| /* |
| * 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 <aaudio/AAudio.h> |
| #include <audio_utils/primitives.h> |
| #include <media/AudioRecord.h> |
| #include <utils/String16.h> |
| |
| #include "legacy/AudioStreamLegacy.h" |
| #include "legacy/AudioStreamRecord.h" |
| #include "utility/AudioClock.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 |
| |
| const aaudio_session_id_t requestedSessionId = builder.getSessionId(); |
| const audio_session_t sessionId = AAudioConvert_aaudioToAndroidSessionId(requestedSessionId); |
| |
| // TODO Support UNSPECIFIED in AudioRecord. For now, use stereo if unspecified. |
| int32_t samplesPerFrame = (getSamplesPerFrame() == AAUDIO_UNSPECIFIED) |
| ? 2 : getSamplesPerFrame(); |
| audio_channel_mask_t channelMask = samplesPerFrame <= 2 ? |
| audio_channel_in_mask_from_count(samplesPerFrame) : |
| audio_channel_mask_for_index_assignment_from_count(samplesPerFrame); |
| |
| size_t frameCount = (builder.getBufferCapacity() == AAUDIO_UNSPECIFIED) ? 0 |
| : builder.getBufferCapacity(); |
| |
| |
| audio_input_flags_t flags; |
| aaudio_performance_mode_t perfMode = getPerformanceMode(); |
| switch (perfMode) { |
| case AAUDIO_PERFORMANCE_MODE_LOW_LATENCY: |
| // If the app asks for a sessionId then it means they want to use effects. |
| // So don't use RAW flag. |
| flags = (audio_input_flags_t) ((requestedSessionId == AAUDIO_SESSION_ID_NONE) |
| ? (AUDIO_INPUT_FLAG_FAST | AUDIO_INPUT_FLAG_RAW) |
| : (AUDIO_INPUT_FLAG_FAST)); |
| break; |
| |
| case AAUDIO_PERFORMANCE_MODE_POWER_SAVING: |
| case AAUDIO_PERFORMANCE_MODE_NONE: |
| default: |
| flags = AUDIO_INPUT_FLAG_NONE; |
| break; |
| } |
| |
| // Preserve behavior of API 26 |
| if (getFormat() == AUDIO_FORMAT_DEFAULT) { |
| setFormat(AUDIO_FORMAT_PCM_FLOAT); |
| } |
| |
| // Maybe change device format to get a FAST path. |
| // AudioRecord does not support FAST mode for FLOAT data. |
| // TODO AudioRecord should allow FLOAT data paths for FAST tracks. |
| // So IF the user asks for low latency FLOAT |
| // AND the sampleRate is likely to be compatible with FAST |
| // THEN request I16 and convert to FLOAT when passing to user. |
| // Note that hard coding 48000 Hz is not ideal because the sampleRate |
| // for a FAST path might not be 48000 Hz. |
| // It normally is but there is a chance that it is not. |
| // And there is no reliable way to know that in advance. |
| // Luckily the consequences of a wrong guess are minor. |
| // We just may not get a FAST track. |
| // But we wouldn't have anyway without this hack. |
| constexpr int32_t kMostLikelySampleRateForFast = 48000; |
| if (getFormat() == AUDIO_FORMAT_PCM_FLOAT |
| && perfMode == AAUDIO_PERFORMANCE_MODE_LOW_LATENCY |
| && (samplesPerFrame <= 2) // FAST only for mono and stereo |
| && (getSampleRate() == kMostLikelySampleRateForFast |
| || getSampleRate() == AAUDIO_UNSPECIFIED)) { |
| setDeviceFormat(AUDIO_FORMAT_PCM_16_BIT); |
| } else { |
| setDeviceFormat(getFormat()); |
| } |
| |
| 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(); |
| |
| // Don't call mAudioRecord->setInputDevice() because it will be overwritten by set()! |
| audio_port_handle_t selectedDeviceId = (getDeviceId() == AAUDIO_UNSPECIFIED) |
| ? AUDIO_PORT_HANDLE_NONE |
| : getDeviceId(); |
| |
| const audio_content_type_t contentType = |
| AAudioConvert_contentTypeToInternal(builder.getContentType()); |
| const audio_source_t source = |
| AAudioConvert_inputPresetToAudioSource(builder.getInputPreset()); |
| |
| const audio_attributes_t attributes = { |
| .content_type = contentType, |
| .usage = AUDIO_USAGE_UNKNOWN, // only used for output |
| .source = source, |
| .flags = AUDIO_FLAG_NONE, // Different than the AUDIO_INPUT_FLAGS |
| .tags = "" |
| }; |
| |
| // ----------- open the AudioRecord --------------------- |
| // Might retry, but never more than once. |
| for (int i = 0; i < 2; i ++) { |
| const audio_format_t requestedInternalFormat = getDeviceFormat(); |
| |
| mAudioRecord = new AudioRecord( |
