| /* |
| * Copyright 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 "AudioStreamRecord" |
| //#define LOG_NDEBUG 0 |
| #include <utils/Log.h> |
| |
| #include <stdint.h> |
| #include <utils/String16.h> |
| #include <media/AudioRecord.h> |
| #include <aaudio/AAudio.h> |
| |
| #include "AudioClock.h" |
| #include "legacy/AudioStreamLegacy.h" |
| #include "legacy/AudioStreamRecord.h" |
| #include "utility/FixedBlockWriter.h" |
| |
| using namespace android; |
| using namespace aaudio; |
| |
| AudioStreamRecord::AudioStreamRecord() |
| : AudioStreamLegacy() |
| , mFixedBlockWriter(*this) |
| { |
| } |
| |
| AudioStreamRecord::~AudioStreamRecord() |
| { |
| const aaudio_stream_state_t state = getState(); |
| bool bad = !(state == AAUDIO_STREAM_STATE_UNINITIALIZED || state == AAUDIO_STREAM_STATE_CLOSED); |
| ALOGE_IF(bad, "stream not closed, in state %d", state); |
| } |
| |
| aaudio_result_t AudioStreamRecord::open(const AudioStreamBuilder& builder) |
| { |
| aaudio_result_t result = AAUDIO_OK; |
| |
| result = AudioStream::open(builder); |
| if (result != AAUDIO_OK) { |
| return result; |
| } |
| |
| // Try to create an AudioRecord |
| |
| // TODO Support UNSPECIFIED in AudioRecord. For now, use stereo if unspecified. |
| int32_t samplesPerFrame = (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) |
| ? 2 : getSamplesPerFrame(); |
| audio_channel_mask_t channelMask = audio_channel_in_mask_from_count(samplesPerFrame); |
| |
| size_t frameCount = (builder.getBufferCapacity() == AAUDIO_UNSPECIFIED) ? 0 |
| : builder.getBufferCapacity(); |
| |
| // TODO implement an unspecified Android format then use that. |
| audio_format_t format = (getFormat() == AAUDIO_FORMAT_UNSPECIFIED) |
| ? AUDIO_FORMAT_PCM_FLOAT |
| : AAudioConvert_aaudioToAndroidDataFormat(getFormat()); |
| |
| audio_input_flags_t flags = AUDIO_INPUT_FLAG_NONE; |
| aaudio_performance_mode_t perfMode = getPerformanceMode(); |
| switch (perfMode) { |
| case AAUDIO_PERFORMANCE_MODE_LOW_LATENCY: |
| flags = (audio_input_flags_t) (AUDIO_INPUT_FLAG_FAST | AUDIO_INPUT_FLAG_RAW); |
| break; |
| |
| case AAUDIO_PERFORMANCE_MODE_POWER_SAVING: |
| case AAUDIO_PERFORMANCE_MODE_NONE: |
| default: |
| // No flags. |
| break; |
| } |
| |
| uint32_t notificationFrames = 0; |
| |
| // Setup the callback if there is one. |
| AudioRecord::callback_t callback = nullptr; |
| void *callbackData = nullptr; |
| AudioRecord::transfer_type streamTransferType = AudioRecord::transfer_type::TRANSFER_SYNC; |
| if (builder.getDataCallbackProc() != nullptr) { |
| streamTransferType = AudioRecord::transfer_type::TRANSFER_CALLBACK; |
| callback = getLegacyCallback(); |
| callbackData = this; |
| notificationFrames = builder.getFramesPerDataCallback(); |
| } |
| mCallbackBufferSize = builder.getFramesPerDataCallback(); |
| |
| ALOGD("AudioStreamRecord::open(), request notificationFrames = %u, frameCount = %u", |
| notificationFrames, (uint)frameCount); |
| mAudioRecord = new AudioRecord( |
| mOpPackageName // const String16& opPackageName TODO does not compile |
| ); |
| if (getDeviceId() != AAUDIO_UNSPECIFIED) { |
| mAudioRecord->setInputDevice(getDeviceId()); |
| } |
| mAudioRecord->set( |
| AUDIO_SOURCE_VOICE_RECOGNITION, |
| getSampleRate(), |
| format, |
| channelMask, |
| frameCount, |
| callback, |
| callbackData, |
| notificationFrames, |
| false /*threadCanCallJava*/, |
| AUDIO_SESSION_ALLOCATE, |
| streamTransferType, |
| flags |
| // int uid = -1, |
| // pid_t pid = -1, |
| // const audio_attributes_t* pAttributes = nullptr |
| ); |
| |
| // Did we get a valid track? |
| status_t status = mAudioRecord->initCheck(); |
| if (status != OK) { |
| close(); |
| ALOGE("AudioStreamRecord::open(), initCheck() returned %d", status); |
| return AAudioConvert_androidToAAudioResult(status); |
