPriv.h"
-
-namespace android {
-
-static const size_t kAESBlockSize = 16; // AES_BLOCK_SIZE
-
-static const char *AMediaFormatKeyGroupInt32[] = {
- AMEDIAFORMAT_KEY_AAC_DRC_ATTENUATION_FACTOR,
- AMEDIAFORMAT_KEY_AAC_DRC_BOOST_FACTOR,
- AMEDIAFORMAT_KEY_AAC_DRC_HEAVY_COMPRESSION,
- AMEDIAFORMAT_KEY_AAC_DRC_TARGET_REFERENCE_LEVEL,
- AMEDIAFORMAT_KEY_AAC_ENCODED_TARGET_LEVEL,
- AMEDIAFORMAT_KEY_AAC_MAX_OUTPUT_CHANNEL_COUNT,
- AMEDIAFORMAT_KEY_AAC_PROFILE,
- AMEDIAFORMAT_KEY_AAC_SBR_MODE,
- AMEDIAFORMAT_KEY_AUDIO_SESSION_ID,
- AMEDIAFORMAT_KEY_BITRATE_MODE,
- AMEDIAFORMAT_KEY_BIT_RATE,
- AMEDIAFORMAT_KEY_CAPTURE_RATE,
- AMEDIAFORMAT_KEY_CHANNEL_COUNT,
- AMEDIAFORMAT_KEY_CHANNEL_MASK,
- AMEDIAFORMAT_KEY_COLOR_FORMAT,
- AMEDIAFORMAT_KEY_COLOR_RANGE,
- AMEDIAFORMAT_KEY_COLOR_STANDARD,
- AMEDIAFORMAT_KEY_COLOR_TRANSFER,
- AMEDIAFORMAT_KEY_COMPLEXITY,
- AMEDIAFORMAT_KEY_CREATE_INPUT_SURFACE_SUSPENDED,
- AMEDIAFORMAT_KEY_CRYPTO_DEFAULT_IV_SIZE,
- AMEDIAFORMAT_KEY_CRYPTO_ENCRYPTED_BYTE_BLOCK,
- AMEDIAFORMAT_KEY_CRYPTO_MODE,
- AMEDIAFORMAT_KEY_CRYPTO_SKIP_BYTE_BLOCK,
- AMEDIAFORMAT_KEY_FLAC_COMPRESSION_LEVEL,
- AMEDIAFORMAT_KEY_GRID_COLUMNS,
- AMEDIAFORMAT_KEY_GRID_ROWS,
- AMEDIAFORMAT_KEY_HAPTIC_CHANNEL_COUNT,
- AMEDIAFORMAT_KEY_HEIGHT,
- AMEDIAFORMAT_KEY_INTRA_REFRESH_PERIOD,
- AMEDIAFORMAT_KEY_IS_ADTS,
- AMEDIAFORMAT_KEY_IS_AUTOSELECT,
- AMEDIAFORMAT_KEY_IS_DEFAULT,
- AMEDIAFORMAT_KEY_IS_FORCED_SUBTITLE,
- AMEDIAFORMAT_KEY_LATENCY,
- AMEDIAFORMAT_KEY_LEVEL,
- AMEDIAFORMAT_KEY_MAX_HEIGHT,
- AMEDIAFORMAT_KEY_MAX_INPUT_SIZE,
- AMEDIAFORMAT_KEY_MAX_WIDTH,
- AMEDIAFORMAT_KEY_PCM_ENCODING,
- AMEDIAFORMAT_KEY_PRIORITY,
- AMEDIAFORMAT_KEY_PROFILE,
- AMEDIAFORMAT_KEY_PUSH_BLANK_BUFFERS_ON_STOP,
- AMEDIAFORMAT_KEY_ROTATION,
- AMEDIAFORMAT_KEY_SAMPLE_RATE,
- AMEDIAFORMAT_KEY_SLICE_HEIGHT,
- AMEDIAFORMAT_KEY_STRIDE,
- AMEDIAFORMAT_KEY_TRACK_ID,
- AMEDIAFORMAT_KEY_WIDTH,
- AMEDIAFORMAT_KEY_DISPLAY_HEIGHT,
- AMEDIAFORMAT_KEY_DISPLAY_WIDTH,
- AMEDIAFORMAT_KEY_TEMPORAL_LAYER_ID,
- AMEDIAFORMAT_KEY_TILE_HEIGHT,
- AMEDIAFORMAT_KEY_TILE_WIDTH,
- AMEDIAFORMAT_KEY_TRACK_INDEX,
-};
-
-static const char *AMediaFormatKeyGroupInt64[] = {
- AMEDIAFORMAT_KEY_DURATION,
- AMEDIAFORMAT_KEY_MAX_PTS_GAP_TO_ENCODER,
- AMEDIAFORMAT_KEY_REPEAT_PREVIOUS_FRAME_AFTER,
- AMEDIAFORMAT_KEY_TIME_US,
-};
-
-static const char *AMediaFormatKeyGroupString[] = {
- AMEDIAFORMAT_KEY_LANGUAGE,
- AMEDIAFORMAT_KEY_MIME,
- AMEDIAFORMAT_KEY_TEMPORAL_LAYERING,
-};
-
-static const char *AMediaFormatKeyGroupBuffer[] = {
- AMEDIAFORMAT_KEY_CRYPTO_IV,
- AMEDIAFORMAT_KEY_CRYPTO_KEY,
- AMEDIAFORMAT_KEY_HDR_STATIC_INFO,
- AMEDIAFORMAT_KEY_SEI,
- AMEDIAFORMAT_KEY_MPEG_USER_DATA,
-};
-
-static const char *AMediaFormatKeyGroupCsd[] = {
- AMEDIAFORMAT_KEY_CSD_0,
- AMEDIAFORMAT_KEY_CSD_1,
- AMEDIAFORMAT_KEY_CSD_2,
-};
-
-static const char *AMediaFormatKeyGroupRect[] = {
- AMEDIAFORMAT_KEY_DISPLAY_CROP,
-};
-
-static const char *AMediaFormatKeyGroupFloatInt32[] = {
- AMEDIAFORMAT_KEY_FRAME_RATE,
- AMEDIAFORMAT_KEY_I_FRAME_INTERVAL,
- AMEDIAFORMAT_KEY_MAX_FPS_TO_ENCODER,
- AMEDIAFORMAT_KEY_OPERATING_RATE,
-};
-
-static status_t translateErrorCode(media_status_t err) {
- if (err == AMEDIA_OK) {
- return OK;
- } else if (err == AMEDIA_ERROR_END_OF_STREAM) {
- return ERROR_END_OF_STREAM;
- } else if (err == AMEDIA_ERROR_IO) {
- return ERROR_IO;
- } else if (err == AMEDIACODEC_INFO_TRY_AGAIN_LATER) {
- return -EAGAIN;
- }
-
- ALOGE("ndk error code: %d", err);
- return UNKNOWN_ERROR;
-}
-
-static int32_t translateActionCode(int32_t actionCode) {
- if (AMediaCodecActionCode_isTransient(actionCode)) {
- return ACTION_CODE_TRANSIENT;
- } else if (AMediaCodecActionCode_isRecoverable(actionCode)) {
- return ACTION_CODE_RECOVERABLE;
- }
- return ACTION_CODE_FATAL;
-}
-
-static CryptoPlugin::Mode translateToCryptoPluginMode(cryptoinfo_mode_t mode) {
- CryptoPlugin::Mode ret = CryptoPlugin::kMode_Unencrypted;
- switch (mode) {
- case AMEDIACODECRYPTOINFO_MODE_AES_CTR: {
- ret = CryptoPlugin::kMode_AES_CTR;
- break;
- }
-
- case AMEDIACODECRYPTOINFO_MODE_AES_WV: {
- ret = CryptoPlugin::kMode_AES_WV;
- break;
- }
-
- case AMEDIACODECRYPTOINFO_MODE_AES_CBC: {
- ret = CryptoPlugin::kMode_AES_CBC;
- break;
- }
-
- default:
- break;
- }
-
- return ret;
-}
-
-static cryptoinfo_mode_t translateToCryptoInfoMode(CryptoPlugin::Mode mode) {
- cryptoinfo_mode_t ret = AMEDIACODECRYPTOINFO_MODE_CLEAR;
- switch (mode) {
- case CryptoPlugin::kMode_AES_CTR: {
- ret = AMEDIACODECRYPTOINFO_MODE_AES_CTR;
- break;
- }
-
- case CryptoPlugin::kMode_AES_WV: {
- ret = AMEDIACODECRYPTOINFO_MODE_AES_WV;
- break;
- }
-
- case CryptoPlugin::kMode_AES_CBC: {
- ret = AMEDIACODECRYPTOINFO_MODE_AES_CBC;
- break;
- }
-
- default:
- break;
- }
-
- return ret;
-}
-
-//////////// AMediaFormatWrapper
-// static
-sp<AMediaFormatWrapper> AMediaFormatWrapper::Create(const sp<AMessage> &message) {
- sp<AMediaFormatWrapper> aMediaFormat = new AMediaFormatWrapper();
-
- for (size_t i = 0; i < message->countEntries(); ++i) {
- AMessage::Type valueType;
- const char *key = message->getEntryNameAt(i, &valueType);
-
- switch (valueType) {
- case AMessage::kTypeInt32: {
- int32_t val;
- if (!message->findInt32(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setInt32(key, val);
- break;
- }
-
- case AMessage::kTypeInt64: {
- int64_t val;
- if (!message->findInt64(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setInt64(key, val);
- break;
- }
-
- case AMessage::kTypeFloat: {
- float val;
- if (!message->findFloat(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setFloat(key, val);
- break;
- }
-
- case AMessage::kTypeDouble: {
- double val;
- if (!message->findDouble(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setDouble(key, val);
- break;
- }
-
- case AMessage::kTypeSize: {
- size_t val;
- if (!message->findSize(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setSize(key, val);
- break;
- }
-
- case AMessage::kTypeRect: {
- int32_t left, top, right, bottom;
- if (!message->findRect(key, &left, &top, &right, &bottom)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setRect(key, left, top, right, bottom);
- break;
- }
-
- case AMessage::kTypeString: {
- AString val;
- if (!message->findString(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setString(key, val);
- break;
- }
-
- case AMessage::kTypeBuffer: {
- sp<ABuffer> val;
- if (!message->findBuffer(key, &val)) {
- ALOGE("AMediaFormatWrapper::Create: error at item %zu", i);
- continue;
- }
- aMediaFormat->setBuffer(key, val->data(), val->size());
- break;
- }
-
- default: {
- break;
- }
- }
- }
-
- return aMediaFormat;
-}
-
-AMediaFormatWrapper::AMediaFormatWrapper() {
- mAMediaFormat = AMediaFormat_new();
-}
-
-AMediaFormatWrapper::AMediaFormatWrapper(AMediaFormat *aMediaFormat)
- : mAMediaFormat(aMediaFormat) {
-}
-
-AMediaFormatWrapper::~AMediaFormatWrapper() {
- release();
-}
-
-status_t AMediaFormatWrapper::release() {
- if (mAMediaFormat != NULL) {
- media_status_t err = AMediaFormat_delete(mAMediaFormat);
- mAMediaFormat = NULL;
- return translateErrorCode(err);
- }
- return OK;
-}
-
-AMediaFormat *AMediaFormatWrapper::getAMediaFormat() const {
- return mAMediaFormat;
-}
-
-sp<AMessage> AMediaFormatWrapper::toAMessage() const {
- sp<AMessage> msg;
- writeToAMessage(msg);
- return msg;
-}
-
-void AMediaFormatWrapper::writeToAMessage(sp<AMessage> &msg) const {
- if (mAMediaFormat == NULL) {
- msg = NULL;
- }
-
- if (msg == NULL) {
- msg = new AMessage;
- }
- for (auto& key : AMediaFormatKeyGroupInt32) {
- int32_t val;
- if (getInt32(key, &val)) {
- msg->setInt32(key, val);
- }
- }
- for (auto& key : AMediaFormatKeyGroupInt64) {
- int64_t val;
- if (getInt64(key, &val)) {
- msg->setInt64(key, val);
- }
- }
- for (auto& key : AMediaFormatKeyGroupString) {
- AString val;
- if (getString(key, &val)) {
- msg->setString(key, val);
- }
- }
- for (auto& key : AMediaFormatKeyGroupBuffer) {
- void *data;
- size_t size;
- if (getBuffer(key, &data, &size)) {
- sp<ABuffer> buffer = ABuffer::CreateAsCopy(data, size);
- msg->setBuffer(key, buffer);
- }
- }
- for (auto& key : AMediaFormatKeyGroupCsd) {
- void *data;
- size_t size;
- if (getBuffer(key, &data, &size)) {
- sp<ABuffer> buffer = ABuffer::CreateAsCopy(data, size);
- buffer->meta()->setInt32(AMEDIAFORMAT_KEY_CSD, 1);
- buffer->meta()->setInt64(AMEDIAFORMAT_KEY_TIME_US, 0);
- msg->setBuffer(key, buffer);
- }
- }
- for (auto& key : AMediaFormatKeyGroupRect) {
- int32_t left, top, right, bottom;
- if (getRect(key, &left, &top, &right, &bottom)) {
- msg->setRect(key, left, top, right, bottom);
- }
- }
- for (auto& key : AMediaFormatKeyGroupFloatInt32) {
- float valFloat;
- if (getFloat(key, &valFloat)) {
- msg->setFloat(key, valFloat);
- } else {
- int32_t valInt32;
- if (getInt32(key, &valInt32)) {
- msg->setFloat(key, (float)valInt32);
- }
- }
- }
-}
-
-const char* AMediaFormatWrapper::toString() const {
- if (mAMediaFormat == NULL) {
- return NULL;
- }
- return AMediaFormat_toString(mAMediaFormat);
-}
-
-bool AMediaFormatWrapper::getInt32(const char *name, int32_t *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getInt32(mAMediaFormat, name, out);
-}
-
-bool AMediaFormatWrapper::getInt64(const char *name, int64_t *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getInt64(mAMediaFormat, name, out);
-}
-
-bool AMediaFormatWrapper::getFloat(const char *name, float *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getFloat(mAMediaFormat, name, out);
-}
-
-bool AMediaFormatWrapper::getDouble(const char *name, double *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getDouble(mAMediaFormat, name, out);
-}
-
-bool AMediaFormatWrapper::getSize(const char *name, size_t *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getSize(mAMediaFormat, name, out);
-}
-
-bool AMediaFormatWrapper::getRect(
- const char *name, int32_t *left, int32_t *top, int32_t *right, int32_t *bottom) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getRect(mAMediaFormat, name, left, top, right, bottom);
-}
-
-bool AMediaFormatWrapper::getBuffer(const char *name, void** data, size_t *outSize) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- return AMediaFormat_getBuffer(mAMediaFormat, name, data, outSize);
-}
-
-bool AMediaFormatWrapper::getString(const char *name, AString *out) const {
- if (mAMediaFormat == NULL) {
- return false;
- }
- const char *outChar = NULL;
- bool ret = AMediaFormat_getString(mAMediaFormat, name, &outChar);
- if (ret) {
- *out = AString(outChar);
- }
- return ret;
-}
-
-void AMediaFormatWrapper::setInt32(const char* name, int32_t value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setInt32(mAMediaFormat, name, value);
- }
-}
-
-void AMediaFormatWrapper::setInt64(const char* name, int64_t value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setInt64(mAMediaFormat, name, value);
- }
-}
-
-void AMediaFormatWrapper::setFloat(const char* name, float value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setFloat(mAMediaFormat, name, value);
- }
-}
-
-void AMediaFormatWrapper::setDouble(const char* name, double value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setDouble(mAMediaFormat, name, value);
- }
-}
-
-void AMediaFormatWrapper::setSize(const char* name, size_t value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setSize(mAMediaFormat, name, value);
- }
-}
-
-void AMediaFormatWrapper::setRect(
- const char* name, int32_t left, int32_t top, int32_t right, int32_t bottom) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setRect(mAMediaFormat, name, left, top, right, bottom);
- }
-}
-
-void AMediaFormatWrapper::setString(const char* name, const AString &value) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setString(mAMediaFormat, name, value.c_str());
- }
-}
-
-void AMediaFormatWrapper::setBuffer(const char* name, void* data, size_t size) {
- if (mAMediaFormat != NULL) {
- AMediaFormat_setBuffer(mAMediaFormat, name, data, size);
- }
-}
-
-
-//////////// ANativeWindowWrapper
-ANativeWindowWrapper::ANativeWindowWrapper(ANativeWindow *aNativeWindow)
- : mANativeWindow(aNativeWindow) {
- if (aNativeWindow != NULL) {
- ANativeWindow_acquire(aNativeWindow);
- }
-}
-
-ANativeWindowWrapper::~ANativeWindowWrapper() {
- release();
-}
-
-status_t ANativeWindowWrapper::release() {
- if (mANativeWindow != NULL) {
- ANativeWindow_release(mANativeWindow);
- mANativeWindow = NULL;
- }
- return OK;
-}
-
-ANativeWindow *ANativeWindowWrapper::getANativeWindow() const {
- return mANativeWindow;
-}
-
-
-//////////// AMediaDrmWrapper
-AMediaDrmWrapper::AMediaDrmWrapper(const uint8_t uuid[16]) {
- mAMediaDrm = AMediaDrm_createByUUID(uuid);
-}
-
