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  /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));
audio_device_buffer.h 52 virtual int32_t SetRecordedBuffer(const void* audioBuffer,
62 virtual int32_t GetPlayoutData(void* audioBuffer);
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);
  /frameworks/av/media/libaudiohal/include/media/audiohal/
EffectBufferHalInterface.h 33 virtual audio_buffer_t* audioBuffer() = 0;
38 return externalData() != nullptr ? externalData() : audioBuffer()->raw;
  /system/bt/btif/include/
btif_avrcp_audio_track.h 67 int BtifAvrcpAudioTrackWriteData(void* handle, void* audioBuffer,
  /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,
media_file_impl.h 33 int32_t PlayoutAudioData(int8_t* audioBuffer,
63 int32_t IncomingAudioData(const int8_t* audioBuffer,
  /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);
  /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);
  /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);
  /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);
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();
  /external/webrtc/webrtc/modules/audio_device/android/
opensles_player.h 82 void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
  /external/webrtc/webrtc/modules/utility/source/
file_recorder_impl.h 77 int32_t WriteEncodedAudioData(const int8_t* audioBuffer,
file_recorder_impl.cc 256 int32_t FileRecorderImpl::WriteEncodedAudioData(const int8_t* audioBuffer,
259 return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
  /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];
  /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;
  /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;
  /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;
  /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);
  /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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