/external/chromium_org/media/filters/ |
audio_renderer_algorithm.h | 43 // data from our |audio_buffer_|. Data is scaled based on |playback_rate|, 46 // Data from |audio_buffer_| is consumed in proportion to the playback rate. 51 // Clears |audio_buffer_|. 58 // Returns true if |audio_buffer_| is at or exceeds capacity. 61 // Returns the capacity of |audio_buffer_| in frames. 64 // Increase the capacity of |audio_buffer_| if possible. 67 // Returns the number of frames left in |audio_buffer_|, which may be larger 69 // than |audio_buffer_| was intending to hold. 70 int frames_buffered() { return audio_buffer_.frames(); } 87 // Fill |dest| with frames from |audio_buffer_| starting from fram 123 AudioBufferQueue audio_buffer_; member in class:media::AudioRendererAlgorithm [all...] |
audio_renderer_algorithm.cc | 63 // The maximum size in seconds for the |audio_buffer_|. Arbitrarily determined. 66 // The starting size in frames for |audio_buffer_|. Previous usage maintained a 156 std::min(static_cast<int>(audio_buffer_.frames() / playback_rate), 166 audio_buffer_.SeekFrames(seek_frames); 184 std::min(audio_buffer_.frames(), requested_frames); 185 const int frames_read = audio_buffer_.ReadFrames(frames_to_copy, 0, dest); 201 audio_buffer_.Clear(); 216 audio_buffer_.Append(buffer_in); 220 return audio_buffer_.frames() >= capacity_; 232 const int frames = audio_buffer_.frames() [all...] |
/external/chromium_org/third_party/webrtc/examples/android/opensl_loopback/jni/ |
opensl_runner.cc | 34 output_.AttachAudioBuffer(&audio_buffer_); 41 input_.AttachAudioBuffer(&audio_buffer_); 63 audio_buffer_.ClearBuffer(); 69 FakeAudioDeviceBuffer audio_buffer_; member in class:webrtc::OpenSlRunnerTemplate
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/external/chromium_org/third_party/webrtc/modules/audio_device/android/ |
opensles_input.cc | 62 audio_buffer_(NULL), 243 audio_buffer_ = audioBuffer; 248 audio_buffer_->SetRecordingSampleRate(rec_sampling_rate_); 249 audio_buffer_->SetRecordingChannels(kNumChannels); 517 audio_buffer_->SetRecordedBuffer(audio, buffer_size_samples()); 518 audio_buffer_->SetVQEData(delay_provider_->PlayoutDelayMs(), 520 audio_buffer_->DeliverRecordedData();
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opensles_output.cc | 61 audio_buffer_(NULL), 269 audio_buffer_ = audioBuffer; 291 if (audio_buffer_->SetPlayoutSampleRate(speaker_sampling_rate_) < 0) { 294 if (audio_buffer_->SetPlayoutChannels(kNumChannels) < 0) { 333 fine_buffer_.reset(new FineAudioBuffer(audio_buffer_, buffer_size_bytes_,
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opensles_input.h | 206 AudioDeviceBuffer* audio_buffer_; member in class:webrtc::OpenSlesInput
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opensles_output.h | 224 AudioDeviceBuffer* audio_buffer_; member in class:webrtc::OpenSlesOutput
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/external/chromium_org/media/audio/alsa/ |
alsa_input.cc | 91 audio_buffer_.reset(new uint8[bytes_per_buffer_]); 210 int frames_read = wrapper_->PcmReadi(device_handle_, audio_buffer_.get(), 213 audio_bus_->FromInterleaved(audio_buffer_.get(), 267 audio_buffer_.reset();
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alsa_input.h | 84 scoped_ptr<uint8[]> audio_buffer_; // Buffer used for reading audio data. member in class:media::AlsaPcmInputStream
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/external/chromium_org/media/ffmpeg/ |
ffmpeg_unittest.cc | 94 audio_buffer_.reset(av_frame_alloc()); 236 av_frame_unref(audio_buffer_.get()); 237 result = avcodec_decode_audio4(av_audio_context(), audio_buffer_.get(), 249 double microseconds = 1.0L * audio_buffer_->nb_samples / 385 scoped_ptr<AVFrame, media::ScopedPtrAVFreeFrame> audio_buffer_; member in class:media::FFmpegTest
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
acm_receiver.cc | 371 ptr_audio_buffer = audio_buffer_; 412 if (ptr_audio_buffer == audio_buffer_) { 416 resampler_.Resample10Msec(audio_buffer_, 428 memcpy(audio_frame->data_, audio_buffer_, samples_per_channel * 441 audio_buffer_); 446 memcpy(audio_frame->data_, audio_buffer_, samples_per_channel *
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acm_receiver.h | 337 int16_t audio_buffer_[AudioFrame::kMaxDataSizeSamples] GUARDED_BY(crit_sect_);
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