-AMediaDrmWrapper::AMediaDrmWrapper(AMediaDrm *aMediaDrm)
- : mAMediaDrm(aMediaDrm) {
-}
-
-AMediaDrmWrapper::~AMediaDrmWrapper() {
- release();
-}
-
-status_t AMediaDrmWrapper::release() {
- if (mAMediaDrm != NULL) {
- AMediaDrm_release(mAMediaDrm);
- mAMediaDrm = NULL;
- }
- return OK;
-}
-
-AMediaDrm *AMediaDrmWrapper::getAMediaDrm() const {
- return mAMediaDrm;
-}
-
-// static
-bool AMediaDrmWrapper::isCryptoSchemeSupported(
- const uint8_t uuid[16],
- const char *mimeType) {
- return AMediaDrm_isCryptoSchemeSupported(uuid, mimeType);
-}
-
-
-//////////// AMediaCryptoWrapper
-AMediaCryptoWrapper::AMediaCryptoWrapper(
- const uint8_t uuid[16], const void *initData, size_t initDataSize) {
- mAMediaCrypto = AMediaCrypto_new(uuid, initData, initDataSize);
-}
-
-AMediaCryptoWrapper::AMediaCryptoWrapper(AMediaCrypto *aMediaCrypto)
- : mAMediaCrypto(aMediaCrypto) {
-}
-
-AMediaCryptoWrapper::~AMediaCryptoWrapper() {
- release();
-}
-
-status_t AMediaCryptoWrapper::release() {
- if (mAMediaCrypto != NULL) {
- AMediaCrypto_delete(mAMediaCrypto);
- mAMediaCrypto = NULL;
- }
- return OK;
-}
-
-AMediaCrypto *AMediaCryptoWrapper::getAMediaCrypto() const {
- return mAMediaCrypto;
-}
-
-bool AMediaCryptoWrapper::isCryptoSchemeSupported(const uint8_t uuid[16]) {
- if (mAMediaCrypto == NULL) {
- return false;
- }
- return AMediaCrypto_isCryptoSchemeSupported(uuid);
-}
-
-bool AMediaCryptoWrapper::requiresSecureDecoderComponent(const char *mime) {
- if (mAMediaCrypto == NULL) {
- return false;
- }
- return AMediaCrypto_requiresSecureDecoderComponent(mime);
-}
-
-
-//////////// AMediaCodecCryptoInfoWrapper
-// static
-sp<AMediaCodecCryptoInfoWrapper> AMediaCodecCryptoInfoWrapper::Create(MetaDataBase &meta) {
-
- uint32_t type;
- const void *crypteddata;
- size_t cryptedsize;
-
- if (!meta.findData(kKeyEncryptedSizes, &type, &crypteddata, &cryptedsize)) {
- return NULL;
- }
-
- int numSubSamples = cryptedsize / sizeof(size_t);
-
- if (numSubSamples <= 0) {
- ALOGE("Create: INVALID numSubSamples: %d", numSubSamples);
- return NULL;
- }
-
- const void *cleardata;
- size_t clearsize;
- if (meta.findData(kKeyPlainSizes, &type, &cleardata, &clearsize)) {
- if (clearsize != cryptedsize) {
- // The two must be of the same length.
- ALOGE("Create: mismatch cryptedsize: %zu != clearsize: %zu", cryptedsize, clearsize);
- return NULL;
- }
- }
-
- const void *key;
- size_t keysize;
- if (meta.findData(kKeyCryptoKey, &type, &key, &keysize)) {
- if (keysize != kAESBlockSize) {
- // Keys must be 16 bytes in length.
- ALOGE("Create: Keys must be %zu bytes in length: %zu", kAESBlockSize, keysize);
- return NULL;
- }
- }
-
- const void *iv;
- size_t ivsize;
- if (meta.findData(kKeyCryptoIV, &type, &iv, &ivsize)) {
- if (ivsize != kAESBlockSize) {
- // IVs must be 16 bytes in length.
- ALOGE("Create: IV must be %zu bytes in length: %zu", kAESBlockSize, ivsize);
- return NULL;
- }
- }
-
- int32_t mode;
- if (!meta.findInt32(kKeyCryptoMode, &mode)) {
- mode = CryptoPlugin::kMode_AES_CTR;
- }
-
- return new AMediaCodecCryptoInfoWrapper(
- numSubSamples,
- (uint8_t*) key,
- (uint8_t*) iv,
- (CryptoPlugin::Mode)mode,
- (size_t*) cleardata,
- (size_t*) crypteddata);
-}
-
-AMediaCodecCryptoInfoWrapper::AMediaCodecCryptoInfoWrapper(
- int numsubsamples,
- uint8_t key[16],
- uint8_t iv[16],
- CryptoPlugin::Mode mode,
- size_t *clearbytes,
- size_t *encryptedbytes) {
- mAMediaCodecCryptoInfo =
- AMediaCodecCryptoInfo_new(numsubsamples,
- key,
- iv,
- translateToCryptoInfoMode(mode),
- clearbytes,
- encryptedbytes);
-}
-
-AMediaCodecCryptoInfoWrapper::AMediaCodecCryptoInfoWrapper(
- AMediaCodecCryptoInfo *aMediaCodecCryptoInfo)
- : mAMediaCodecCryptoInfo(aMediaCodecCryptoInfo) {
-}
-
-AMediaCodecCryptoInfoWrapper::~AMediaCodecCryptoInfoWrapper() {
- release();
-}
-
-status_t AMediaCodecCryptoInfoWrapper::release() {
- if (mAMediaCodecCryptoInfo != NULL) {
- media_status_t err = AMediaCodecCryptoInfo_delete(mAMediaCodecCryptoInfo);
- mAMediaCodecCryptoInfo = NULL;
- return translateErrorCode(err);
- }
- return OK;
-}
-
-AMediaCodecCryptoInfo *AMediaCodecCryptoInfoWrapper::getAMediaCodecCryptoInfo() const {
- return mAMediaCodecCryptoInfo;
-}
-
-void AMediaCodecCryptoInfoWrapper::setPattern(CryptoPlugin::Pattern *pattern) {
- if (mAMediaCodecCryptoInfo == NULL || pattern == NULL) {
- return;
- }
- cryptoinfo_pattern_t ndkPattern = {(int32_t)pattern->mEncryptBlocks,
- (int32_t)pattern->mSkipBlocks };
- return AMediaCodecCryptoInfo_setPattern(mAMediaCodecCryptoInfo, &ndkPattern);
-}
-
-size_t AMediaCodecCryptoInfoWrapper::getNumSubSamples() {
- if (mAMediaCodecCryptoInfo == NULL) {
- return 0;
- }
- return AMediaCodecCryptoInfo_getNumSubSamples(mAMediaCodecCryptoInfo);
-}
-
-status_t AMediaCodecCryptoInfoWrapper::getKey(uint8_t *dst) {
- if (mAMediaCodecCryptoInfo == NULL) {
- return DEAD_OBJECT;
- }
- if (dst == NULL) {
- return BAD_VALUE;
- }
- return translateErrorCode(
- AMediaCodecCryptoInfo_getKey(mAMediaCodecCryptoInfo, dst));
-}
-
-status_t AMediaCodecCryptoInfoWrapper::getIV(uint8_t *dst) {
- if (mAMediaCodecCryptoInfo == NULL) {
- return DEAD_OBJECT;
- }
- if (dst == NULL) {
- return BAD_VALUE;
- }
- return translateErrorCode(
- AMediaCodecCryptoInfo_getIV(mAMediaCodecCryptoInfo, dst));
-}
-
-CryptoPlugin::Mode AMediaCodecCryptoInfoWrapper::getMode() {
- if (mAMediaCodecCryptoInfo == NULL) {
- return CryptoPlugin::kMode_Unencrypted;
- }
- return translateToCryptoPluginMode(
- AMediaCodecCryptoInfo_getMode(mAMediaCodecCryptoInfo));
-}
-
-status_t AMediaCodecCryptoInfoWrapper::getClearBytes(size_t *dst) {
- if (mAMediaCodecCryptoInfo == NULL) {
- return DEAD_OBJECT;
- }
- if (dst == NULL) {
- return BAD_VALUE;
- }
- return translateErrorCode(
- AMediaCodecCryptoInfo_getClearBytes(mAMediaCodecCryptoInfo, dst));
-}
-
-status_t AMediaCodecCryptoInfoWrapper::getEncryptedBytes(size_t *dst) {
- if (mAMediaCodecCryptoInfo == NULL) {
- return DEAD_OBJECT;
- }
- if (dst == NULL) {
- return BAD_VALUE;
- }
- return translateErrorCode(
- AMediaCodecCryptoInfo_getEncryptedBytes(mAMediaCodecCryptoInfo, dst));
-}
-
-
-//////////// AMediaCodecWrapper
-// static
-sp<AMediaCodecWrapper> AMediaCodecWrapper::CreateCodecByName(const AString &name) {
- AMediaCodec *aMediaCodec = AMediaCodec_createCodecByName(name.c_str());
- return new AMediaCodecWrapper(aMediaCodec);
-}
-
-// static
-sp<AMediaCodecWrapper> AMediaCodecWrapper::CreateDecoderByType(const AString &mimeType) {
- AMediaCodec *aMediaCodec = AMediaCodec_createDecoderByType(mimeType.c_str());
- return new AMediaCodecWrapper(aMediaCodec);
-}
-
-// static
-void AMediaCodecWrapper::OnInputAvailableCB(
- AMediaCodec * /* aMediaCodec */,
- void *userdata,
- int32_t index) {
- ALOGV("OnInputAvailableCB: index(%d)", index);
- sp<AMessage> msg = sp<AMessage>((AMessage *)userdata)->dup();
- msg->setInt32("callbackID", CB_INPUT_AVAILABLE);
- msg->setInt32("index", index);
- msg->post();
-}
-
-// static
-void AMediaCodecWrapper::OnOutputAvailableCB(
- AMediaCodec * /* aMediaCodec */,
- void *userdata,
- int32_t index,
- AMediaCodecBufferInfo *bufferInfo) {
- ALOGV("OnOutputAvailableCB: index(%d), (%d, %d, %lld, 0x%x)",
- index, bufferInfo->offset, bufferInfo->size,
- (long long)bufferInfo->presentationTimeUs, bufferInfo->flags);
- sp<AMessage> msg = sp<AMessage>((AMessage *)userdata)->dup();
- msg->setInt32("callbackID", CB_OUTPUT_AVAILABLE);
- msg->setInt32("index", index);
- msg->setSize("offset", (size_t)(bufferInfo->offset));
- msg->setSize("size", (size_t)(bufferInfo->size));
- msg->setInt64("timeUs", bufferInfo->presentationTimeUs);
- msg->setInt32("flags", (int32_t)(bufferInfo->flags));
- msg->post();
-}
-
-// static
-void AMediaCodecWrapper::OnFormatChangedCB(
- AMediaCodec * /* aMediaCodec */,
- void *userdata,
- AMediaFormat *format) {
- sp<AMediaFormatWrapper> formatWrapper = new AMediaFormatWrapper(format);
- sp<AMessage> outputFormat = formatWrapper->toAMessage();
- ALOGV("OnFormatChangedCB: format(%s)", outputFormat->debugString().c_str());
-
- sp<AMessage> msg = sp<AMessage>((AMessage *)userdata)->dup();
- msg->setInt32("callbackID", CB_OUTPUT_FORMAT_CHANGED);
- msg->setMessage("format", outputFormat);
- msg->post();
-}
-
-// static
-void AMediaCodecWrapper::OnErrorCB(
- AMediaCodec * /* aMediaCodec */,
- void *userdata,
- media_status_t err,
- int32_t actionCode,
- const char *detail) {
- ALOGV("OnErrorCB: err(%d), actionCode(%d), detail(%s)", err, actionCode, detail);
- sp<AMessage> msg = sp<AMessage>((AMessage *)userdata)->dup();
- msg->setInt32("callbackID", CB_ERROR);
- msg->setInt32("err", translateErrorCode(err));
- msg->setInt32("actionCode", translateActionCode(actionCode));
- msg->setString("detail", detail);
- msg->post();
-}
-
-AMediaCodecWrapper::AMediaCodecWrapper(AMediaCodec *aMediaCodec)
- : mAMediaCodec(aMediaCodec) {
-}
-
-AMediaCodecWrapper::~AMediaCodecWrapper() {
- release();
-}
-
-status_t AMediaCodecWrapper::release() {
- if (mAMediaCodec != NULL) {
- AMediaCodecOnAsyncNotifyCallback aCB = {};
- AMediaCodec_setAsyncNotifyCallback(mAMediaCodec, aCB, NULL);
- mCallback = NULL;
-
- media_status_t err = AMediaCodec_delete(mAMediaCodec);
- mAMediaCodec = NULL;
- return translateErrorCode(err);
- }
- return OK;
-}
-
-AMediaCodec *AMediaCodecWrapper::getAMediaCodec() const {
- return mAMediaCodec;
-}
-
-status_t AMediaCodecWrapper::getName(AString *outComponentName) const {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- char *name = NULL;
- media_status_t err = AMediaCodec_getName(mAMediaCodec, &name);
- if (err != AMEDIA_OK) {
- return translateErrorCode(err);
- }
-
- *outComponentName = AString(name);
- AMediaCodec_releaseName(mAMediaCodec, name);
- return OK;
-}
-
-status_t AMediaCodecWrapper::configure(
- const sp<AMediaFormatWrapper> &format,
- const sp<ANativeWindowWrapper> &nww,
- const sp<AMediaCryptoWrapper> &crypto,
- uint32_t flags) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
-
- media_status_t err = AMediaCodec_configure(
- mAMediaCodec,
- format->getAMediaFormat(),
- (nww == NULL ? NULL : nww->getANativeWindow()),
- crypto == NULL ? NULL : crypto->getAMediaCrypto(),
- flags);
-
- return translateErrorCode(err);
-}
-
-status_t AMediaCodecWrapper::setCallback(const sp<AMessage> &callback) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
-
- mCallback = callback;
-
- AMediaCodecOnAsyncNotifyCallback aCB = {
- OnInputAvailableCB,
- OnOutputAvailableCB,
- OnFormatChangedCB,
- OnErrorCB
- };
-
- return translateErrorCode(
- AMediaCodec_setAsyncNotifyCallback(mAMediaCodec, aCB, callback.get()));
-}
-
-status_t AMediaCodecWrapper::releaseCrypto() {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaCodec_releaseCrypto(mAMediaCodec));
-}
-
-status_t AMediaCodecWrapper::start() {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaCodec_start(mAMediaCodec));
-}
-
-status_t AMediaCodecWrapper::stop() {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaCodec_stop(mAMediaCodec));
-}
-
-status_t AMediaCodecWrapper::flush() {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaCodec_flush(mAMediaCodec));
-}
-
-uint8_t* AMediaCodecWrapper::getInputBuffer(size_t idx, size_t *out_size) {
- if (mAMediaCodec == NULL) {
- return NULL;
- }
- return AMediaCodec_getInputBuffer(mAMediaCodec, idx, out_size);
-}
-
-uint8_t* AMediaCodecWrapper::getOutputBuffer(size_t idx, size_t *out_size) {
- if (mAMediaCodec == NULL) {
- return NULL;
- }
- return AMediaCodec_getOutputBuffer(mAMediaCodec, idx, out_size);
-}
-
-status_t AMediaCodecWrapper::queueInputBuffer(
- size_t idx,
- size_t offset,
- size_t size,
- uint64_t time,
- uint32_t flags) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_queueInputBuffer(mAMediaCodec, idx, offset, size, time, flags));
-}
-
-status_t AMediaCodecWrapper::queueSecureInputBuffer(
- size_t idx,
- size_t offset,
- sp<AMediaCodecCryptoInfoWrapper> &codecCryptoInfo,
- uint64_t time,
- uint32_t flags) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_queueSecureInputBuffer(
- mAMediaCodec,
- idx,
- offset,
- codecCryptoInfo->getAMediaCodecCryptoInfo(),
- time,
- flags));
-}
-
-sp<AMediaFormatWrapper> AMediaCodecWrapper::getOutputFormat() {
- if (mAMediaCodec == NULL) {
- return NULL;
- }
- return new AMediaFormatWrapper(AMediaCodec_getOutputFormat(mAMediaCodec));
-}
-
-sp<AMediaFormatWrapper> AMediaCodecWrapper::getInputFormat() {
- if (mAMediaCodec == NULL) {
- return NULL;
- }
- return new AMediaFormatWrapper(AMediaCodec_getInputFormat(mAMediaCodec));
-}
-
-status_t AMediaCodecWrapper::releaseOutputBuffer(size_t idx, bool render) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_releaseOutputBuffer(mAMediaCodec, idx, render));
-}
-