| } |
| |
| // Get the actual values from the AudioRecord. |
| setSamplesPerFrame(mAudioRecord->channelCount()); |
| setFormat(AAudioConvert_androidToAAudioDataFormat(mAudioRecord->format())); |
| |
| int32_t actualSampleRate = mAudioRecord->getSampleRate(); |
| ALOGW_IF(actualSampleRate != getSampleRate(), |
| "AudioStreamRecord::open() sampleRate changed from %d to %d", |
| getSampleRate(), actualSampleRate); |
| setSampleRate(actualSampleRate); |
| |
| // We may need to pass the data through a block size adapter to guarantee constant size. |
| if (mCallbackBufferSize != AAUDIO_UNSPECIFIED) { |
| int callbackSizeBytes = getBytesPerFrame() * mCallbackBufferSize; |
| mFixedBlockWriter.open(callbackSizeBytes); |
| mBlockAdapter = &mFixedBlockWriter; |
| } else { |
| mBlockAdapter = nullptr; |
| } |
| |
| // Update performance mode based on the actual stream. |
| // For example, if the sample rate does not match native then you won't get a FAST track. |
| audio_input_flags_t actualFlags = mAudioRecord->getFlags(); |
| aaudio_performance_mode_t actualPerformanceMode = AAUDIO_PERFORMANCE_MODE_NONE; |
| // FIXME Some platforms do not advertise RAW mode for low latency inputs. |
| if ((actualFlags & (AUDIO_INPUT_FLAG_FAST)) |
| == (AUDIO_INPUT_FLAG_FAST)) { |
| actualPerformanceMode = AAUDIO_PERFORMANCE_MODE_LOW_LATENCY; |
| } |
| setPerformanceMode(actualPerformanceMode); |
| |
| setSharingMode(AAUDIO_SHARING_MODE_SHARED); // EXCLUSIVE mode not supported in legacy |
| |
| // Log warning if we did not get what we asked for. |
| ALOGW_IF(actualFlags != flags, |
| "AudioStreamRecord::open() flags changed from 0x%08X to 0x%08X", |
| flags, actualFlags); |
| ALOGW_IF(actualPerformanceMode != perfMode, |
| "AudioStreamRecord::open() perfMode changed from %d to %d", |
| perfMode, actualPerformanceMode); |
| |
| setState(AAUDIO_STREAM_STATE_OPEN); |
| setDeviceId(mAudioRecord->getRoutedDeviceId()); |
| mAudioRecord->addAudioDeviceCallback(mDeviceCallback); |
| |
| return AAUDIO_OK; |
| } |
| |
| aaudio_result_t AudioStreamRecord::close() |
| { |
| // TODO add close() or release() to AudioRecord API then call it from here |
| if (getState() != AAUDIO_STREAM_STATE_CLOSED) { |
| mAudioRecord->removeAudioDeviceCallback(mDeviceCallback); |
| mAudioRecord.clear(); |
| setState(AAUDIO_STREAM_STATE_CLOSED); |
| } |
| mFixedBlockWriter.close(); |
| return AudioStream::close(); |
| } |
| |
| void AudioStreamRecord::processCallback(int event, void *info) { |
| switch (event) { |
| case AudioRecord::EVENT_MORE_DATA: |
| processCallbackCommon(AAUDIO_CALLBACK_OPERATION_PROCESS_DATA, info); |
| break; |
| |
| // Stream got rerouted so we disconnect. |
| case AudioRecord::EVENT_NEW_IAUDIORECORD: |
| processCallbackCommon(AAUDIO_CALLBACK_OPERATION_DISCONNECTED, info); |
| break; |
| |
| default: |
| break; |
| } |
| return; |
| } |
| |
| aaudio_result_t AudioStreamRecord::requestStart() |
| { |
| if (mAudioRecord.get() == nullptr) { |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| // Get current position so we can detect when the track is recording. |
| status_t err = mAudioRecord->getPosition(&mPositionWhenStarting); |
| if (err != OK) { |
| return AAudioConvert_androidToAAudioResult(err); |
| } |
| |
| err = mAudioRecord->start(); |
| if (err != OK) { |
| return AAudioConvert_androidToAAudioResult(err); |
| } else { |
| onStart(); |
| setState(AAUDIO_STREAM_STATE_STARTING); |
| } |
| return AAUDIO_OK; |
| } |
| |
| aaudio_result_t AudioStreamRecord::requestStop() { |
| if (mAudioRecord.get() == nullptr) { |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| onStop(); |
| setState(AAUDIO_STREAM_STATE_STOPPING); |
| incrementFramesWritten(getFramesRead() - getFramesWritten()); // TODO review |
| mTimestampPosition.set(getFramesRead()); |
