/external/webrtc/webrtc/modules/audio_device/ |
mock_audio_device_buffer.h | 24 MOCK_METHOD1(GetPlayoutData, int32_t(void* audioBuffer)); 26 int32_t(const void* audioBuffer, size_t nSamples));
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audio_device_buffer.h | 52 virtual int32_t SetRecordedBuffer(const void* audioBuffer, 62 virtual int32_t GetPlayoutData(void* audioBuffer);
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audio_device_buffer.cc | 383 int32_t AudioDeviceBuffer::SetRecordedBuffer(const void* audioBuffer, 405 memcpy(&_recBuffer[0], audioBuffer, _recSize); 409 int16_t* ptr16In = (int16_t*)audioBuffer; 560 int32_t AudioDeviceBuffer::GetPlayoutData(void* audioBuffer) 573 memcpy(audioBuffer, &_playBuffer[0], _playSize);
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/frameworks/av/media/libaudiohal/include/media/audiohal/ |
EffectBufferHalInterface.h | 33 virtual audio_buffer_t* audioBuffer() = 0; 38 return externalData() != nullptr ? externalData() : audioBuffer()->raw;
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/system/bt/btif/include/ |
btif_avrcp_audio_track.h | 67 int BtifAvrcpAudioTrackWriteData(void* handle, void* audioBuffer,
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/frameworks/av/media/libaaudio/src/legacy/ |
AudioStreamLegacy.cpp | 92 // TODO define our own AudioBuffer and pass it from the subclasses. 93 AudioTrack::Buffer *audioBuffer = static_cast<AudioTrack::Buffer *>(info); 96 audioBuffer->size = SIZE_STOP_CALLBACKS; 99 audioBuffer->size = SIZE_STOP_CALLBACKS; 101 if (audioBuffer->frameCount == 0) { 108 int32_t byteCount = audioBuffer->frameCount * getBytesPerDeviceFrame(); 110 (uint8_t *) audioBuffer->raw, byteCount); 113 callbackResult = callDataCallbackFrames((uint8_t *)audioBuffer->raw, 114 audioBuffer->frameCount); 117 audioBuffer->size = audioBuffer->frameCount * getBytesPerDeviceFrame() [all...] |
/frameworks/base/media/java/android/media/ |
MediaSync.java | 53 * public void onAudioBufferConsumed(MediaSync sync, ByteBuffer audioBuffer, int bufferId) { 126 * @param audioBuffer The returned audio buffer. 127 * @param bufferId The ID associated with audioBuffer as passed into 131 @NonNull MediaSync sync, @NonNull ByteBuffer audioBuffer, int bufferId); 172 private static class AudioBuffer { 177 public AudioBuffer(@NonNull ByteBuffer byteBuffer, int bufferId, 201 private List<AudioBuffer> mAudioBuffers = new LinkedList<AudioBuffer>(); 515 mAudioBuffers.add(new AudioBuffer(audioData, bufferId, presentationTimeUs)); 536 AudioBuffer audioBuffer = mAudioBuffers.get(0) [all...] |
/external/webrtc/webrtc/modules/media_file/ |
media_file_utility.h | 38 // Put 10-60ms of audio data from stream into the audioBuffer depending on 39 // codec frame size. dataLengthInBytes indicates the size of audioBuffer. 40 // The return value is the number of bytes written to audioBuffer. 44 int32_t ReadWavDataAsMono(InStream& stream, int8_t* audioBuffer, 66 // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer, 69 // The return value is the number of bytes written to audioBuffer. 71 const int8_t* audioBuffer, 91 // Put 10-60ms of audio data from stream into the audioBuffer depending on 92 // codec frame size. dataLengthInBytes indicates the size of audioBuffer. 93 // The return value is the number of bytes written to audioBuffer [all...] |
media_file.h | 29 // Put 10-60ms of audio data from file into the audioBuffer depending on 31 // parameter. As input parameter it indicates the size of audioBuffer. 33 // audioBuffer. 38 int8_t* audioBuffer, 107 // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer, 112 const int8_t* audioBuffer,
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media_file_impl.h | 33 int32_t PlayoutAudioData(int8_t* audioBuffer, 63 int32_t IncomingAudioData(const int8_t* audioBuffer,
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/frameworks/av/media/libaudioclient/ |
AudioRecord.cpp | 843 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, int32_t waitCount, size_t *nonContig) 845 if (audioBuffer == NULL) { 852 audioBuffer->frameCount = 0; 853 audioBuffer->size = 0; 854 audioBuffer->raw = NULL; 876 return obtainBuffer(audioBuffer, requested, NULL /*elapsed*/, nonContig); 879 status_t AudioRecord::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested, 930 buffer.mFrameCount = audioBuffer->frameCount; 936 audioBuffer->frameCount = buffer.mFrameCount; 937 audioBuffer->size = buffer.mFrameCount * mFrameSize [all...] |
/frameworks/av/media/libaaudio/tests/ |