-status_t AMediaCodecWrapper::setOutputSurface(const sp<ANativeWindowWrapper> &nww) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_setOutputSurface(mAMediaCodec,
- (nww == NULL ? NULL : nww->getANativeWindow())));
-}
-
-status_t AMediaCodecWrapper::releaseOutputBufferAtTime(size_t idx, int64_t timestampNs) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_releaseOutputBufferAtTime(mAMediaCodec, idx, timestampNs));
-}
-
-status_t AMediaCodecWrapper::setParameters(const sp<AMediaFormatWrapper> ¶ms) {
- if (mAMediaCodec == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(
- AMediaCodec_setParameters(mAMediaCodec, params->getAMediaFormat()));
-}
-
-//////////// AMediaExtractorWrapper
-
-AMediaExtractorWrapper::AMediaExtractorWrapper(AMediaExtractor *aMediaExtractor)
- : mAMediaExtractor(aMediaExtractor) {
-}
-
-AMediaExtractorWrapper::~AMediaExtractorWrapper() {
- release();
-}
-
-status_t AMediaExtractorWrapper::release() {
- if (mAMediaExtractor != NULL) {
- media_status_t err = AMediaExtractor_delete(mAMediaExtractor);
- mAMediaExtractor = NULL;
- return translateErrorCode(err);
- }
- return OK;
-}
-
-AMediaExtractor *AMediaExtractorWrapper::getAMediaExtractor() const {
- return mAMediaExtractor;
-}
-
-status_t AMediaExtractorWrapper::setDataSource(int fd, off64_t offset, off64_t length) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaExtractor_setDataSourceFd(
- mAMediaExtractor, fd, offset, length));
-}
-
-status_t AMediaExtractorWrapper::setDataSource(const char *location) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaExtractor_setDataSource(mAMediaExtractor, location));
-}
-
-status_t AMediaExtractorWrapper::setDataSource(AMediaDataSource *source) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaExtractor_setDataSourceCustom(mAMediaExtractor, source));
-}
-
-size_t AMediaExtractorWrapper::getTrackCount() {
- if (mAMediaExtractor == NULL) {
- return 0;
- }
- return AMediaExtractor_getTrackCount(mAMediaExtractor);
-}
-
-sp<AMediaFormatWrapper> AMediaExtractorWrapper::getFormat() {
- if (mAMediaExtractor == NULL) {
- return NULL;
- }
- return new AMediaFormatWrapper(AMediaExtractor_getFileFormat(mAMediaExtractor));
-}
-
-sp<AMediaFormatWrapper> AMediaExtractorWrapper::getTrackFormat(size_t idx) {
- if (mAMediaExtractor == NULL) {
- return NULL;
- }
- return new AMediaFormatWrapper(AMediaExtractor_getTrackFormat(mAMediaExtractor, idx));
-}
-
-status_t AMediaExtractorWrapper::selectTrack(size_t idx) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaExtractor_selectTrack(mAMediaExtractor, idx));
-}
-
-status_t AMediaExtractorWrapper::unselectTrack(size_t idx) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- return translateErrorCode(AMediaExtractor_unselectTrack(mAMediaExtractor, idx));
-}
-
-status_t AMediaExtractorWrapper::selectSingleTrack(size_t idx) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- for (size_t i = 0; i < AMediaExtractor_getTrackCount(mAMediaExtractor); ++i) {
- if (i == idx) {
- media_status_t err = AMediaExtractor_selectTrack(mAMediaExtractor, i);
- if (err != AMEDIA_OK) {
- return translateErrorCode(err);
- }
- } else {
- media_status_t err = AMediaExtractor_unselectTrack(mAMediaExtractor, i);
- if (err != AMEDIA_OK) {
- return translateErrorCode(err);
- }
- }
- }
- return OK;
-}
-
-ssize_t AMediaExtractorWrapper::readSampleData(const sp<ABuffer> &buffer) {
- if (mAMediaExtractor == NULL) {
- return -1;
- }
- return AMediaExtractor_readSampleData(mAMediaExtractor, buffer->data(), buffer->capacity());
-}
-
-ssize_t AMediaExtractorWrapper::getSampleSize() {
- if (mAMediaExtractor == NULL) {
- return 0;
- }
- return AMediaExtractor_getSampleSize(mAMediaExtractor);
-}
-
-uint32_t AMediaExtractorWrapper::getSampleFlags() {
- if (mAMediaExtractor == NULL) {
- return 0;
- }
- return AMediaExtractor_getSampleFlags(mAMediaExtractor);
-}
-
-int AMediaExtractorWrapper::getSampleTrackIndex() {
- if (mAMediaExtractor == NULL) {
- return -1;
- }
- return AMediaExtractor_getSampleTrackIndex(mAMediaExtractor);
-}
-
-int64_t AMediaExtractorWrapper::getSampleTime() {
- if (mAMediaExtractor == NULL) {
- return -1;
- }
- return AMediaExtractor_getSampleTime(mAMediaExtractor);
-}
-
-status_t AMediaExtractorWrapper::getSampleFormat(sp<AMediaFormatWrapper> &formatWrapper) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
- AMediaFormat *format = AMediaFormat_new();
- formatWrapper = new AMediaFormatWrapper(format);
- return translateErrorCode(AMediaExtractor_getSampleFormat(mAMediaExtractor, format));
-}
-
-int64_t AMediaExtractorWrapper::getCachedDuration() {
- if (mAMediaExtractor == NULL) {
- return -1;
- }
- return AMediaExtractor_getCachedDuration(mAMediaExtractor);
-}
-
-bool AMediaExtractorWrapper::advance() {
- if (mAMediaExtractor == NULL) {
- return false;
- }
- return AMediaExtractor_advance(mAMediaExtractor);
-}
-
-status_t AMediaExtractorWrapper::seekTo(int64_t seekPosUs, MediaSource::ReadOptions::SeekMode mode) {
- if (mAMediaExtractor == NULL) {
- return DEAD_OBJECT;
- }
-
- SeekMode aMode;
- switch (mode) {
- case MediaSource::ReadOptions::SEEK_PREVIOUS_SYNC: {
- aMode = AMEDIAEXTRACTOR_SEEK_PREVIOUS_SYNC;
- break;
- }
- case MediaSource::ReadOptions::SEEK_NEXT_SYNC: {
- aMode = AMEDIAEXTRACTOR_SEEK_NEXT_SYNC;
- break;
- }
- default: {
- aMode = AMEDIAEXTRACTOR_SEEK_CLOSEST_SYNC;
- break;
- }
- }
- return AMediaExtractor_seekTo(mAMediaExtractor, seekPosUs, aMode);
-}
-
-PsshInfo* AMediaExtractorWrapper::getPsshInfo() {
- if (mAMediaExtractor == NULL) {
- return NULL;
- }
- return AMediaExtractor_getPsshInfo(mAMediaExtractor);
-}
-
-sp<AMediaCodecCryptoInfoWrapper> AMediaExtractorWrapper::getSampleCryptoInfo() {
- if (mAMediaExtractor == NULL) {
- return NULL;
- }
- AMediaCodecCryptoInfo *cryptoInfo = AMediaExtractor_getSampleCryptoInfo(mAMediaExtractor);
- if (cryptoInfo == NULL) {
- return NULL;
- }
- return new AMediaCodecCryptoInfoWrapper(cryptoInfo);
-}
-
-AMediaDataSourceWrapper::AMediaDataSourceWrapper(const sp<DataSource> &dataSource)
- : mDataSource(dataSource),
- mAMediaDataSource(convertDataSourceToAMediaDataSource(dataSource)) {
-}
-
-AMediaDataSourceWrapper::AMediaDataSourceWrapper(AMediaDataSource *aDataSource)
- : mDataSource(NULL),
- mAMediaDataSource(aDataSource) {
-}
-
-AMediaDataSourceWrapper::~AMediaDataSourceWrapper() {
- if (mAMediaDataSource == NULL) {
- return;
- }
- AMediaDataSource_close(mAMediaDataSource);
- AMediaDataSource_delete(mAMediaDataSource);
- mAMediaDataSource = NULL;
-}
-
-AMediaDataSource* AMediaDataSourceWrapper::getAMediaDataSource() {
- return mAMediaDataSource;
-}
-
-void AMediaDataSourceWrapper::close() {
- AMediaDataSource_close(mAMediaDataSource);
-}
-
-} // namespace android
diff --git a/media/libmedia/Visualizer.cpp b/media/libmedia/Visualizer.cpp
deleted file mode 100644
index 2bf0802..0000000
--- a/media/libmedia/Visualizer.cpp
+++ /dev/null
@@ -1,445 +0,0 @@
-/*
-**
-** Copyright 2010, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "Visualizer"
-#include <utils/Log.h>
-
-#include <stdint.h>
-#include <sys/types.h>
-#include <limits.h>
-
-#include <media/Visualizer.h>
-#include <audio_utils/fixedfft.h>
-#include <utils/Thread.h>
-
-namespace android {
-
-// ---------------------------------------------------------------------------
-
-Visualizer::Visualizer (const String16& opPackageName,
- int32_t priority,
- effect_callback_t cbf,
- void* user,
- audio_session_t sessionId)
- : AudioEffect(SL_IID_VISUALIZATION, opPackageName, NULL, priority, cbf, user, sessionId),
- mCaptureRate(CAPTURE_RATE_DEF),
- mCaptureSize(CAPTURE_SIZE_DEF),
- mSampleRate(44100000),
- mScalingMode(VISUALIZER_SCALING_MODE_NORMALIZED),
- mMeasurementMode(MEASUREMENT_MODE_NONE),
- mCaptureCallBack(NULL),
- mCaptureCbkUser(NULL)
-{
- initCaptureSize();
-}
-
-Visualizer::~Visualizer()
-{
- ALOGV("Visualizer::~Visualizer()");
- setEnabled(false);
- setCaptureCallBack(NULL, NULL, 0, 0);
-}
-
-void Visualizer::release()
-{
- ALOGV("Visualizer::release()");
- setEnabled(false);
- Mutex::Autolock _l(mCaptureLock);
-
- mCaptureThread.clear();
- mCaptureCallBack = NULL;
- mCaptureCbkUser = NULL;
- mCaptureFlags = 0;
- mCaptureRate = 0;
-}
-
-status_t Visualizer::setEnabled(bool enabled)
-{
- Mutex::Autolock _l(mCaptureLock);
-
- sp<CaptureThread> t = mCaptureThread;
- if (t != 0) {
- if (enabled) {
- if (t->exitPending()) {
- mCaptureLock.unlock();
- if (t->requestExitAndWait() == WOULD_BLOCK) {
- mCaptureLock.lock();
- ALOGE("Visualizer::enable() called from thread");
- return INVALID_OPERATION;
- }
- mCaptureLock.lock();
- }
- }
- t->mLock.lock();
- }
-
- status_t status = AudioEffect::setEnabled(enabled);
-
- if (t != 0) {
- if (enabled && status == NO_ERROR) {
- t->run("Visualizer");
- } else {
- t->requestExit();
- }
- }
-
- if (t != 0) {
- t->mLock.unlock();
- }
-
- return status;
-}
-
-status_t Visualizer::setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t flags,
- uint32_t rate)
-{
- if (rate > CAPTURE_RATE_MAX) {
- return BAD_VALUE;
- }
- Mutex::Autolock _l(mCaptureLock);
-
- if (mEnabled) {
- return INVALID_OPERATION;
- }
-
- if (mCaptureThread != 0) {
- mCaptureLock.unlock();
- mCaptureThread->requestExitAndWait();
- mCaptureLock.lock();
- }
-
- mCaptureThread.clear();
- mCaptureCallBack = cbk;
- mCaptureCbkUser = user;
- mCaptureFlags = flags;
- mCaptureRate = rate;
-
- if (cbk != NULL) {
- mCaptureThread = new CaptureThread(this, rate, ((flags & CAPTURE_CALL_JAVA) != 0));
- }
- ALOGV("setCaptureCallBack() rate: %d thread %p flags 0x%08x",
- rate, mCaptureThread.get(), mCaptureFlags);
- return NO_ERROR;
-}
-
-status_t Visualizer::setCaptureSize(uint32_t size)
-{
- if (size > VISUALIZER_CAPTURE_SIZE_MAX ||
- size < VISUALIZER_CAPTURE_SIZE_MIN ||
- popcount(size) != 1) {
- return BAD_VALUE;
- }
-
- Mutex::Autolock _l(mCaptureLock);
- if (mEnabled) {
- return INVALID_OPERATION;
- }
-
- uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
- effect_param_t *p = (effect_param_t *)buf32;
-
- p->psize = sizeof(uint32_t);
- p->vsize = sizeof(uint32_t);
- *(int32_t *)p->data = VISUALIZER_PARAM_CAPTURE_SIZE;
- *((int32_t *)p->data + 1)= size;
- status_t status = setParameter(p);
-
- ALOGV("setCaptureSize size %d status %d p->status %d", size, status, p->status);
-
- if (status == NO_ERROR) {
- status = p->status;
- if (status == NO_ERROR) {
- mCaptureSize = size;
- }
- }
-
- return status;
-}
-
-status_t Visualizer::setScalingMode(uint32_t mode) {
- if ((mode != VISUALIZER_SCALING_MODE_NORMALIZED)
- && (mode != VISUALIZER_SCALING_MODE_AS_PLAYED)) {
- return BAD_VALUE;
- }
-
- Mutex::Autolock _l(mCaptureLock);
-
- uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
- effect_param_t *p = (effect_param_t *)buf32;
-
- p->psize = sizeof(uint32_t);
- p->vsize = sizeof(uint32_t);
- *(int32_t *)p->data = VISUALIZER_PARAM_SCALING_MODE;
- *((int32_t *)p->data + 1)= mode;
- status_t status = setParameter(p);
-
- ALOGV("setScalingMode mode %d status %d p->status %d", mode, status, p->status);
-
- if (status == NO_ERROR) {
- status = p->status;
- if (status == NO_ERROR) {
- mScalingMode = mode;
- }
- }
-
- return status;
-}
-
-status_t Visualizer::setMeasurementMode(uint32_t mode) {
- if ((mode != MEASUREMENT_MODE_NONE)
- //Note: needs to be handled as a mask when more measurement modes are added
- && ((mode & MEASUREMENT_MODE_PEAK_RMS) != mode)) {
- return BAD_VALUE;
- }
-
- Mutex::Autolock _l(mCaptureLock);
-
- uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
- effect_param_t *p = (effect_param_t *)buf32;
-
- p->psize = sizeof(uint32_t);
- p->vsize = sizeof(uint32_t);
- *(int32_t *)p->data = VISUALIZER_PARAM_MEASUREMENT_MODE;
- *((int32_t *)p->data + 1)= mode;
- status_t status = setParameter(p);
-
- ALOGV("setMeasurementMode mode %d status %d p->status %d", mode, status, p->status);
-
- if (status == NO_ERROR) {
- status = p->status;
- if (status == NO_ERROR) {
- mMeasurementMode = mode;
- }
- }
- return status;
-}
-
-status_t Visualizer::getIntMeasurements(uint32_t type, uint32_t number, int32_t *measurements) {
- if (mMeasurementMode == MEASUREMENT_MODE_NONE) {
- ALOGE("Cannot retrieve int measurements, no measurement mode set");
- return INVALID_OPERATION;
- }
- if (!(mMeasurementMode & type)) {
- // measurement type has not been set on this Visualizer
- ALOGE("Cannot retrieve int measurements, requested measurement mode 0x%x not set(0x%x)",
- type, mMeasurementMode);
- return INVALID_OPERATION;
- }
- // only peak+RMS measurement supported
- if ((type != MEASUREMENT_MODE_PEAK_RMS)
- // for peak+RMS measurement, the results are 2 int32_t values
- || (number != 2)) {
- ALOGE("Cannot retrieve int measurements, MEASUREMENT_MODE_PEAK_RMS returns 2 ints, not %d",
