/external/chromium_org/third_party/webrtc/common_audio/ |
H A D | wav_header.h | 21 kWavFormatPcm = 1, // PCM, each sample of size bytes_per_sample 30 int bytes_per_sample, 41 int bytes_per_sample,
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H A D | wav_header.cc | 34 int bytes_per_sample, 36 // num_channels, sample_rate, and bytes_per_sample must be positive, must fit 39 if (num_channels <= 0 || sample_rate <= 0 || bytes_per_sample <= 0) 46 if (static_cast<uint64_t>(bytes_per_sample) * 8 > 49 if (static_cast<uint64_t>(sample_rate) * num_channels * bytes_per_sample > 53 // format and bytes_per_sample must agree. 57 if (bytes_per_sample != 1 && bytes_per_sample != 2) 62 if (bytes_per_sample != 1) 74 bytes_per_sample; 31 CheckWavParameters(int num_channels, int sample_rate, WavFormat format, int bytes_per_sample, uint32_t num_samples) argument 98 WriteWavHeader(uint8_t* buf, int num_channels, int sample_rate, WavFormat format, int bytes_per_sample, uint32_t num_samples) argument [all...] |
/external/chromium_org/content/browser/speech/ |
H A D | audio_buffer.cc | 12 AudioChunk::AudioChunk(int bytes_per_sample) argument 13 : bytes_per_sample_(bytes_per_sample) { 16 AudioChunk::AudioChunk(const uint8* data, size_t length, int bytes_per_sample) argument 18 bytes_per_sample_(bytes_per_sample) { 19 DCHECK_EQ(length % bytes_per_sample, 0U); 44 AudioBuffer::AudioBuffer(int bytes_per_sample) 45 : bytes_per_sample_(bytes_per_sample) { 46 DCHECK(bytes_per_sample == 1 || 47 bytes_per_sample == 2 || 48 bytes_per_sample [all...] |
H A D | audio_buffer.h | 21 explicit AudioChunk(int bytes_per_sample); 22 AudioChunk(const uint8* data, size_t length, int bytes_per_sample); 25 int bytes_per_sample() const { return bytes_per_sample_; } function in class:content::AudioChunk 46 explicit AudioBuffer(int bytes_per_sample);
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/external/flac/libFLAC/include/private/ |
H A D | md5.h | 42 FLAC__bool FLAC__MD5Accumulate(FLAC__MD5Context *ctx, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample);
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/external/chromium_org/media/base/ |
H A D | audio_block_fifo.cc | 27 int bytes_per_sample) { 30 DCHECK_GT(bytes_per_sample, 0); 47 source_ptr, write_pos_, push_frames, bytes_per_sample); 56 source_ptr += push_frames * bytes_per_sample * channels_; 25 Push(const void* source, int frames, int bytes_per_sample) argument
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H A D | audio_block_fifo.h | 29 void Push(const void* source, int frames, int bytes_per_sample);
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H A D | audio_bus.h | 59 // |bytes_per_sample| per value. Values are scaled and bias corrected during 63 void FromInterleaved(const void* source, int frames, int bytes_per_sample); 64 void ToInterleaved(int frames, int bytes_per_sample, void* dest) const; 65 void ToInterleavedPartial(int start_frame, int frames, int bytes_per_sample, 73 int bytes_per_sample);
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H A D | audio_bus.cc | 241 int frames, int bytes_per_sample) { 243 switch (bytes_per_sample) { 273 int bytes_per_sample) { 274 FromInterleavedPartial(source, 0, frames, bytes_per_sample); 277 void AudioBus::ToInterleaved(int frames, int bytes_per_sample, argument 279 ToInterleavedPartial(0, frames, bytes_per_sample, dest); 284 int bytes_per_sample, void* dest) const { 286 switch (bytes_per_sample) { 301 memset(dest, 0, frames * bytes_per_sample); 240 FromInterleavedPartial(const void* source, int start_frame, int frames, int bytes_per_sample) argument 272 FromInterleaved(const void* source, int frames, int bytes_per_sample) argument 283 ToInterleavedPartial(int start_frame, int frames, int bytes_per_sample, void* dest) const argument
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H A D | audio_block_fifo_unittest.cc | 31 const int bytes_per_sample = 2; local 32 const int data_byte_size = bytes_per_sample * channels * frames_to_push; 35 fifo->Push(data.get(), frames_to_push, bytes_per_sample);
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/external/chromium_org/remoting/codec/ |
H A D | audio_encoder_verbatim.cc | 21 DCHECK_NE(AudioPacket::BYTES_PER_SAMPLE_INVALID, packet->bytes_per_sample());
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H A D | audio_decoder_verbatim.cc | 24 (packet->bytes_per_sample() != AudioPacket::BYTES_PER_SAMPLE_2) ||