test_aaudio_monkey.cpp | 178 int16_t *audioBuffer = (int16_t *) audioData; 180 mSine1.render(&audioBuffer[0], samplesPerFrame, numFrames); 183 mSine2.render(&audioBuffer[1], samplesPerFrame, numFrames); 188 float *audioBuffer = (float *) audioData; 190 mSine1.render(&audioBuffer[0], samplesPerFrame, numFrames); 193 mSine2.render(&audioBuffer[1], samplesPerFrame, numFrames);
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/system/bt/btif/src/ |
btif_avrcp_audio_track.cc | 131 int BtifAvrcpAudioTrackWriteData(void* handle, void* audioBuffer, 139 fwrite((audioBuffer), 1, (size_t)bufferlen, outputPcmSampleFile); 142 retval = trackHolder->track->write(audioBuffer, (size_t)bufferlen);
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/system/chre/platform/android/ |
platform_audio.cc | 59 int16_t *audioBuffer = platformAudio->mBuffer.data(); 63 audioBuffer = &platformAudio->mBuffer.data()[seekAmount]; 73 platformAudio->mStream, audioBuffer, readAmount, 1);
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/cts/tests/tests/media/src/android/media/cts/ |
CodecState.java | 342 ByteBuffer audioBuffer = ByteBuffer.allocate(buffer.remaining()); 343 audioBuffer.put(buffer); 344 audioBuffer.clear(); 346 mAudioTrack.write(audioBuffer, info.size, info.presentationTimeUs*1000);
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AudioHelper.java | 351 public int read(ByteBuffer audioBuffer, int sizeInBytes) { 352 int bytes = super.read(audioBuffer, sizeInBytes); 354 // read does not affect position and limit of the audioBuffer. 357 ByteBuffer copy = audioBuffer.duplicate(); 368 public int read(ByteBuffer audioBuffer, int sizeInBytes, int readMode) { 369 int bytes = super.read(audioBuffer, sizeInBytes, readMode); 371 // read does not affect position and limit of the audioBuffer. 374 ByteBuffer copy = audioBuffer.duplicate();
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/external/webrtc/webrtc/modules/audio_device/android/ |
opensles_player.h | 82 void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
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/external/webrtc/webrtc/modules/utility/source/ |
file_recorder_impl.h | 77 int32_t WriteEncodedAudioData(const int8_t* audioBuffer,
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file_recorder_impl.cc | 256 int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer, 259 return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
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/frameworks/av/media/libaaudio/examples/utils/ |
AAudioSimpleRecorder.h | 267 int16_t *audioBuffer = (int16_t *) audioData; 270 sample = audioBuffer[frameIndex * samplesPerFrame] * (1.0/32768); 279 float *audioBuffer = (float *) audioData; 282 sample = audioBuffer[frameIndex * samplesPerFrame];
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/frameworks/av/media/libaudiohal/2.0/ |
EffectBufferHalHidl.h | 26 using android::hardware::audio::effect::V2_0::AudioBuffer; 38 virtual audio_buffer_t* audioBuffer(); 52 const AudioBuffer& hidlBuffer() const { return mHidlBuffer; } 62 AudioBuffer mHidlBuffer;
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/frameworks/av/media/libaudiohal/4.0/ |
EffectBufferHalHidl.h | 26 using android::hardware::audio::effect::V4_0::AudioBuffer; 39 virtual audio_buffer_t* audioBuffer(); 53 const AudioBuffer& hidlBuffer() const { return mHidlBuffer; } 63 AudioBuffer mHidlBuffer;
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/external/aac/libDRCdec/src/ |
drcDec_gainDecoder.cpp | 313 FIXP_DBL* audioBuffer) { 360 drcDec_GainDecoder_SetChannelGains_func1(audioBuffer, gain, stepsize, 364 audioBuffer[i] = fMultDiv2(audioBuffer[i], gain) << n_min; 368 audioBuffer += audioBufferChannelOffset;
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/external/tensorflow/tensorflow/examples/android/src/org/tensorflow/demo/ |
SpeechActivity.java | 212 short[] audioBuffer = new short[bufferSize / 2]; 233 int numberRead = record.read(audioBuffer, 0, audioBuffer.length); 243 System.arraycopy(audioBuffer, 0, recordingBuffer, recordingOffset, firstCopyLength); 244 System.arraycopy(audioBuffer, firstCopyLength, recordingBuffer, 0, secondCopyLength);
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/frameworks/av/media/libstagefright/ |
AudioSource.cpp | 324 status_t AudioSource::dataCallback(const AudioRecord::Buffer& audioBuffer) { 354 const size_t bufferSize = audioBuffer.size; 392 CHECK_EQ(audioBuffer.size & 1, 0u); 414 if (audioBuffer.size == 0) { 421 audioBuffer.i16, audioBuffer.size);
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