- number);
- return BAD_VALUE;
- }
-
- status_t status = NO_ERROR;
- if (mEnabled) {
- uint32_t replySize = number * sizeof(int32_t);
- status = command(VISUALIZER_CMD_MEASURE,
- sizeof(uint32_t) /*cmdSize*/,
- &type /*cmdData*/,
- &replySize, measurements);
- ALOGV("getMeasurements() command returned %d", status);
- if ((status == NO_ERROR) && (replySize == 0)) {
- status = NOT_ENOUGH_DATA;
- }
- } else {
- ALOGV("getMeasurements() disabled");
- return INVALID_OPERATION;
- }
- return status;
-}
-
-status_t Visualizer::getWaveForm(uint8_t *waveform)
-{
- if (waveform == NULL) {
- return BAD_VALUE;
- }
- if (mCaptureSize == 0) {
- return NO_INIT;
- }
-
- status_t status = NO_ERROR;
- if (mEnabled) {
- uint32_t replySize = mCaptureSize;
- status = command(VISUALIZER_CMD_CAPTURE, 0, NULL, &replySize, waveform);
- ALOGV("getWaveForm() command returned %d", status);
- if ((status == NO_ERROR) && (replySize == 0)) {
- status = NOT_ENOUGH_DATA;
- }
- } else {
- ALOGV("getWaveForm() disabled");
- memset(waveform, 0x80, mCaptureSize);
- }
- return status;
-}
-
-status_t Visualizer::getFft(uint8_t *fft)
-{
- if (fft == NULL) {
- return BAD_VALUE;
- }
- if (mCaptureSize == 0) {
- return NO_INIT;
- }
-
- status_t status = NO_ERROR;
- if (mEnabled) {
- uint8_t buf[mCaptureSize];
- status = getWaveForm(buf);
- if (status == NO_ERROR) {
- status = doFft(fft, buf);
- }
- } else {
- memset(fft, 0, mCaptureSize);
- }
- return status;
-}
-
-status_t Visualizer::doFft(uint8_t *fft, uint8_t *waveform)
-{
- int32_t workspace[mCaptureSize >> 1];
- int32_t nonzero = 0;
-
- for (uint32_t i = 0; i < mCaptureSize; i += 2) {
- workspace[i >> 1] =
- ((waveform[i] ^ 0x80) << 24) | ((waveform[i + 1] ^ 0x80) << 8);
- nonzero |= workspace[i >> 1];
- }
-
- if (nonzero) {
- fixed_fft_real(mCaptureSize >> 1, workspace);
- }
-
- for (uint32_t i = 0; i < mCaptureSize; i += 2) {
- short tmp = workspace[i >> 1] >> 21;
- while (tmp > 127 || tmp < -128) tmp >>= 1;
- fft[i] = tmp;
- tmp = workspace[i >> 1];
- tmp >>= 5;
- while (tmp > 127 || tmp < -128) tmp >>= 1;
- fft[i + 1] = tmp;
- }
-
- return NO_ERROR;
-}
-
-void Visualizer::periodicCapture()
-{
- Mutex::Autolock _l(mCaptureLock);
- ALOGV("periodicCapture() %p mCaptureCallBack %p mCaptureFlags 0x%08x",
- this, mCaptureCallBack, mCaptureFlags);
- if (mCaptureCallBack != NULL &&
- (mCaptureFlags & (CAPTURE_WAVEFORM|CAPTURE_FFT)) &&
- mCaptureSize != 0) {
- uint8_t waveform[mCaptureSize];
- status_t status = getWaveForm(waveform);
- if (status != NO_ERROR) {
- return;
- }
- uint8_t fft[mCaptureSize];
- if (mCaptureFlags & CAPTURE_FFT) {
- status = doFft(fft, waveform);
- }
- if (status != NO_ERROR) {
- return;
- }
- uint8_t *wavePtr = NULL;
- uint8_t *fftPtr = NULL;
- uint32_t waveSize = 0;
- uint32_t fftSize = 0;
- if (mCaptureFlags & CAPTURE_WAVEFORM) {
- wavePtr = waveform;
- waveSize = mCaptureSize;
- }
- if (mCaptureFlags & CAPTURE_FFT) {
- fftPtr = fft;
- fftSize = mCaptureSize;
- }
- mCaptureCallBack(mCaptureCbkUser, waveSize, wavePtr, fftSize, fftPtr, mSampleRate);
- }
-}
-
-uint32_t Visualizer::initCaptureSize()
-{
- uint32_t buf32[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
- effect_param_t *p = (effect_param_t *)buf32;
-
- p->psize = sizeof(uint32_t);
- p->vsize = sizeof(uint32_t);
- *(int32_t *)p->data = VISUALIZER_PARAM_CAPTURE_SIZE;
- status_t status = getParameter(p);
-
- if (status == NO_ERROR) {
- status = p->status;
- }
-
- uint32_t size = 0;
- if (status == NO_ERROR) {
- size = *((int32_t *)p->data + 1);
- }
- mCaptureSize = size;
-
- ALOGV("initCaptureSize size %d status %d", mCaptureSize, status);
-
- return size;
-}
-
-void Visualizer::controlStatusChanged(bool controlGranted) {
- if (controlGranted) {
- // this Visualizer instance regained control of the effect, reset the scaling mode
- // and capture size as has been cached through it.
- ALOGV("controlStatusChanged(true) causes effect parameter reset:");
- ALOGV(" scaling mode reset to %d", mScalingMode);
- setScalingMode(mScalingMode);
- ALOGV(" capture size reset to %d", mCaptureSize);
- setCaptureSize(mCaptureSize);
- }
- AudioEffect::controlStatusChanged(controlGranted);
-}
-
-//-------------------------------------------------------------------------
-
-Visualizer::CaptureThread::CaptureThread(Visualizer* receiver, uint32_t captureRate,
- bool bCanCallJava)
- : Thread(bCanCallJava), mReceiver(receiver)
-{
- mSleepTimeUs = 1000000000 / captureRate;
- ALOGV("CaptureThread cstor %p captureRate %d mSleepTimeUs %d", this, captureRate, mSleepTimeUs);
-}
-
-bool Visualizer::CaptureThread::threadLoop()
-{
- ALOGV("CaptureThread %p enter", this);
- sp<Visualizer> receiver = mReceiver.promote();
- if (receiver == NULL) {
- return false;
- }
- while (!exitPending())
- {
- usleep(mSleepTimeUs);
- receiver->periodicCapture();
- }
- ALOGV("CaptureThread %p exiting", this);
- return false;
-}
-
-} // namespace android
diff --git a/media/libmedia/include/media/DataSourceDesc.h b/media/libmedia/include/media/DataSourceDesc.h
deleted file mode 100644
index 4336767..0000000
--- a/media/libmedia/include/media/DataSourceDesc.h
+++ /dev/null
@@ -1,73 +0,0 @@
-/*
- * Copyright 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_DATASOURCEDESC_H
-#define ANDROID_DATASOURCEDESC_H
-
-#include <media/stagefright/foundation/ABase.h>
-#include <utils/RefBase.h>
-#include <utils/KeyedVector.h>
-#include <utils/String8.h>
-
-namespace android {
-
-class DataSource;
-struct MediaHTTPService;
-
-// A binder interface for implementing a stagefright DataSource remotely.
-struct DataSourceDesc : public RefBase {
-public:
- // intentionally less than INT64_MAX
- // keep consistent with JAVA code
- static const int64_t kMaxTimeMs = 0x7ffffffffffffffll / 1000;
- static const int64_t kMaxTimeUs = kMaxTimeMs * 1000;
-
- enum {
- /* No data source has been set yet */
- TYPE_NONE = 0,
- /* data source is type of MediaDataSource */
- TYPE_CALLBACK = 1,
- /* data source is type of FileDescriptor */
- TYPE_FD = 2,
- /* data source is type of Url */
- TYPE_URL = 3,
- };
-
- DataSourceDesc();
-
- int mType;
-
- sp<MediaHTTPService> mHttpService;
- String8 mUrl;
- KeyedVector<String8, String8> mHeaders;
-
- int mFD;
- int64_t mFDOffset;
- int64_t mFDLength;
-
- sp<DataSource> mCallbackSource;
-
- int64_t mId;
- int64_t mStartPositionMs;
- int64_t mEndPositionMs;
-
-private:
- DISALLOW_EVIL_CONSTRUCTORS(DataSourceDesc);
-};
-
-}; // namespace android
-
-#endif // ANDROID_DATASOURCEDESC_H
diff --git a/media/libmedia/include/media/JetPlayer.h b/media/libmedia/include/media/JetPlayer.h
deleted file mode 100644
index bb569bc..0000000
--- a/media/libmedia/include/media/JetPlayer.h
+++ /dev/null
@@ -1,126 +0,0 @@
-/*
- * Copyright (C) 2008 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef JETPLAYER_H_
-#define JETPLAYER_H_
-
-#include <utils/threads.h>
-
-#include <libsonivox/jet.h>
-#include <libsonivox/eas_types.h>
-#include <media/AudioTrack.h>
-#include <media/MidiIoWrapper.h>
-
-
-namespace android {
-
-typedef void (*jetevent_callback)(int eventType, int val1, int val2, void *cookie);
-
-class JetPlayer {
-
-public:
-
- // to keep in sync with the JetPlayer class constants
- // defined in frameworks/base/media/java/android/media/JetPlayer.java
- static const int JET_EVENT = 1;
- static const int JET_USERID_UPDATE = 2;
- static const int JET_NUMQUEUEDSEGMENT_UPDATE = 3;
- static const int JET_PAUSE_UPDATE = 4;
-
- JetPlayer(void *javaJetPlayer,
- int maxTracks = 32,
- int trackBufferSize = 1200);
- ~JetPlayer();
- int init();
- int release();
-
- int loadFromFile(const char* url);
- int loadFromFD(const int fd, const long long offset, const long long length);
- int closeFile();
- int play();
- int pause();
- int queueSegment(int segmentNum, int libNum, int repeatCount, int transpose,
- EAS_U32 muteFlags, EAS_U8 userID);
- int setMuteFlags(EAS_U32 muteFlags, bool sync);
- int setMuteFlag(int trackNum, bool muteFlag, bool sync);
- int triggerClip(int clipId);
- int clearQueue();
-
- void setEventCallback(jetevent_callback callback);
-
- int getMaxTracks() { return mMaxTracks; };
-
-
-private:
- int render();
- void fireUpdateOnStatusChange();
- void fireEventsFromJetQueue();
-
- JetPlayer() {} // no default constructor
- void dump();
- void dumpJetStatus(S_JET_STATUS* pJetStatus);
-
- jetevent_callback mEventCallback;
-
- void* mJavaJetPlayerRef;
- Mutex mMutex; // mutex to sync the render and playback thread with the JET calls
- pid_t mTid;
- Condition mCondition;
- volatile bool mRender;
- bool mPaused;
-
- EAS_STATE mState;
- int* mMemFailedVar;
-
- int mMaxTracks; // max number of MIDI tracks, usually 32
- EAS_DATA_HANDLE mEasData;
- MidiIoWrapper* mIoWrapper;
- EAS_PCM* mAudioBuffer;// EAS renders the MIDI data into this buffer,
- sp<AudioTrack> mAudioTrack; // and we play it in this audio track
- int mTrackBufferSize;
- S_JET_STATUS mJetStatus;
- S_JET_STATUS mPreviousJetStatus;
-
- class JetPlayerThread : public Thread {
- public:
- JetPlayerThread(JetPlayer *player) : mPlayer(player) {
- }
-
- protected:
- virtual ~JetPlayerThread() {}
-
- private:
- JetPlayer *mPlayer;
-
- bool threadLoop() {
- int result;
- result = mPlayer->render();
- return false;
- }
-
- JetPlayerThread(const JetPlayerThread &);
- JetPlayerThread &operator=(const JetPlayerThread &);
- };
-
- sp<JetPlayerThread> mThread;
-
-}; // end class JetPlayer
-
-} // end namespace android
-
-
-
-#endif /*JETPLAYER_H_*/
diff --git a/media/libmedia/include/media/MidiDeviceInfo.h b/media/libmedia/include/media/MidiDeviceInfo.h
deleted file mode 100644
index 5b4a241..0000000
--- a/media/libmedia/include/media/MidiDeviceInfo.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (C) 2016 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_MEDIA_MIDI_DEVICE_INFO_H
-#define ANDROID_MEDIA_MIDI_DEVICE_INFO_H
-
-#include <binder/Parcelable.h>
-#include <binder/PersistableBundle.h>
-#include <utils/String16.h>
-#include <utils/Vector.h>
-
-namespace android {
-namespace media {
-namespace midi {
-
-class MidiDeviceInfo : public Parcelable {
-public:
- MidiDeviceInfo() = default;
- virtual ~MidiDeviceInfo() = default;
- MidiDeviceInfo(const MidiDeviceInfo& midiDeviceInfo) = default;
-
- status_t writeToParcel(Parcel* parcel) const override;
- status_t readFromParcel(const Parcel* parcel) override;
-
- int getType() const { return mType; }
- int getUid() const { return mId; }
- bool isPrivate() const { return mIsPrivate; }
- const Vector<String16>& getInputPortNames() const { return mInputPortNames; }
- const Vector<String16>& getOutputPortNames() const { return mOutputPortNames; }
- String16 getProperty(const char* propertyName);
-
- // The constants need to be kept in sync with MidiDeviceInfo.java
- enum {
- TYPE_USB = 1,
- TYPE_VIRTUAL = 2,
- TYPE_BLUETOOTH = 3,
- };
- static const char* const PROPERTY_NAME;
- static const char* const PROPERTY_MANUFACTURER;
- static const char* const PROPERTY_PRODUCT;
- static const char* const PROPERTY_VERSION;
- static const char* const PROPERTY_SERIAL_NUMBER;
- static const char* const PROPERTY_ALSA_CARD;
- static const char* const PROPERTY_ALSA_DEVICE;
-
- friend bool operator==(const MidiDeviceInfo& lhs, const MidiDeviceInfo& rhs);
- friend bool operator!=(const MidiDeviceInfo& lhs, const MidiDeviceInfo& rhs) {
- return !(lhs == rhs);
- }
-
-private:
- status_t readStringVector(
- const Parcel* parcel, Vector<String16> *vectorPtr, size_t defaultLength);
- status_t writeStringVector(Parcel* parcel, const Vector<String16>& vector) const;
-
- int32_t mType;
- int32_t mId;
- Vector<String16> mInputPortNames;
- Vector<String16> mOutputPortNames;
- os::PersistableBundle mProperties;
- bool mIsPrivate;
-};
-
-} // namespace midi
-} // namespace media
-} // namespace android
-
-#endif // ANDROID_MEDIA_MIDI_DEVICE_INFO_H
diff --git a/media/libmedia/include/media/Visualizer.h b/media/libmedia/include/media/Visualizer.h
deleted file mode 100644
index 8078e36..0000000
--- a/media/libmedia/include/media/Visualizer.h
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#ifndef ANDROID_MEDIA_VISUALIZER_H
-#define ANDROID_MEDIA_VISUALIZER_H
-
-#include <media/AudioEffect.h>
-#include <system/audio_effects/effect_visualizer.h>
-#include <utils/Thread.h>
-
-/**
- * The Visualizer class enables application to retrieve part of the currently playing audio for
- * visualization purpose. It is not an audio recording interface and only returns partial and low
- * quality audio content. However, to protect privacy of certain audio data (e.g voice mail) the use
- * of the visualizer requires the permission android.permission.RECORD_AUDIO.