| mOpPackageName // const String16& opPackageName TODO does not compile |
| ); |
| mAudioRecord->set( |
| AUDIO_SOURCE_DEFAULT, // ignored because we pass attributes below |
| getSampleRate(), |
| requestedInternalFormat, |
| channelMask, |
| frameCount, |
| callback, |
| callbackData, |
| notificationFrames, |
| false /*threadCanCallJava*/, |
| sessionId, |
| streamTransferType, |
| flags, |
| AUDIO_UID_INVALID, // DEFAULT uid |
| -1, // DEFAULT pid |
| &attributes, |
| selectedDeviceId |
| ); |
| |
| // Did we get a valid track? |
| status_t status = mAudioRecord->initCheck(); |
| if (status != OK) { |
| close(); |
| ALOGE("open(), initCheck() returned %d", status); |
| return AAudioConvert_androidToAAudioResult(status); |
| } |
| |
| // Check to see if it was worth hacking the deviceFormat. |
| bool gotFastPath = (mAudioRecord->getFlags() & AUDIO_INPUT_FLAG_FAST) |
| == AUDIO_INPUT_FLAG_FAST; |
| if (getFormat() != getDeviceFormat() && !gotFastPath) { |
| // We tried to get a FAST path by switching the device format. |
| // But it didn't work. So we might as well reopen using the same |
| // format for device and for app. |
| ALOGD("%s() used a different device format but no FAST path, reopen", __func__); |
| mAudioRecord.clear(); |
| setDeviceFormat(getFormat()); |
| } else { |
| break; // Keep the one we just opened. |
| } |
| } |
| |
| // Get the actual values from the AudioRecord. |
| setSamplesPerFrame(mAudioRecord->channelCount()); |
| |
| int32_t actualSampleRate = mAudioRecord->getSampleRate(); |
| ALOGW_IF(actualSampleRate != getSampleRate(), |
| "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; |
| } |
| |
| // Allocate format conversion buffer if needed. |
| if (getDeviceFormat() == AUDIO_FORMAT_PCM_16_BIT |
| && getFormat() == AUDIO_FORMAT_PCM_FLOAT) { |
| |
| if (builder.getDataCallbackProc() != nullptr) { |
| // If we have a callback then we need to convert the data into an internal float |
| // array and then pass that entire array to the app. |
| mFormatConversionBufferSizeInFrames = |
| (mCallbackBufferSize != AAUDIO_UNSPECIFIED) |
| ? mCallbackBufferSize : getFramesPerBurst(); |
| int32_t numSamples = mFormatConversionBufferSizeInFrames * getSamplesPerFrame(); |
| mFormatConversionBufferFloat = std::make_unique<float[]>(numSamples); |
| } else { |
| // If we don't have a callback then we will read into an internal short array |
| // and then convert into the app float array in read(). |
| mFormatConversionBufferSizeInFrames = getFramesPerBurst(); |
| int32_t numSamples = mFormatConversionBufferSizeInFrames * getSamplesPerFrame(); |
| mFormatConversionBufferI16 = std::make_unique<int16_t[]>(numSamples); |
| } |
| ALOGD("%s() setup I16>FLOAT conversion buffer with %d frames", |
| __func__, mFormatConversionBufferSizeInFrames); |
| } |
| |
| // 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, |
| "open() flags changed from 0x%08X to 0x%08X", |
| flags, actualFlags); |
| ALOGW_IF(actualPerformanceMode != perfMode, |
| "open() perfMode changed from %d to %d", |
| perfMode, actualPerformanceMode); |
| |
| setState(AAUDIO_STREAM_STATE_OPEN); |
| setDeviceId(mAudioRecord->getRoutedDeviceId()); |
| |
| aaudio_session_id_t actualSessionId = |
| (requestedSessionId == AAUDIO_SESSION_ID_NONE) |
| ? AAUDIO_SESSION_ID_NONE |
| : (aaudio_session_id_t) mAudioRecord->getSessionId(); |
| setSessionId(actualSessionId); |
| |
| 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(); |
| } |
| |
| const void * AudioStreamRecord::maybeConvertDeviceData(const void *audioData, int32_t numFrames) { |
| if (mFormatConversionBufferFloat.get() != nullptr) { |
| LOG_ALWAYS_FATAL_IF(numFrames > mFormatConversionBufferSizeInFrames, |
| "%s() conversion size %d too large for buffer %d", |
| __func__, numFrames, mFormatConversionBufferSizeInFrames); |
| |
| int32_t numSamples = numFrames * getSamplesPerFrame(); |
| // Only conversion supported is I16 to FLOAT |
| memcpy_to_float_from_i16( |
| mFormatConversionBufferFloat.get(), |
| (const int16_t *) audioData, |
| numSamples); |