| mAudioRecord->stop(); |
| mFramesRead.reset32(); |
| mTimestampPosition.reset32(); |
| checkForDisconnectRequest(); |
| return AAUDIO_OK; |
| } |
| |
| aaudio_result_t AudioStreamRecord::updateStateMachine() |
| { |
| aaudio_result_t result = AAUDIO_OK; |
| aaudio_wrapping_frames_t position; |
| status_t err; |
| switch (getState()) { |
| // TODO add better state visibility to AudioRecord |
| case AAUDIO_STREAM_STATE_STARTING: |
| err = mAudioRecord->getPosition(&position); |
| if (err != OK) { |
| result = AAudioConvert_androidToAAudioResult(err); |
| } else if (position != mPositionWhenStarting) { |
| setState(AAUDIO_STREAM_STATE_STARTED); |
| } |
| break; |
| case AAUDIO_STREAM_STATE_STOPPING: |
| if (mAudioRecord->stopped()) { |
| setState(AAUDIO_STREAM_STATE_STOPPED); |
| } |
| break; |
| default: |
| break; |
| } |
| return result; |
| } |
| |
| aaudio_result_t AudioStreamRecord::read(void *buffer, |
| int32_t numFrames, |
| int64_t timeoutNanoseconds) |
| { |
| int32_t bytesPerFrame = getBytesPerFrame(); |
| int32_t numBytes; |
| aaudio_result_t result = AAudioConvert_framesToBytes(numFrames, bytesPerFrame, &numBytes); |
| if (result != AAUDIO_OK) { |
| return result; |
| } |
| |
| if (getState() == AAUDIO_STREAM_STATE_DISCONNECTED) { |
| return AAUDIO_ERROR_DISCONNECTED; |
| } |
| |
| // TODO add timeout to AudioRecord |
| bool blocking = (timeoutNanoseconds > 0); |
| ssize_t bytesRead = mAudioRecord->read(buffer, numBytes, blocking); |
| if (bytesRead == WOULD_BLOCK) { |
| return 0; |
| } else if (bytesRead < 0) { |
| // in this context, a DEAD_OBJECT is more likely to be a disconnect notification due to |
| // AudioRecord invalidation |
| if (bytesRead == DEAD_OBJECT) { |
| setState(AAUDIO_STREAM_STATE_DISCONNECTED); |
| return AAUDIO_ERROR_DISCONNECTED; |
| } |
| return AAudioConvert_androidToAAudioResult(bytesRead); |
| } |
| int32_t framesRead = (int32_t)(bytesRead / bytesPerFrame); |
| incrementFramesRead(framesRead); |
| |
| result = updateStateMachine(); |
| if (result != AAUDIO_OK) { |
| return result; |
| } |
| |
| return (aaudio_result_t) framesRead; |
| } |
| |
| aaudio_result_t AudioStreamRecord::setBufferSize(int32_t requestedFrames) |
| { |
| return getBufferSize(); |
| } |
| |
| int32_t AudioStreamRecord::getBufferSize() const |
| { |
| return getBufferCapacity(); // TODO implement in AudioRecord? |
| } |
| |
| int32_t AudioStreamRecord::getBufferCapacity() const |
| { |
| return static_cast<int32_t>(mAudioRecord->frameCount()); |
| } |
| |
| int32_t AudioStreamRecord::getXRunCount() const |
| { |
| return 0; // TODO implement when AudioRecord supports it |
| } |
| |
| int32_t AudioStreamRecord::getFramesPerBurst() const |
| { |
| return static_cast<int32_t>(mAudioRecord->getNotificationPeriodInFrames()); |
| } |
| |
| aaudio_result_t AudioStreamRecord::getTimestamp(clockid_t clockId, |
| int64_t *framePosition, |
| int64_t *timeNanoseconds) { |
| ExtendedTimestamp extendedTimestamp; |
| status_t status = mAudioRecord->getTimestamp(&extendedTimestamp); |
| if (status == WOULD_BLOCK) { |
| return AAUDIO_ERROR_INVALID_STATE; |
| } else if (status != NO_ERROR) { |
| return AAudioConvert_androidToAAudioResult(status); |
| } |
| return getBestTimestamp(clockId, framePosition, timeNanoseconds, &extendedTimestamp); |
| } |
| |
| int64_t AudioStreamRecord::getFramesWritten() { |
| aaudio_wrapping_frames_t position; |
| status_t result; |
| switch (getState()) { |
| case AAUDIO_STREAM_STATE_STARTING: |
| case AAUDIO_STREAM_STATE_STARTED: |
| case AAUDIO_STREAM_STATE_STOPPING: |
| result = mAudioRecord->getPosition(&position); |
| if (result == OK) { |
| mFramesWritten.update32(position); |
| } |
| break; |
| default: |
| break; |
| } |
| return AudioStreamLegacy::getFramesWritten(); |
| } |