- * The audio session ID passed to the constructor indicates which audio content should be
- * visualized:
- * - If the session is 0, the audio output mix is visualized
- * - If the session is not 0, the audio from a particular MediaPlayer or AudioTrack
- * using this audio session is visualized
- * Two types of representation of audio content can be captured:
- * - Waveform data: consecutive 8-bit (unsigned) mono samples by using the getWaveForm() method
- * - Frequency data: 8-bit magnitude FFT by using the getFft() method
- *
- * The length of the capture can be retrieved or specified by calling respectively
- * getCaptureSize() and setCaptureSize() methods. Note that the size of the FFT
- * is half of the specified capture size but both sides of the spectrum are returned yielding in a
- * number of bytes equal to the capture size. The capture size must be a power of 2 in the range
- * returned by getMinCaptureSize() and getMaxCaptureSize().
- * In addition to the polling capture mode, a callback mode is also available by installing a
- * callback function by use of the setCaptureCallBack() method. The rate at which the callback
- * is called as well as the type of data returned is specified.
- * Before capturing data, the Visualizer must be enabled by calling the setEnabled() method.
- * When data capture is not needed any more, the Visualizer should be disabled.
- */
-
-
-namespace android {
-
-// ----------------------------------------------------------------------------
-
-class Visualizer: public AudioEffect {
-public:
-
- enum callback_flags {
- CAPTURE_WAVEFORM = 0x00000001, // capture callback returns a PCM wave form
- CAPTURE_FFT = 0x00000002, // apture callback returns a frequency representation
- CAPTURE_CALL_JAVA = 0x00000004 // the callback thread can call java
- };
-
-
- /* Constructor.
- * See AudioEffect constructor for details on parameters.
- */
- Visualizer(const String16& opPackageName,
- int32_t priority = 0,
- effect_callback_t cbf = NULL,
- void* user = NULL,
- audio_session_t sessionId = AUDIO_SESSION_OUTPUT_MIX);
-
- ~Visualizer();
-
- virtual status_t setEnabled(bool enabled);
-
- // maximum capture size in samples
- static uint32_t getMaxCaptureSize() { return VISUALIZER_CAPTURE_SIZE_MAX; }
- // minimum capture size in samples
- static uint32_t getMinCaptureSize() { return VISUALIZER_CAPTURE_SIZE_MIN; }
- // maximum capture rate in millihertz
- static uint32_t getMaxCaptureRate() { return CAPTURE_RATE_MAX; }
-
- // callback used to return periodic PCM or FFT captures to the application. Either one or both
- // types of data are returned (PCM and FFT) according to flags indicated when installing the
- // callback. When a type of data is not present, the corresponding size (waveformSize or
- // fftSize) is 0.
- typedef void (*capture_cbk_t)(void* user,
- uint32_t waveformSize,
- uint8_t *waveform,
- uint32_t fftSize,
- uint8_t *fft,
- uint32_t samplingrate);
-
- // install a callback to receive periodic captures. The capture rate is specified in milliHertz
- // and the capture format is according to flags (see callback_flags).
- status_t setCaptureCallBack(capture_cbk_t cbk, void* user, uint32_t flags, uint32_t rate);
-
- // set the capture size capture size must be a power of two in the range
- // [VISUALIZER_CAPTURE_SIZE_MAX. VISUALIZER_CAPTURE_SIZE_MIN]
- // must be called when the visualizer is not enabled
- status_t setCaptureSize(uint32_t size);
- uint32_t getCaptureSize() { return mCaptureSize; }
-
- // returns the capture rate indicated when installing the callback
- uint32_t getCaptureRate() { return mCaptureRate; }
-
- // returns the sampling rate of the audio being captured
- uint32_t getSamplingRate() { return mSampleRate; }
-
- // set the way volume affects the captured data
- // mode must one of VISUALIZER_SCALING_MODE_NORMALIZED,
- // VISUALIZER_SCALING_MODE_AS_PLAYED
- status_t setScalingMode(uint32_t mode);
- uint32_t getScalingMode() { return mScalingMode; }
-
- // set which measurements are done on the audio buffers processed by the effect.
- // valid measurements (mask): MEASUREMENT_MODE_PEAK_RMS
- status_t setMeasurementMode(uint32_t mode);
- uint32_t getMeasurementMode() { return mMeasurementMode; }
-
- // return a set of int32_t measurements
- status_t getIntMeasurements(uint32_t type, uint32_t number, int32_t *measurements);
-
- // return a capture in PCM 8 bit unsigned format. The size of the capture is equal to
- // getCaptureSize()
- status_t getWaveForm(uint8_t *waveform);
-
- // return a capture in FFT 8 bit signed format. The size of the capture is equal to
- // getCaptureSize() but the length of the FFT is half of the size (both parts of the spectrum
- // are returned
- status_t getFft(uint8_t *fft);
- void release();
-
-protected:
- // from IEffectClient
- virtual void controlStatusChanged(bool controlGranted);
-
-private:
-
- static const uint32_t CAPTURE_RATE_MAX = 20000;
- static const uint32_t CAPTURE_RATE_DEF = 10000;
- static const uint32_t CAPTURE_SIZE_DEF = VISUALIZER_CAPTURE_SIZE_MAX;
-
- /* internal class to handle the callback */
- class CaptureThread : public Thread
- {
- public:
- CaptureThread(Visualizer* visualizer, uint32_t captureRate, bool bCanCallJava = false);
-
- private:
- friend class Visualizer;
- virtual bool threadLoop();
- wp<Visualizer> mReceiver;
- Mutex mLock;
- uint32_t mSleepTimeUs;
- };
-
- status_t doFft(uint8_t *fft, uint8_t *waveform);
- void periodicCapture();
- uint32_t initCaptureSize();
-
- Mutex mCaptureLock;
- uint32_t mCaptureRate;
- uint32_t mCaptureSize;
- uint32_t mSampleRate;
- uint32_t mScalingMode;
- uint32_t mMeasurementMode;
- capture_cbk_t mCaptureCallBack;
- void *mCaptureCbkUser;
- sp<CaptureThread> mCaptureThread;
- uint32_t mCaptureFlags;
-};
-
-
-}; // namespace android
-
-#endif // ANDROID_MEDIA_VISUALIZER_H
diff --git a/media/libmediaplayer2/Android.bp b/media/libmediaplayer2/Android.bp
deleted file mode 100644
index dca6bb6..0000000
--- a/media/libmediaplayer2/Android.bp
+++ /dev/null
@@ -1,129 +0,0 @@
-cc_library_headers {
- name: "libmediaplayer2_headers",
- vendor_available: true,
- export_include_dirs: ["include"],
-}
-
-cc_library_static {
- name: "libmediaplayer2",
-
- srcs: [
- "MediaPlayer2AudioOutput.cpp",
- "mediaplayer2.cpp",
- ],
-
- shared_libs: [
- "libandroid_runtime",
- "libaudioclient",
- "libbinder",
- "libbinder_ndk",
- "libcutils",
- "libgui",
- "liblog",
- "libmedia_omx",
- "libui",
- "libutils",
-
- "libcrypto",
- "libmediametrics",
- "libmediandk",
- "libmediandk_utils",
- "libmediautils",
- "libmemunreachable",
- "libnativewindow",
- "libpowermanager",
- "libstagefright_httplive",
- ],
-
- export_shared_lib_headers: [
- "libaudioclient",
- "libbinder",
- "libgui",
- "libmedia_omx",
- ],
-
- header_libs: [
- "media_plugin_headers",
- ],
-
- include_dirs: [
- "frameworks/base/core/jni",
- ],
-
- static_libs: [
- "libmedia_helper",
- "libmediaplayer2-protos",
- "libmedia_player2_util",
- "libprotobuf-cpp-lite",
- "libstagefright_foundation_without_imemory",
- "libstagefright_nuplayer2",
- "libstagefright_player2",
- "libstagefright_rtsp",
- "libstagefright_timedtext2",
- "libmedia2_jni_core",
- ],
-
- export_include_dirs: [
- "include",
- ],
-
- cflags: [
- "-Werror",
- "-Wno-error=deprecated-declarations",
- "-Wall",
- ],
-
- sanitize: {
- misc_undefined: [
- "unsigned-integer-overflow",
- "signed-integer-overflow",
- ],
- cfi: true,
- },
-}
-
-cc_library {
- name: "libmedia2_jni_core",
-
- srcs: [
- "JavaVMHelper.cpp",
- "JAudioTrack.cpp",
- "JMedia2HTTPService.cpp",
- "JMedia2HTTPConnection.cpp",
- ],
-
- header_libs: [
- "libbinder_headers",
- "libnativehelper_header_only",
- ],
-
- shared_libs: [
- "liblog",
- "libutils",
- "libdl",
- ],
-
- include_dirs: [
- "frameworks/av/media/libmedia/include",
- "frameworks/base/core/jni",
- ],
-
- export_include_dirs: [
- "include",
- ],
-
- cflags: [
- "-Werror",
- "-Wno-error=deprecated-declarations",
- "-Wall",
- ],
-
- sanitize: {
- misc_undefined: [
- "unsigned-integer-overflow",
- "signed-integer-overflow",
- ],
- cfi: true,
- },
-
-}
diff --git a/media/libmediaplayer2/JAudioTrack.cpp b/media/libmediaplayer2/JAudioTrack.cpp
deleted file mode 100644
index fab6c64..0000000
--- a/media/libmediaplayer2/JAudioTrack.cpp
+++ /dev/null
@@ -1,768 +0,0 @@
-/*
- * Copyright 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "JAudioTrack"
-
-#include "media/JAudioAttributes.h"
-#include "media/JAudioFormat.h"
-#include "mediaplayer2/JAudioTrack.h"
-
-#include <android_media_AudioErrors.h>
-#include <mediaplayer2/JavaVMHelper.h>
-
-namespace android {
-
-// TODO: Store Java class/methodID as a member variable in the class.
-// TODO: Add NULL && Exception checks after every JNI call.
-JAudioTrack::JAudioTrack( // < Usages of the arguments are below >
- uint32_t sampleRate, // AudioFormat && bufferSizeInBytes
- audio_format_t format, // AudioFormat && bufferSizeInBytes
- audio_channel_mask_t channelMask, // AudioFormat && bufferSizeInBytes
- callback_t cbf, // Offload
- void* user, // Offload
- size_t frameCount, // bufferSizeInBytes
- int32_t sessionId, // AudioTrack
- const jobject attributes, // AudioAttributes
- float maxRequiredSpeed) { // bufferSizeInBytes
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jclass jAudioTrackCls = env->FindClass("android/media/AudioTrack");
- mAudioTrackCls = reinterpret_cast<jclass>(env->NewGlobalRef(jAudioTrackCls));
- env->DeleteLocalRef(jAudioTrackCls);
-
- maxRequiredSpeed = std::min(std::max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);
-
- int bufferSizeInBytes = 0;
- if (sampleRate == 0 || frameCount > 0) {
- // Manually calculate buffer size.
- bufferSizeInBytes = audio_channel_count_from_out_mask(channelMask)
- * audio_bytes_per_sample(format) * (frameCount > 0 ? frameCount : 1);
- } else if (sampleRate > 0) {
- // Call Java AudioTrack::getMinBufferSize().
- jmethodID jGetMinBufferSize =
- env->GetStaticMethodID(mAudioTrackCls, "getMinBufferSize", "(III)I");
- bufferSizeInBytes = env->CallStaticIntMethod(mAudioTrackCls, jGetMinBufferSize,
- sampleRate, outChannelMaskFromNative(channelMask), audioFormatFromNative(format));
- }
- bufferSizeInBytes = (int) (bufferSizeInBytes * maxRequiredSpeed);
-
- // Create a Java AudioTrack object through its Builder.
- jclass jBuilderCls = env->FindClass("android/media/AudioTrack$Builder");
- jmethodID jBuilderCtor = env->GetMethodID(jBuilderCls, "<init>", "()V");
- jobject jBuilderObj = env->NewObject(jBuilderCls, jBuilderCtor);
-
- {
- sp<JObjectHolder> audioAttributesObj;
- if (attributes != NULL) {
- audioAttributesObj = new JObjectHolder(attributes);
- } else {
- audioAttributesObj = new JObjectHolder(
- JAudioAttributes::createAudioAttributesObj(env, NULL));
- }
- jmethodID jSetAudioAttributes = env->GetMethodID(jBuilderCls, "setAudioAttributes",
- "(Landroid/media/AudioAttributes;)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj,
- jSetAudioAttributes, audioAttributesObj->getJObject());
- }
-
- jmethodID jSetAudioFormat = env->GetMethodID(jBuilderCls, "setAudioFormat",
- "(Landroid/media/AudioFormat;)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj, jSetAudioFormat,
- JAudioFormat::createAudioFormatObj(env, sampleRate, format, channelMask));
-
- jmethodID jSetBufferSizeInBytes = env->GetMethodID(jBuilderCls, "setBufferSizeInBytes",
- "(I)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj, jSetBufferSizeInBytes, bufferSizeInBytes);
-
- // We only use streaming mode of Java AudioTrack.