| return mFormatConversionBufferFloat.get(); |
| } else { |
| return audioData; |
| } |
| } |
| |
| 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; |
| } |
| |
| // Enable callback before starting AudioRecord to avoid shutting |
| // down because of a race condition. |
| mCallbackEnabled.store(true); |
| mFramesWritten.reset32(); // service writes frames |
| mTimestampPosition.reset32(); |
| status_t err = mAudioRecord->start(); // resets position to zero |
| if (err != OK) { |
| return AAudioConvert_androidToAAudioResult(err); |
| } else { |
| setState(AAUDIO_STREAM_STATE_STARTING); |
| } |
| return AAUDIO_OK; |
| } |
| |
| aaudio_result_t AudioStreamRecord::requestStop() { |
| if (mAudioRecord.get() == nullptr) { |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| setState(AAUDIO_STREAM_STATE_STOPPING); |
| mFramesWritten.catchUpTo(getFramesRead()); |
| mTimestampPosition.catchUpTo(getFramesRead()); |
| mAudioRecord->stop(); |
| mCallbackEnabled.store(false); |
| // Pass false to prevent errorCallback from being called after disconnect |
| // when app has already requested a stop(). |
| return checkForDisconnectRequest(false); |
| } |
| |
| 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: |
| // When starting, the position will begin at zero and then go positive. |
| // The position can wrap but by that time the state will not be STARTING. |
| err = mAudioRecord->getPosition(&position); |
| if (err != OK) { |
| result = AAudioConvert_androidToAAudioResult(err); |
| } else if (position > 0) { |
| 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 bytesPerDeviceFrame = getBytesPerDeviceFrame(); |
| int32_t numBytes; |
| // This will detect out of range values for numFrames. |
| aaudio_result_t result = AAudioConvert_framesToBytes(numFrames, bytesPerDeviceFrame, &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 bytesActuallyRead = 0; |
| ssize_t totalBytesRead = 0; |
| if (mFormatConversionBufferI16.get() != nullptr) { |
| // Convert I16 data to float using an intermediate buffer. |
| float *floatBuffer = (float *) buffer; |
| int32_t framesLeft = numFrames; |
| // Perform conversion using multiple read()s if necessary. |
| while (framesLeft > 0) { |
| // Read into short internal buffer. |
| int32_t framesToRead = std::min(framesLeft, mFormatConversionBufferSizeInFrames); |
| size_t bytesToRead = framesToRead * bytesPerDeviceFrame; |
| bytesActuallyRead = mAudioRecord->read(mFormatConversionBufferI16.get(), bytesToRead, blocking); |
| if (bytesActuallyRead <= 0) { |
| break; |
| } |
| totalBytesRead += bytesActuallyRead; |
| int32_t framesToConvert = bytesActuallyRead / bytesPerDeviceFrame; |
| // Convert into app float buffer. |
| size_t numSamples = framesToConvert * getSamplesPerFrame(); |
| memcpy_to_float_from_i16( |
| floatBuffer, |
| mFormatConversionBufferI16.get(), |
| numSamples); |
| floatBuffer += numSamples; |
| framesLeft -= framesToConvert; |
| } |
| } else { |
| bytesActuallyRead = mAudioRecord->read(buffer, numBytes, blocking); |
| totalBytesRead = bytesActuallyRead; |
| } |
| if (bytesActuallyRead == WOULD_BLOCK) { |
| return 0; |
| } else if (bytesActuallyRead < 0) { |
| // In this context, a DEAD_OBJECT is more likely to be a disconnect notification due to |
| // AudioRecord invalidation. |
| if (bytesActuallyRead == DEAD_OBJECT) { |
| setState(AAUDIO_STREAM_STATE_DISCONNECTED); |
| return AAUDIO_ERROR_DISCONNECTED; |
| } |
| return AAudioConvert_androidToAAudioResult(bytesActuallyRead); |
| } |
| int32_t framesRead = (int32_t)(totalBytesRead / bytesPerDeviceFrame); |
| 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; |
| if (getState() != AAUDIO_STREAM_STATE_STARTED) { |
| return AAUDIO_ERROR_INVALID_STATE; |
| } |
| 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: |
| result = mAudioRecord->getPosition(&position); |
| if (result == OK) { |
| mFramesWritten.update32(position); |
| } |
| break; |
| case AAUDIO_STREAM_STATE_STOPPING: |
| default: |
| break; |
| } |
| return AudioStreamLegacy::getFramesWritten(); |
| } |