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H A D | audio_decoder_opus.cc | 105 decoded_packet->bytes_per_sample(); 128 decoded_packet->bytes_per_sample();
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/external/qemu/distrib/sdl-1.2.15/src/audio/paudio/ |
H A D | SDL_paudio.c | 237 int bytes_per_sample; local 330 bytes_per_sample = 1; 336 bytes_per_sample = 1; 343 bytes_per_sample = 2; 350 bytes_per_sample = 2; 358 bytes_per_sample = 2; 364 bytes_per_sample = 2; 399 / bytes_per_sample 405 / bytes_per_sample 409 paud_init.bsize = bytes_per_sample * spe [all...] |
/external/chromium_org/third_party/libvpx/source/libvpx/test/ |
H A D | md5_helper.h | 31 const int bytes_per_sample = local 36 img->x_chroma_shift : img->d_w) * bytes_per_sample;
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H A D | y4m_test.cc | 60 const int bytes_per_sample = (img->fmt & VPX_IMG_FMT_HIGHBITDEPTH) ? 2 : 1; local 66 fwrite(buf, bytes_per_sample, w, file);
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/external/chromium_org/media/audio/ |
H A D | simple_sources_unittest.cc | 20 static const uint32 bytes_per_sample = 2; local 25 AudioParameters::kTelephoneSampleRate, bytes_per_sample * 8, samples);
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/external/flac/libFLAC/ |
H A D | md5.c | 277 static void format_input_(FLAC__byte *buf, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample) argument 285 if(channels == 2 && bytes_per_sample == 2) { 291 else if(channels == 1 && bytes_per_sample == 2) { 298 if(bytes_per_sample == 2) { 326 else if(bytes_per_sample == 3) { 358 else if(bytes_per_sample == 1) { 382 else { /* bytes_per_sample == 4, maybe optimize more later */ 398 FLAC__bool FLAC__MD5Accumulate(FLAC__MD5Context *ctx, const FLAC__int32 * const signal[], unsigned channels, unsigned samples, unsigned bytes_per_sample) argument 400 const size_t bytes_needed = (size_t)channels * (size_t)samples * (size_t)bytes_per_sample; 403 if((size_t)channels > SIZE_MAX / (size_t)bytes_per_sample) [all...] |
/external/srec/srec/EventLog/include/ |
H A D | riff.h | 62 * number of channels * bytes_per_sample 66 * bytes_per_sample * 8 (PCM-specific field) 229 int bytes_per_sample,
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/external/chromium_org/media/audio/mac/ |
H A D | audio_low_latency_input_mac_unittest.cc | 84 const int bytes_per_sample = sizeof(*interleaved); variable 85 src->ToInterleaved(src->frames(), bytes_per_sample, interleaved.get()); 90 const int size = bytes_per_sample * num_samples;
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/external/chromium_org/content/renderer/media/ |
H A D | webrtc_audio_device_impl.cc | 141 int bytes_per_sample = sizeof(render_buffer_[0]); local 143 audio_bus->channels() * frames_per_10_ms * bytes_per_sample; 162 audio_transport_callback_->PullRenderData(bytes_per_sample * kBitsPerByte, 174 bytes_per_sample, 193 bytes_per_sample);
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/external/chromium_org/remoting/client/ |
H A D | audio_player.cc | 38 DCHECK_EQ(kSampleSizeBytes, packet->bytes_per_sample());
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/external/chromium_org/media/filters/ |
H A D | audio_file_reader.cc | 135 size_t bytes_per_sample = av_get_bytes_per_sample(codec_context_->sample_fmt); 228 av_frame->data[0], current_frame, frames_read, bytes_per_sample);
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/external/srec/srec/EventLog/src/ |
H A D | riff.c | 769 /* assuming nchannels = 1, usually bytes_per_sample==blockAlign / nchannels (not aurora!) */ 939 * sampling rate = bytes_per_sample = -1; other fields of WaveFormat undefined 946 int bytes_per_sample, 967 (bytes_per_sample == -1 && format_tag != WAVEFORMAT_AURORA && 974 if (bytes_per_sample > 0) 975 num_samples = num_bytes / bytes_per_sample; 1024 bytes_sec = (short)(rate * num_channels * bytes_per_sample); 1025 block_align = bytes_per_sample * num_channels; 1042 if (bytes_per_sample == 2) 941 convertBuf2Riff( unsigned char *waveform, unsigned int num_bytes, wchar_t *audio_type, int rate, int bytes_per_sample, SwiRiffStruct *swichunk, unsigned char **buf, unsigned int *buflen) argument
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/external/chromium_org/media/audio/win/ |
H A D | audio_low_latency_input_win_unittest.cc | 132 const int bytes_per_sample = sizeof(*interleaved); local 133 src->ToInterleaved(src->frames(), bytes_per_sample, interleaved.get()); 138 const int size = bytes_per_sample * num_samples;
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