- jfieldID jModeStream = env->GetStaticFieldID(mAudioTrackCls, "MODE_STREAM", "I");
- jint transferMode = env->GetStaticIntField(mAudioTrackCls, jModeStream);
- jmethodID jSetTransferMode = env->GetMethodID(jBuilderCls, "setTransferMode",
- "(I)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj, jSetTransferMode,
- transferMode /* Java AudioTrack::MODE_STREAM */);
-
- if (sessionId != 0) {
- jmethodID jSetSessionId = env->GetMethodID(jBuilderCls, "setSessionId",
- "(I)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj, jSetSessionId, sessionId);
- }
-
- mFlags = AUDIO_OUTPUT_FLAG_NONE;
- if (cbf != NULL) {
- jmethodID jSetOffloadedPlayback = env->GetMethodID(jBuilderCls, "setOffloadedPlayback",
- "(Z)Landroid/media/AudioTrack$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderObj, jSetOffloadedPlayback, true);
- mFlags = AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD;
- }
-
- jmethodID jBuild = env->GetMethodID(jBuilderCls, "build", "()Landroid/media/AudioTrack;");
- jobject jAudioTrackObj = env->CallObjectMethod(jBuilderObj, jBuild);
- mAudioTrackObj = reinterpret_cast<jobject>(env->NewGlobalRef(jAudioTrackObj));
- env->DeleteLocalRef(jBuilderObj);
-
- if (cbf != NULL) {
- // Set offload mode callback
- jobject jStreamEventCallbackObj = createStreamEventCallback(cbf, user);
- jobject jExecutorObj = createCallbackExecutor();
- jmethodID jSetStreamEventCallback = env->GetMethodID(
- jAudioTrackCls,
- "setStreamEventCallback",
- "(Ljava/util/concurrent/Executor;Landroid/media/AudioTrack$StreamEventCallback;)V");
- env->CallVoidMethod(
- mAudioTrackObj, jSetStreamEventCallback, jExecutorObj, jStreamEventCallbackObj);
- }
-}
-
-JAudioTrack::~JAudioTrack() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- env->DeleteGlobalRef(mAudioTrackCls);
- env->DeleteGlobalRef(mAudioTrackObj);
-}
-
-size_t JAudioTrack::frameCount() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetBufferSizeInFrames = env->GetMethodID(
- mAudioTrackCls, "getBufferSizeInFrames", "()I");
- return env->CallIntMethod(mAudioTrackObj, jGetBufferSizeInFrames);
-}
-
-size_t JAudioTrack::channelCount() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetChannelCount = env->GetMethodID(mAudioTrackCls, "getChannelCount", "()I");
- return env->CallIntMethod(mAudioTrackObj, jGetChannelCount);
-}
-
-uint32_t JAudioTrack::latency() {
- // TODO: Currently hard-coded as returning zero.
- return 0;
-}
-
-status_t JAudioTrack::getPosition(uint32_t *position) {
- if (position == NULL) {
- return BAD_VALUE;
- }
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetPlaybackHeadPosition = env->GetMethodID(
- mAudioTrackCls, "getPlaybackHeadPosition", "()I");
- *position = env->CallIntMethod(mAudioTrackObj, jGetPlaybackHeadPosition);
-
- return NO_ERROR;
-}
-
-status_t JAudioTrack::getTimestamp(AudioTimestamp& timestamp) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jclass jAudioTimeStampCls = env->FindClass("android/media/AudioTimestamp");
- jobject jAudioTimeStampObj = env->AllocObject(jAudioTimeStampCls);
-
- jfieldID jFramePosition = env->GetFieldID(jAudioTimeStampCls, "framePosition", "J");
- jfieldID jNanoTime = env->GetFieldID(jAudioTimeStampCls, "nanoTime", "J");
-
- jmethodID jGetTimestamp = env->GetMethodID(mAudioTrackCls,
- "getTimestamp", "(Landroid/media/AudioTimestamp;)Z");
- bool success = env->CallBooleanMethod(mAudioTrackObj, jGetTimestamp, jAudioTimeStampObj);
-
- if (!success) {
- return NO_INIT;
- }
-
- long long framePosition = env->GetLongField(jAudioTimeStampObj, jFramePosition);
- long long nanoTime = env->GetLongField(jAudioTimeStampObj, jNanoTime);
-
- struct timespec ts;
- const long long secondToNano = 1000000000LL; // 1E9
- ts.tv_sec = nanoTime / secondToNano;
- ts.tv_nsec = nanoTime % secondToNano;
- timestamp.mTime = ts;
- timestamp.mPosition = (uint32_t) framePosition;
-
- return NO_ERROR;
-}
-
-status_t JAudioTrack::getTimestamp(ExtendedTimestamp *timestamp __unused) {
- // TODO: Implement this after appropriate Java AudioTrack method is available.
- return NO_ERROR;
-}
-
-status_t JAudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate) {
- // TODO: existing native AudioTrack returns INVALID_OPERATION on offload/direct/fast tracks.
- // Should we do the same thing?
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jclass jPlaybackParamsCls = env->FindClass("android/media/PlaybackParams");
- jmethodID jPlaybackParamsCtor = env->GetMethodID(jPlaybackParamsCls, "<init>", "()V");
- jobject jPlaybackParamsObj = env->NewObject(jPlaybackParamsCls, jPlaybackParamsCtor);
-
- jmethodID jSetAudioFallbackMode = env->GetMethodID(
- jPlaybackParamsCls, "setAudioFallbackMode", "(I)Landroid/media/PlaybackParams;");
- jPlaybackParamsObj = env->CallObjectMethod(
- jPlaybackParamsObj, jSetAudioFallbackMode, playbackRate.mFallbackMode);
-
- jmethodID jSetAudioStretchMode = env->GetMethodID(
- jPlaybackParamsCls, "setAudioStretchMode", "(I)Landroid/media/PlaybackParams;");
- jPlaybackParamsObj = env->CallObjectMethod(
- jPlaybackParamsObj, jSetAudioStretchMode, playbackRate.mStretchMode);
-
- jmethodID jSetPitch = env->GetMethodID(
- jPlaybackParamsCls, "setPitch", "(F)Landroid/media/PlaybackParams;");
- jPlaybackParamsObj = env->CallObjectMethod(jPlaybackParamsObj, jSetPitch, playbackRate.mPitch);
-
- jmethodID jSetSpeed = env->GetMethodID(
- jPlaybackParamsCls, "setSpeed", "(F)Landroid/media/PlaybackParams;");
- jPlaybackParamsObj = env->CallObjectMethod(jPlaybackParamsObj, jSetSpeed, playbackRate.mSpeed);
-
-
- // Set this Java PlaybackParams object into Java AudioTrack.
- jmethodID jSetPlaybackParams = env->GetMethodID(
- mAudioTrackCls, "setPlaybackParams", "(Landroid/media/PlaybackParams;)V");
- env->CallVoidMethod(mAudioTrackObj, jSetPlaybackParams, jPlaybackParamsObj);
- // TODO: Should we catch the Java IllegalArgumentException?
-
- return NO_ERROR;
-}
-
-const AudioPlaybackRate JAudioTrack::getPlaybackRate() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jmethodID jGetPlaybackParams = env->GetMethodID(
- mAudioTrackCls, "getPlaybackParams", "()Landroid/media/PlaybackParams;");
- jobject jPlaybackParamsObj = env->CallObjectMethod(mAudioTrackObj, jGetPlaybackParams);
-
- AudioPlaybackRate playbackRate;
- jclass jPlaybackParamsCls = env->FindClass("android/media/PlaybackParams");
-
- jmethodID jGetAudioFallbackMode = env->GetMethodID(
- jPlaybackParamsCls, "getAudioFallbackMode", "()I");
- // TODO: Should we enable passing AUDIO_TIMESTRETCH_FALLBACK_CUT_REPEAT?
- // The enum is internal only, so it is not defined in PlaybackParmas.java.
- // TODO: Is this right way to convert an int to an enum?
- playbackRate.mFallbackMode = static_cast<AudioTimestretchFallbackMode>(
- env->CallIntMethod(jPlaybackParamsObj, jGetAudioFallbackMode));
-
- jmethodID jGetAudioStretchMode = env->GetMethodID(
- jPlaybackParamsCls, "getAudioStretchMode", "()I");
- playbackRate.mStretchMode = static_cast<AudioTimestretchStretchMode>(
- env->CallIntMethod(jPlaybackParamsObj, jGetAudioStretchMode));
-
- jmethodID jGetPitch = env->GetMethodID(jPlaybackParamsCls, "getPitch", "()F");
- playbackRate.mPitch = env->CallFloatMethod(jPlaybackParamsObj, jGetPitch);
-
- jmethodID jGetSpeed = env->GetMethodID(jPlaybackParamsCls, "getSpeed", "()F");
- playbackRate.mSpeed = env->CallFloatMethod(jPlaybackParamsObj, jGetSpeed);
-
- return playbackRate;
-}
-
-media::VolumeShaper::Status JAudioTrack::applyVolumeShaper(
- const sp<media::VolumeShaper::Configuration>& configuration,
- const sp<media::VolumeShaper::Operation>& operation) {
-
- jobject jConfigurationObj = createVolumeShaperConfigurationObj(configuration);
- jobject jOperationObj = createVolumeShaperOperationObj(operation);
-
- if (jConfigurationObj == NULL || jOperationObj == NULL) {
- return media::VolumeShaper::Status(BAD_VALUE);
- }
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jmethodID jCreateVolumeShaper = env->GetMethodID(mAudioTrackCls, "createVolumeShaper",
- "(Landroid/media/VolumeShaper$Configuration;)Landroid/media/VolumeShaper;");
- jobject jVolumeShaperObj = env->CallObjectMethod(
- mAudioTrackObj, jCreateVolumeShaper, jConfigurationObj);
-
- jclass jVolumeShaperCls = env->FindClass("android/media/VolumeShaper");
- jmethodID jApply = env->GetMethodID(jVolumeShaperCls, "apply",
- "(Landroid/media/VolumeShaper$Operation;)V");
- env->CallVoidMethod(jVolumeShaperObj, jApply, jOperationObj);
-
- return media::VolumeShaper::Status(NO_ERROR);
-}
-
-status_t JAudioTrack::setAuxEffectSendLevel(float level) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jSetAuxEffectSendLevel = env->GetMethodID(
- mAudioTrackCls, "setAuxEffectSendLevel", "(F)I");
- int result = env->CallIntMethod(mAudioTrackObj, jSetAuxEffectSendLevel, level);
- return javaToNativeStatus(result);
-}
-
-status_t JAudioTrack::attachAuxEffect(int effectId) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jAttachAuxEffect = env->GetMethodID(mAudioTrackCls, "attachAuxEffect", "(I)I");
- int result = env->CallIntMethod(mAudioTrackObj, jAttachAuxEffect, effectId);
- return javaToNativeStatus(result);
-}
-
-status_t JAudioTrack::setVolume(float left, float right) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- // TODO: Java setStereoVolume is deprecated. Do we really need this method?
- jmethodID jSetStereoVolume = env->GetMethodID(mAudioTrackCls, "setStereoVolume", "(FF)I");
- int result = env->CallIntMethod(mAudioTrackObj, jSetStereoVolume, left, right);
- return javaToNativeStatus(result);
-}
-
-status_t JAudioTrack::setVolume(float volume) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jSetVolume = env->GetMethodID(mAudioTrackCls, "setVolume", "(F)I");
- int result = env->CallIntMethod(mAudioTrackObj, jSetVolume, volume);
- return javaToNativeStatus(result);
-}
-
-status_t JAudioTrack::start() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jPlay = env->GetMethodID(mAudioTrackCls, "play", "()V");
- // TODO: Should we catch the Java IllegalStateException from play()?
- env->CallVoidMethod(mAudioTrackObj, jPlay);
- return NO_ERROR;
-}
-
-ssize_t JAudioTrack::write(const void* buffer, size_t size, bool blocking) {
- if (buffer == NULL) {
- return BAD_VALUE;
- }
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jbyteArray jAudioData = env->NewByteArray(size);
- env->SetByteArrayRegion(jAudioData, 0, size, (jbyte *) buffer);
-
- jclass jByteBufferCls = env->FindClass("java/nio/ByteBuffer");
- jmethodID jWrap = env->GetStaticMethodID(jByteBufferCls, "wrap", "([B)Ljava/nio/ByteBuffer;");
- jobject jByteBufferObj = env->CallStaticObjectMethod(jByteBufferCls, jWrap, jAudioData);
-
- int writeMode = 0;
- if (blocking) {
- jfieldID jWriteBlocking = env->GetStaticFieldID(mAudioTrackCls, "WRITE_BLOCKING", "I");
- writeMode = env->GetStaticIntField(mAudioTrackCls, jWriteBlocking);
- } else {
- jfieldID jWriteNonBlocking = env->GetStaticFieldID(
- mAudioTrackCls, "WRITE_NON_BLOCKING", "I");
- writeMode = env->GetStaticIntField(mAudioTrackCls, jWriteNonBlocking);
- }
-
- jmethodID jWrite = env->GetMethodID(mAudioTrackCls, "write", "(Ljava/nio/ByteBuffer;II)I");
- int result = env->CallIntMethod(mAudioTrackObj, jWrite, jByteBufferObj, size, writeMode);
-
- if (result >= 0) {
- return result;
- } else {
- return javaToNativeStatus(result);
- }
-}
-
-void JAudioTrack::stop() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jStop = env->GetMethodID(mAudioTrackCls, "stop", "()V");
- env->CallVoidMethod(mAudioTrackObj, jStop);
- // TODO: Should we catch IllegalStateException?
-}
-
-// TODO: Is the right implementation?
-bool JAudioTrack::stopped() const {
- return !isPlaying();
-}
-
-void JAudioTrack::flush() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jFlush = env->GetMethodID(mAudioTrackCls, "flush", "()V");
- env->CallVoidMethod(mAudioTrackObj, jFlush);
-}
-
-void JAudioTrack::pause() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jPause = env->GetMethodID(mAudioTrackCls, "pause", "()V");
- env->CallVoidMethod(mAudioTrackObj, jPause);
- // TODO: Should we catch IllegalStateException?
-}
-
-bool JAudioTrack::isPlaying() const {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetPlayState = env->GetMethodID(mAudioTrackCls, "getPlayState", "()I");
- int currentPlayState = env->CallIntMethod(mAudioTrackObj, jGetPlayState);
-
- // TODO: In Java AudioTrack, there is no STOPPING state.
- // This means while stopping, isPlaying() will return different value in two class.
- // - in existing native AudioTrack: true
- // - in JAudioTrack: false
- // If not okay, also modify the implementation of stopped().
- jfieldID jPlayStatePlaying = env->GetStaticFieldID(mAudioTrackCls, "PLAYSTATE_PLAYING", "I");
- int statePlaying = env->GetStaticIntField(mAudioTrackCls, jPlayStatePlaying);
- return currentPlayState == statePlaying;
-}
-
-uint32_t JAudioTrack::getSampleRate() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetSampleRate = env->GetMethodID(mAudioTrackCls, "getSampleRate", "()I");
- return env->CallIntMethod(mAudioTrackObj, jGetSampleRate);
-}
-
-status_t JAudioTrack::getBufferDurationInUs(int64_t *duration) {
- if (duration == nullptr) {
- return BAD_VALUE;
- }
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetBufferSizeInFrames = env->GetMethodID(
- mAudioTrackCls, "getBufferSizeInFrames", "()I");
- int bufferSizeInFrames = env->CallIntMethod(mAudioTrackObj, jGetBufferSizeInFrames);
-
- const double secondToMicro = 1000000LL; // 1E6
- int sampleRate = JAudioTrack::getSampleRate();
- float speed = JAudioTrack::getPlaybackRate().mSpeed;
-
- *duration = (int64_t) (bufferSizeInFrames * secondToMicro / (sampleRate * speed));
- return NO_ERROR;
-}
-
-audio_format_t JAudioTrack::format() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetAudioFormat = env->GetMethodID(mAudioTrackCls, "getAudioFormat", "()I");
- int javaFormat = env->CallIntMethod(mAudioTrackObj, jGetAudioFormat);
- return audioFormatToNative(javaFormat);
-}
-
-size_t JAudioTrack::frameSize() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetFormat = env->GetMethodID(mAudioTrackCls,
- "getFormat", "()Landroid/media/AudioFormat;");
- jobject jAudioFormatObj = env->CallObjectMethod(mAudioTrackObj, jGetFormat);
-
- jclass jAudioFormatCls = env->FindClass("android/media/AudioFormat");
- jmethodID jGetFrameSizeInBytes = env->GetMethodID(
- jAudioFormatCls, "getFrameSizeInBytes", "()I");
- jint javaFrameSizeInBytes = env->CallIntMethod(jAudioFormatObj, jGetFrameSizeInBytes);
-
- return (size_t)javaFrameSizeInBytes;
-}
-
-status_t JAudioTrack::dump(int fd, const Vector<String16>& args __unused) const
-{
- String8 result;
-
- result.append(" JAudioTrack::dump\n");
-
- // TODO: Remove logs that includes unavailable information from below.
-// result.appendFormat(" status(%d), state(%d), session Id(%d), flags(%#x)\n",
-// mStatus, mState, mSessionId, mFlags);
-// result.appendFormat(" format(%#x), channel mask(%#x), channel count(%u)\n",
-// format(), mChannelMask, channelCount());
-// result.appendFormat(" sample rate(%u), original sample rate(%u), speed(%f)\n",
-// getSampleRate(), mOriginalSampleRate, mPlaybackRate.mSpeed);
-// result.appendFormat(" frame count(%zu), req. frame count(%zu)\n",
-// frameCount(), mReqFrameCount);
-// result.appendFormat(" notif. frame count(%u), req. notif. frame count(%u),"
-// " req. notif. per buff(%u)\n",
-// mNotificationFramesAct, mNotificationFramesReq, mNotificationsPerBufferReq);
-// result.appendFormat(" latency (%d), selected device Id(%d), routed device Id(%d)\n",
-// latency(), mSelectedDeviceId, getRoutedDeviceId());
-// result.appendFormat(" output(%d) AF latency (%u) AF frame count(%zu) AF SampleRate(%u)\n",
-// mOutput, mAfLatency, mAfFrameCount, mAfSampleRate);
- ::write(fd, result.string(), result.size());
- return NO_ERROR;
-}
-
-jobject JAudioTrack::getRoutedDevice() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetRoutedDevice = env->GetMethodID(mAudioTrackCls, "getRoutedDevice",
- "()Landroid/media/AudioDeviceInfo;");
- return env->CallObjectMethod(mAudioTrackObj, jGetRoutedDevice);
-}
-
-int32_t JAudioTrack::getAudioSessionId() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetAudioSessionId = env->GetMethodID(mAudioTrackCls, "getAudioSessionId", "()I");
- jint sessionId = env->CallIntMethod(mAudioTrackObj, jGetAudioSessionId);
- return sessionId;
-}
-
-status_t JAudioTrack::setPreferredDevice(jobject device) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jSetPreferredDeviceId = env->GetMethodID(mAudioTrackCls, "setPreferredDevice",
- "(Landroid/media/AudioDeviceInfo;)Z");
- jboolean result = env->CallBooleanMethod(mAudioTrackObj, jSetPreferredDeviceId, device);
- return result == true ? NO_ERROR : BAD_VALUE;
-}
-
-audio_stream_type_t JAudioTrack::getAudioStreamType() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jGetAudioAttributes = env->GetMethodID(mAudioTrackCls, "getAudioAttributes",
- "()Landroid/media/AudioAttributes;");
- jobject jAudioAttributes = env->CallObjectMethod(mAudioTrackObj, jGetAudioAttributes);
- jclass jAudioAttributesCls = env->FindClass("android/media/AudioAttributes");
- jmethodID jGetVolumeControlStream = env->GetMethodID(jAudioAttributesCls,
- "getVolumeControlStream", "()I");
- int javaAudioStreamType = env->CallIntMethod(jAudioAttributes, jGetVolumeControlStream);
- return (audio_stream_type_t)javaAudioStreamType;
-}
-
-status_t JAudioTrack::pendingDuration(int32_t *msec) {
- if (msec == nullptr) {
- return BAD_VALUE;
- }
-
- bool isPurePcmData = audio_is_linear_pcm(format()) && (getFlags() & AUDIO_FLAG_HW_AV_SYNC) == 0;
- if (!isPurePcmData) {
- return INVALID_OPERATION;
- }
-
- // TODO: Need to know the difference btw. client and server time.
- // If getTimestamp(ExtendedTimestamp) is ready, and un-comment below and modify appropriately.
- // (copied from AudioTrack.cpp)
-
-// ExtendedTimestamp ets;
-// ExtendedTimestamp::LOCATION location = ExtendedTimestamp::LOCATION_SERVER;
-// if (getTimestamp_l(&ets) == OK && ets.mTimeNs[location] > 0) {
-// int64_t diff = ets.mPosition[ExtendedTimestamp::LOCATION_CLIENT]
-// - ets.mPosition[location];
-// if (diff < 0) {
-// *msec = 0;
-// } else {
-// // ms is the playback time by frames
-// int64_t ms = (int64_t)((double)diff * 1000 /
-// ((double)mSampleRate * mPlaybackRate.mSpeed));
-// // clockdiff is the timestamp age (negative)
-// int64_t clockdiff = (mState != STATE_ACTIVE) ? 0 :
-// ets.mTimeNs[location]
-// + ets.mTimebaseOffset[ExtendedTimestamp::TIMEBASE_MONOTONIC]
-// - systemTime(SYSTEM_TIME_MONOTONIC);
-//
-// //ALOGV("ms: %lld clockdiff: %lld", (long long)ms, (long long)clockdiff);
-// static const int NANOS_PER_MILLIS = 1000000;
-// *msec = (int32_t)(ms + clockdiff / NANOS_PER_MILLIS);
-// }
-// return NO_ERROR;
-// }
-
- return NO_ERROR;
-}
-
-status_t JAudioTrack::addAudioDeviceCallback(jobject listener, jobject handler) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jAddOnRoutingChangedListener = env->GetMethodID(mAudioTrackCls,
- "addOnRoutingChangedListener",
- "(Landroid/media/AudioRouting$OnRoutingChangedListener;Landroid/os/Handler;)V");
- env->CallVoidMethod(mAudioTrackObj, jAddOnRoutingChangedListener, listener, handler);
- return NO_ERROR;
-}
-
-status_t JAudioTrack::removeAudioDeviceCallback(jobject listener) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jmethodID jRemoveOnRoutingChangedListener = env->GetMethodID(mAudioTrackCls,
- "removeOnRoutingChangedListener",
- "(Landroid/media/AudioRouting$OnRoutingChangedListener;)V");
- env->CallVoidMethod(mAudioTrackObj, jRemoveOnRoutingChangedListener, listener);
- return NO_ERROR;
-}
-
-void JAudioTrack::registerRoutingDelegates(
- Vector<std::pair<sp<JObjectHolder>, sp<JObjectHolder>>>& routingDelegates) {
- for (auto it = routingDelegates.begin(); it != routingDelegates.end(); it++) {
- addAudioDeviceCallback(it->second->getJObject(), getHandler(it->second->getJObject()));
- }
-}
-
-/////////////////////////////////////////////////////////////
-/// Static methods begin ///
-/////////////////////////////////////////////////////////////
-jobject JAudioTrack::getListener(const jobject routingDelegateObj) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass jRoutingDelegateCls = env->FindClass("android/media/RoutingDelegate");
- jmethodID jGetListener = env->GetMethodID(jRoutingDelegateCls,
- "getListener", "()Landroid/media/AudioRouting$OnRoutingChangedListener;");
- return env->CallObjectMethod(routingDelegateObj, jGetListener);
-}
-
-jobject JAudioTrack::getHandler(const jobject routingDelegateObj) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass jRoutingDelegateCls = env->FindClass("android/media/RoutingDelegate");
- jmethodID jGetHandler = env->GetMethodID(jRoutingDelegateCls,
- "getHandler", "()Landroid/os/Handler;");
- return env->CallObjectMethod(routingDelegateObj, jGetHandler);
-}
-
-jobject JAudioTrack::findByKey(
- Vector<std::pair<sp<JObjectHolder>, sp<JObjectHolder>>>& mp, const jobject key) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- for (auto it = mp.begin(); it != mp.end(); it++) {
- if (env->IsSameObject(it->first->getJObject(), key)) {
- return it->second->getJObject();
- }
- }
- return nullptr;
-}
-
-void JAudioTrack::eraseByKey(
- Vector<std::pair<sp<JObjectHolder>, sp<JObjectHolder>>>& mp, const jobject key) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- for (auto it = mp.begin(); it != mp.end(); it++) {
- if (env->IsSameObject(it->first->getJObject(), key)) {
- mp.erase(it);
- return;
- }
- }
-}
-
-/////////////////////////////////////////////////////////////
-/// Private method begins ///
-/////////////////////////////////////////////////////////////
-
-jobject JAudioTrack::createVolumeShaperConfigurationObj(
- const sp<media::VolumeShaper::Configuration>& config) {
-
- // TODO: Java VolumeShaper's setId() / setOptionFlags() are hidden.
- if (config == NULL || config->getType() == media::VolumeShaper::Configuration::TYPE_ID) {
- return NULL;
- }
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- // Referenced "android_media_VolumeShaper.h".
- jfloatArray xarray = nullptr;
- jfloatArray yarray = nullptr;
- if (config->getType() == media::VolumeShaper::Configuration::TYPE_SCALE) {
- // convert curve arrays
- xarray = env->NewFloatArray(config->size());
- yarray = env->NewFloatArray(config->size());
- float * const x = env->GetFloatArrayElements(xarray, nullptr /* isCopy */);
- float * const y = env->GetFloatArrayElements(yarray, nullptr /* isCopy */);
- float *xptr = x, *yptr = y;
- for (const auto &pt : *config.get()) {
- *xptr++ = pt.first;
- *yptr++ = pt.second;
- }
- env->ReleaseFloatArrayElements(xarray, x, 0 /* mode */);
- env->ReleaseFloatArrayElements(yarray, y, 0 /* mode */);
- }
-
- jclass jBuilderCls = env->FindClass("android/media/VolumeShaper$Configuration$Builder");
- jmethodID jBuilderCtor = env->GetMethodID(jBuilderCls, "<init>", "()V");
- jobject jBuilderObj = env->NewObject(jBuilderCls, jBuilderCtor);
-
- jmethodID jSetDuration = env->GetMethodID(jBuilderCls, "setDuration",
- "(L)Landroid/media/VolumeShaper$Configuration$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jSetDuration, (jlong) config->getDurationMs());
-
- jmethodID jSetInterpolatorType = env->GetMethodID(jBuilderCls, "setInterpolatorType",
- "(I)Landroid/media/VolumeShaper$Configuration$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jSetInterpolatorType,
- config->getInterpolatorType());
-
- jmethodID jSetCurve = env->GetMethodID(jBuilderCls, "setCurve",
- "([F[F)Landroid/media/VolumeShaper$Configuration$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jSetCurve, xarray, yarray);
-
- jmethodID jBuild = env->GetMethodID(jBuilderCls, "build",
- "()Landroid/media/VolumeShaper$Configuration;");
- return env->CallObjectMethod(jBuilderObj, jBuild);
-}
-
-jobject JAudioTrack::createVolumeShaperOperationObj(
- const sp<media::VolumeShaper::Operation>& operation) {
-
- JNIEnv *env = JavaVMHelper::getJNIEnv();
-
- jclass jBuilderCls = env->FindClass("android/media/VolumeShaper$Operation$Builder");
- jmethodID jBuilderCtor = env->GetMethodID(jBuilderCls, "<init>", "()V");
- jobject jBuilderObj = env->NewObject(jBuilderCls, jBuilderCtor);
-
- // Set XOffset
- jmethodID jSetXOffset = env->GetMethodID(jBuilderCls, "setXOffset",
- "(F)Landroid/media/VolumeShaper$Operation$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jSetXOffset, operation->getXOffset());
-
- int32_t flags = operation->getFlags();
-
- if (operation->getReplaceId() >= 0) {
- jmethodID jReplace = env->GetMethodID(jBuilderCls, "replace",
- "(IB)Landroid/media/VolumeShaper$Operation$Builder;");
- bool join = (flags | media::VolumeShaper::Operation::FLAG_JOIN) != 0;
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jReplace, operation->getReplaceId(), join);
- }
-
- if (flags | media::VolumeShaper::Operation::FLAG_REVERSE) {
- jmethodID jReverse = env->GetMethodID(jBuilderCls, "reverse",
- "()Landroid/media/VolumeShaper$Operation$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jReverse);
- }
-
- // TODO: VolumeShaper Javadoc says "Do not call terminate() directly". Can we call this?
- if (flags | media::VolumeShaper::Operation::FLAG_TERMINATE) {
- jmethodID jTerminate = env->GetMethodID(jBuilderCls, "terminate",
- "()Landroid/media/VolumeShaper$Operation$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jTerminate);
- }
-
- if (flags | media::VolumeShaper::Operation::FLAG_DELAY) {
- jmethodID jDefer = env->GetMethodID(jBuilderCls, "defer",
- "()Landroid/media/VolumeShaper$Operation$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jDefer);
- }
-
- if (flags | media::VolumeShaper::Operation::FLAG_CREATE_IF_NECESSARY) {
- jmethodID jCreateIfNeeded = env->GetMethodID(jBuilderCls, "createIfNeeded",
- "()Landroid/media/VolumeShaper$Operation$Builder;");
- jBuilderObj = env->CallObjectMethod(jBuilderCls, jCreateIfNeeded);
- }
-
- // TODO: Handle error case (can it be NULL?)
- jmethodID jBuild = env->GetMethodID(jBuilderCls, "build",
- "()Landroid/media/VolumeShaper$Operation;");
- return env->CallObjectMethod(jBuilderObj, jBuild);
-}
-
-jobject JAudioTrack::createStreamEventCallback(callback_t cbf, void* user) {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass jCallbackCls = env->FindClass("android/media/MediaPlayer2$StreamEventCallback");
- jmethodID jCallbackCtor = env->GetMethodID(jCallbackCls, "<init>", "(JJJ)V");
- jobject jCallbackObj = env->NewObject(jCallbackCls, jCallbackCtor, this, cbf, user);
- return jCallbackObj;
-}
-
-jobject JAudioTrack::createCallbackExecutor() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- jclass jExecutorsCls = env->FindClass("java/util/concurrent/Executors");
- jmethodID jNewSingleThreadExecutor = env->GetStaticMethodID(jExecutorsCls,
- "newSingleThreadExecutor", "()Ljava/util/concurrent/ExecutorService;");
- jobject jSingleThreadExecutorObj =
- env->CallStaticObjectMethod(jExecutorsCls, jNewSingleThreadExecutor);
- return jSingleThreadExecutorObj;
-}
-
-status_t JAudioTrack::javaToNativeStatus(int javaStatus) {
- switch (javaStatus) {
- case AUDIO_JAVA_SUCCESS:
- return NO_ERROR;
- case AUDIO_JAVA_BAD_VALUE:
- return BAD_VALUE;
- case AUDIO_JAVA_INVALID_OPERATION:
- return INVALID_OPERATION;
- case AUDIO_JAVA_PERMISSION_DENIED:
- return PERMISSION_DENIED;
- case AUDIO_JAVA_NO_INIT:
- return NO_INIT;
- case AUDIO_JAVA_WOULD_BLOCK:
- return WOULD_BLOCK;
- case AUDIO_JAVA_DEAD_OBJECT:
- return DEAD_OBJECT;
- default:
- return UNKNOWN_ERROR;
- }
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/JMedia2HTTPConnection.cpp b/media/libmediaplayer2/JMedia2HTTPConnection.cpp
deleted file mode 100644
index e1baa10..0000000
--- a/media/libmediaplayer2/JMedia2HTTPConnection.cpp
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- * Copyright 2017, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "JMedia2HTTPConnection"
-#include <utils/Log.h>
-
-#include <mediaplayer2/JavaVMHelper.h>
-#include <mediaplayer2/JMedia2HTTPConnection.h>
-#include <media/stagefright/foundation/ADebug.h>
-#include <nativehelper/scoped_local_ref.h>
-
-#include "log/log.h"
-#include "jni.h"
-
-namespace android {
-
-static const size_t kBufferSize = 32768;
-
-JMedia2HTTPConnection::JMedia2HTTPConnection(JNIEnv *env, jobject thiz) {
- mMedia2HTTPConnectionObj = env->NewGlobalRef(thiz);
- CHECK(mMedia2HTTPConnectionObj != NULL);
-
- ScopedLocalRef<jclass> media2HTTPConnectionClass(
- env, env->GetObjectClass(mMedia2HTTPConnectionObj));
- CHECK(media2HTTPConnectionClass.get() != NULL);
-
- mConnectMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "connect",
- "(Ljava/lang/String;Ljava/lang/String;)Z");
- CHECK(mConnectMethod != NULL);
-
- mDisconnectMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "disconnect",
- "()V");
- CHECK(mDisconnectMethod != NULL);
-
- mReadAtMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "readAt",
- "(J[BI)I");
- CHECK(mReadAtMethod != NULL);
-
- mGetSizeMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "getSize",
- "()J");
- CHECK(mGetSizeMethod != NULL);
-
- mGetMIMETypeMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "getMIMEType",
- "()Ljava/lang/String;");
- CHECK(mGetMIMETypeMethod != NULL);
-
- mGetUriMethod = env->GetMethodID(
- media2HTTPConnectionClass.get(),
- "getUri",
- "()Ljava/lang/String;");
- CHECK(mGetUriMethod != NULL);
-
- ScopedLocalRef<jbyteArray> tmp(
- env, env->NewByteArray(kBufferSize));
- mByteArrayObj = (jbyteArray)env->NewGlobalRef(tmp.get());
- CHECK(mByteArrayObj != NULL);
-}
-
-JMedia2HTTPConnection::~JMedia2HTTPConnection() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- env->DeleteGlobalRef(mMedia2HTTPConnectionObj);
- env->DeleteGlobalRef(mByteArrayObj);
-}
-
-bool JMedia2HTTPConnection::connect(
- const char *uri, const KeyedVector<String8, String8> *headers) {
- String8 tmp("");
- if (headers != NULL) {
- for (size_t i = 0; i < headers->size(); ++i) {
- tmp.append(headers->keyAt(i));
- tmp.append(String8(": "));
- tmp.append(headers->valueAt(i));
- tmp.append(String8("\r\n"));
- }
- }
-
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- jstring juri = env->NewStringUTF(uri);
- jstring jheaders = env->NewStringUTF(tmp.string());
-
- jboolean ret =
- env->CallBooleanMethod(mMedia2HTTPConnectionObj, mConnectMethod, juri, jheaders);
-
- env->DeleteLocalRef(juri);
- env->DeleteLocalRef(jheaders);
-
- return (bool)ret;
-}
-
-void JMedia2HTTPConnection::disconnect() {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- env->CallVoidMethod(mMedia2HTTPConnectionObj, mDisconnectMethod);
-}
-
-ssize_t JMedia2HTTPConnection::readAt(off64_t offset, void *data, size_t size) {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
-
- if (size > kBufferSize) {
- size = kBufferSize;
- }
-
- jint n = env->CallIntMethod(
- mMedia2HTTPConnectionObj, mReadAtMethod, (jlong)offset, mByteArrayObj, (jint)size);
-
- if (n > 0) {
- env->GetByteArrayRegion(
- mByteArrayObj,
- 0,
- n,
- (jbyte *)data);
- }
-
- return n;
-}
-
-off64_t JMedia2HTTPConnection::getSize() {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- return (off64_t)(env->CallLongMethod(mMedia2HTTPConnectionObj, mGetSizeMethod));
-}
-
-status_t JMedia2HTTPConnection::getMIMEType(String8 *mimeType) {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- jstring jmime = (jstring)env->CallObjectMethod(mMedia2HTTPConnectionObj, mGetMIMETypeMethod);
- jboolean flag = env->ExceptionCheck();
- if (flag) {
- env->ExceptionClear();
- return UNKNOWN_ERROR;
- }
-
- const char *str = env->GetStringUTFChars(jmime, 0);
- if (str != NULL) {
- *mimeType = String8(str);
- } else {
- *mimeType = "application/octet-stream";
- }
- env->ReleaseStringUTFChars(jmime, str);
- return OK;
-}
-
-status_t JMedia2HTTPConnection::getUri(String8 *uri) {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- jstring juri = (jstring)env->CallObjectMethod(mMedia2HTTPConnectionObj, mGetUriMethod);
- jboolean flag = env->ExceptionCheck();
- if (flag) {
- env->ExceptionClear();
- return UNKNOWN_ERROR;
- }
-
- const char *str = env->GetStringUTFChars(juri, 0);
- *uri = String8(str);
- env->ReleaseStringUTFChars(juri, str);
- return OK;
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/JMedia2HTTPService.cpp b/media/libmediaplayer2/JMedia2HTTPService.cpp
deleted file mode 100644
index 20e3573..0000000
--- a/media/libmediaplayer2/JMedia2HTTPService.cpp
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright 2017, The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "JMedia2HTTPService"
-#include <utils/Log.h>
-
-#include <jni.h>
-
-#include <mediaplayer2/JavaVMHelper.h>
-#include <mediaplayer2/JMedia2HTTPService.h>
-#include <mediaplayer2/JMedia2HTTPConnection.h>
-#include <media/stagefright/foundation/ADebug.h>
-
-#include <nativehelper/scoped_local_ref.h>
-
-namespace android {
-
-JMedia2HTTPService::JMedia2HTTPService(JNIEnv *env, jobject thiz) {
- mMedia2HTTPServiceObj = env->NewGlobalRef(thiz);
- CHECK(mMedia2HTTPServiceObj != NULL);
-
- ScopedLocalRef<jclass> media2HTTPServiceClass(env, env->GetObjectClass(mMedia2HTTPServiceObj));
- CHECK(media2HTTPServiceClass.get() != NULL);
-
- mMakeHTTPConnectionMethod = env->GetMethodID(
- media2HTTPServiceClass.get(),
- "makeHTTPConnection",
- "()Landroid/media/Media2HTTPConnection;");
- CHECK(mMakeHTTPConnectionMethod != NULL);
-}
-
-JMedia2HTTPService::~JMedia2HTTPService() {
- JNIEnv *env = JavaVMHelper::getJNIEnv();
- env->DeleteGlobalRef(mMedia2HTTPServiceObj);
-}
-
-sp<MediaHTTPConnection> JMedia2HTTPService::makeHTTPConnection() {
- JNIEnv* env = JavaVMHelper::getJNIEnv();
- jobject media2HTTPConnectionObj =
- env->CallObjectMethod(mMedia2HTTPServiceObj, mMakeHTTPConnectionMethod);
-
- return new JMedia2HTTPConnection(env, media2HTTPConnectionObj);
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/JavaVMHelper.cpp b/media/libmediaplayer2/JavaVMHelper.cpp
deleted file mode 100644
index 8d03ed0..0000000
--- a/media/libmediaplayer2/JavaVMHelper.cpp
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright 2018 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- */
-
-#define LOG_TAG "JavaVMHelper"
-
-#include "mediaplayer2/JavaVMHelper.h"
-
-#include <media/stagefright/foundation/ADebug.h>
-#include <utils/threads.h>
-
-#include <stdlib.h>
-
-namespace android {
-
-// static
-std::atomic<JavaVM *> JavaVMHelper::sJavaVM(NULL);
-
-/*
- * Makes the current thread visible to the VM.
- *
- * The JNIEnv pointer returned is only valid for the current thread, and
- * thus must be tucked into thread-local storage.
- */
-static int javaAttachThread(const char* threadName, JNIEnv** pEnv) {
- JavaVMAttachArgs args;
- JavaVM* vm;
- jint result;
-
- vm = JavaVMHelper::getJavaVM();
- if (vm == NULL) {
- return JNI_ERR;
- }
-
- args.version = JNI_VERSION_1_4;
- args.name = (char*) threadName;
- args.group = NULL;
-
- result = vm->AttachCurrentThread(pEnv, (void*) &args);
- if (result != JNI_OK) {
- ALOGI("NOTE: attach of thread '%s' failed\n", threadName);
- }
-
- return result;
-}
-
-/*
- * Detach the current thread from the set visible to the VM.
- */
-static int javaDetachThread(void) {
- JavaVM* vm;
- jint result;
-
- vm = JavaVMHelper::getJavaVM();
- if (vm == NULL) {
- return JNI_ERR;
- }
-
- result = vm->DetachCurrentThread();
- if (result != JNI_OK) {
- ALOGE("ERROR: thread detach failed\n");
- }
- return result;
-}
-
-/*
- * When starting a native thread that will be visible from the VM, we
- * bounce through this to get the right attach/detach action.
- * Note that this function calls free(args)
- */
-static int javaThreadShell(void* args) {
- void* start = ((void**)args)[0];
- void* userData = ((void **)args)[1];
- char* name = (char*) ((void **)args)[2]; // we own this storage
- free(args);
- JNIEnv* env;
- int result;
-
- /* hook us into the VM */
- if (javaAttachThread(name, &env) != JNI_OK) {
- return -1;
- }
-
- /* start the thread running */
- result = (*(android_thread_func_t)start)(userData);
-
- /* unhook us */
- javaDetachThread();
- free(name);
-
- return result;
-}
-
-/*
- * This is invoked from androidCreateThreadEtc() via the callback
- * set with androidSetCreateThreadFunc().
- *
- * We need to create the new thread in such a way that it gets hooked
- * into the VM before it really starts executing.
- */
-static int javaCreateThreadEtc(
- android_thread_func_t entryFunction,
- void* userData,
- const char* threadName,
- int32_t threadPriority,
- size_t threadStackSize,
- android_thread_id_t* threadId) {
- void** args = (void**) malloc(3 * sizeof(void*)); // javaThreadShell must free
- int result;
-
- LOG_ALWAYS_FATAL_IF(threadName == nullptr, "threadName not provided to javaCreateThreadEtc");
-
- args[0] = (void*) entryFunction;
- args[1] = userData;
- args[2] = (void*) strdup(threadName); // javaThreadShell must free
-
- result = androidCreateRawThreadEtc(javaThreadShell, args,
- threadName, threadPriority, threadStackSize, threadId);
- return result;
-}
-
-// static
-JNIEnv *JavaVMHelper::getJNIEnv() {
- JNIEnv *env;
- JavaVM *vm = sJavaVM.load();
- CHECK(vm != NULL);
-
- if (vm->GetEnv((void **)&env, JNI_VERSION_1_4) != JNI_OK) {
- return NULL;
- }
-
- return env;
-}
-
-//static
-JavaVM *JavaVMHelper::getJavaVM() {
- return sJavaVM.load();
-}
-
-// static
-void JavaVMHelper::setJavaVM(JavaVM *vm) {
- sJavaVM.store(vm);
-
- // Ensure that Thread(/*canCallJava*/ true) in libutils is attached to the VM.
- // This is supposed to be done by runtime, but when libutils is used with linker
- // namespace, CreateThreadFunc should be initialized separately within the namespace.
- androidSetCreateThreadFunc((android_create_thread_fn) javaCreateThreadEtc);
-}
-
-} // namespace android
diff --git a/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp b/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp
deleted file mode 100644
index b4fa0c1..0000000
--- a/media/libmediaplayer2/MediaPlayer2AudioOutput.cpp
+++ /dev/null
@@ -1,656 +0,0 @@
-/*
-**
-** Copyright 2018, The Android Open Source Project
-**
-** Licensed under the Apache License, Version 2.0 (the "License");
-** you may not use this file except in compliance with the License.
-** You may obtain a copy of the License at
-**
-** http://www.apache.org/licenses/LICENSE-2.0
-**
-** Unless required by applicable law or agreed to in writing, software
-** distributed under the License is distributed on an "AS IS" BASIS,
-** WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-** See the License for the specific language governing permissions and
-** limitations under the License.
-*/
-
-//#define LOG_NDEBUG 0
-#define LOG_TAG "MediaPlayer2AudioOutput"
-#include <mediaplayer2/MediaPlayer2AudioOutput.h>
-
-#include <cutils/properties.h> // for property_get
-#include <utils/Log.h>
-
-#include <media/stagefright/foundation/ADebug.h>
-
-namespace {
-
-const float kMaxRequiredSpeed = 8.0f; // for PCM tracks allow up to 8x speedup.
-
-} // anonymous namespace
-
-namespace android {
-
-// TODO: Find real cause of Audio/Video delay in PV framework and remove this workaround
-/* static */ int MediaPlayer2AudioOutput::mMinBufferCount = 4;
-/* static */ bool MediaPlayer2AudioOutput::mIsOnEmulator = false;
-
-status_t MediaPlayer2AudioOutput::dump(int fd, const Vector<String16>& args) const {
- const size_t SIZE = 256;
- char buffer[SIZE];
- String8 result;
-
- result.append(" MediaPlayer2AudioOutput\n");
- snprintf(buffer, 255, " volume(%f)\n", mVolume);
- result.append(buffer);
- snprintf(buffer, 255, " msec per frame(%f), latency (%d)\n",
- mMsecsPerFrame, (mJAudioTrack != nullptr) ? mJAudioTrack->latency() : -1);
- result.append(buffer);
- snprintf(buffer, 255, " aux effect id(%d), send level (%f)\n",
- mAuxEffectId, mSendLevel);
- result.append(buffer);
-
- ::write(fd, result.string(), result.size());
- if (mJAudioTrack != nullptr) {
- mJAudioTrack->dump(fd, args);
- }
- return NO_ERROR;
-}
-
-MediaPlayer2AudioOutput::MediaPlayer2AudioOutput(int32_t sessionId, uid_t uid, int pid,
- const jobject attributes)
- : mCallback(nullptr),
- mCallbackCookie(nullptr),
- mCallbackData(nullptr),
- mVolume(1.0),
- mPlaybackRate(AUDIO_PLAYBACK_RATE_DEFAULT),
- mSampleRateHz(0),
- mMsecsPerFrame(0),
- mFrameSize(0),
- mSessionId(sessionId),
- mUid(uid),
- mPid(pid),
- mSendLevel(0.0),
- mAuxEffectId(0),
- mFlags(AUDIO_OUTPUT_FLAG_NONE) {
- ALOGV("MediaPlayer2AudioOutput(%d)", sessionId);
-
- if (attributes != nullptr) {
- mAttributes = new JObjectHolder(attributes);
- }
-
- setMinBufferCount();
- mRoutingDelegates.clear();
-}
-
-MediaPlayer2AudioOutput::~MediaPlayer2AudioOutput() {
- close();
- delete mCallbackData;
-}
-
-//static
-void MediaPlayer2AudioOutput::setMinBufferCount() {
- char value[PROPERTY_VALUE_MAX];
- if (property_get("ro.kernel.qemu", value, 0)) {
- mIsOnEmulator = true;
- mMinBufferCount = 12; // to prevent systematic buffer underrun for emulator
- }
-}
-
-// static
-bool MediaPlayer2AudioOutput::isOnEmulator() {
- setMinBufferCo