/external/chromium_org/third_party/webrtc/common_audio/ |
H A D | wav_header.h | 27 bool CheckWavParameters(int num_channels, 38 int num_channels,
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H A D | wav_writer.h | 27 WavFile(const std::string& filename, int sample_rate, int num_channels); 39 int num_channels() const { return num_channels_; } function in class:webrtc::WavFile 59 int num_channels);
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H A D | wav_header.cc | 31 bool CheckWavParameters(int num_channels, argument 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) 43 if (static_cast<uint64_t>(num_channels) > 49 if (static_cast<uint64_t>(sample_rate) * num_channels * bytes_per_sample > 79 if (num_samples % num_channels != 0) 99 int num_channels, 104 assert(CheckWavParameters(num_channels, sample_rate, format, 137 WriteLE16(&header.fmt.NumChannels, num_channels); 139 WriteLE32(&header.fmt.ByteRate, (static_cast<uint32_t>(num_channels) 98 WriteWavHeader(uint8_t* buf, int num_channels, int sample_rate, WavFormat format, int bytes_per_sample, uint32_t num_samples) argument [all...] |
H A D | wav_writer.cc | 27 WavFile::WavFile(const std::string& filename, int sample_rate, int num_channels) argument 29 num_channels_(num_channels), 90 int num_channels) { 92 new webrtc::WavFile(filename, sample_rate, num_channels)); 110 return reinterpret_cast<const webrtc::WavFile*>(wf)->num_channels(); 88 rtc_WavOpen(const char* filename, int sample_rate, int num_channels) argument
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/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
H A D | common.h | 41 ChannelBuffer(int samples_per_channel, int num_channels) argument 42 : data_(new T[samples_per_channel * num_channels]), 43 channels_(new T*[num_channels]), 45 num_channels_(num_channels) { 49 ChannelBuffer(const T* data, int samples_per_channel, int num_channels) argument 50 : data_(new T[samples_per_channel * num_channels]), 51 channels_(new T*[num_channels]), 53 num_channels_(num_channels) { 59 int num_channels) 60 : data_(new T[samples_per_channel * num_channels]), 58 ChannelBuffer(const T* const* channels, int samples_per_channel, int num_channels) argument 93 int num_channels() const { return num_channels_; } function in class:webrtc::ChannelBuffer [all...] |
H A D | audio_processing_impl.cc | 153 fwd_in_format_.num_channels(), 154 fwd_proc_format_.num_channels(), 155 rev_in_format_.num_channels()); 175 rev_in_format_.num_channels(), 177 rev_proc_format_.num_channels(), 180 fwd_in_format_.num_channels(), 182 fwd_proc_format_.num_channels(), 285 num_input_channels == fwd_in_format_.num_channels() && 286 num_output_channels == fwd_proc_format_.num_channels() && 287 num_reverse_channels == rev_in_format_.num_channels()) { 524 const int num_channels = ChannelsFromLayout(layout); local [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
H A D | accelerate.h | 32 Accelerate(int sample_rate_hz, size_t num_channels, argument 34 : TimeStretch(sample_rate_hz, num_channels, background_noise) { 72 size_t num_channels,
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H A D | preemptive_expand.h | 33 size_t num_channels, 36 : TimeStretch(sample_rate_hz, num_channels, background_noise), 81 size_t num_channels, 32 PreemptiveExpand(int sample_rate_hz, size_t num_channels, const BackgroundNoise& background_noise, int overlap_samples) argument
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/mock/ |
H A D | mock_expand.h | 26 size_t num_channels) 27 : Expand(background_noise, sync_buffer, random_vector, fs, num_channels) { 54 size_t num_channels)); 22 MockExpand(BackgroundNoise* background_noise, SyncBuffer* sync_buffer, RandomVector* random_vector, int fs, size_t num_channels) argument
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/external/chromium_org/third_party/webrtc/common_audio/include/ |
H A D | audio_util.h | 65 int num_channels, T* const* deinterleaved) { 66 for (int i = 0; i < num_channels; ++i) { 71 interleaved_idx += num_channels; 78 // (|samples_per_channel| * |num_channels|). 81 int num_channels, T* interleaved) { 82 for (int i = 0; i < num_channels; ++i) { 87 interleaved_idx += num_channels; 64 Deinterleave(const T* interleaved, int samples_per_channel, int num_channels, T* const* deinterleaved) argument 80 Interleave(const T* const* deinterleaved, int samples_per_channel, int num_channels, T* interleaved) argument
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/external/chromium_org/media/audio/ |
H A D | audio_power_monitor.cc | 35 const int num_channels = buffer.channels(); local 36 if (num_frames <= 0 || num_channels <= 0) 43 for (int i = 0; i < num_channels; ++i) { 56 average_power_ = std::max(0.0f, std::min(1.0f, sum_power / num_channels));
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/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | utility.cc | 76 int num_channels, 84 assert(num_channels == 1 || num_channels == 2); 94 if (num_channels == 2 && codec_num_channels == 1) { 98 num_channels = 1; 102 sample_rate_hz, destination_rate, num_channels) != 0) { 107 num_channels); 111 const int in_length = samples_per_channel * num_channels; 119 dst_af->samples_per_channel_ = out_length / num_channels; 121 dst_af->num_channels_ = num_channels; 74 DownConvertToCodecFormat(const int16_t* src_data, int samples_per_channel, int num_channels, int sample_rate_hz, int codec_num_channels, int codec_rate_hz, int16_t* mono_buffer, PushResampler<int16_t>* resampler, AudioFrame* dst_af) argument [all...] |
H A D | utility.h | 46 int num_channels,
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/external/chromium_org/third_party/webrtc/common_audio/resampler/ |
H A D | push_resampler.cc | 35 int num_channels) { 38 num_channels == num_channels_) 43 num_channels <= 0 || num_channels > 2) 48 num_channels_ = num_channels; 33 InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz, int num_channels) argument
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/external/chromium_org/third_party/libwebp/utils/ |
H A D | rescaler.h | 26 int num_channels; // bytes to jump between pixels member in struct:__anon13303 45 int num_channels,
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H A D | rescaler.c | 31 const int x_stride = wrk->num_channels; 32 const int x_out_max = wrk->dst_width * wrk->num_channels; 78 const int x_out_max = wrk->dst_width * wrk->num_channels; 97 const int x_stride = wrk->num_channels; 98 const int x_out_max = wrk->dst_width * wrk->num_channels; 198 const int x_out_max = wrk->dst_width * wrk->num_channels; 260 int dst_stride, int num_channels, int x_add, int x_sub, 269 wrk->num_channels = num_channels; 282 wrk->frow = work + num_channels * dst_widt 258 WebPRescalerInit(WebPRescaler* const wrk, int src_width, int src_height, uint8_t* const dst, int dst_width, int dst_height, int dst_stride, int num_channels, int x_add, int x_sub, int y_add, int y_sub, int32_t* const work) argument [all...] |
/external/chromium_org/third_party/mesa/src/src/gallium/auxiliary/vl/ |
H A D | vl_zscan.h | 45 unsigned num_channels; member in struct:vl_zscan 78 unsigned num_channels);
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/external/chromium_org/third_party/webrtc/modules/audio_processing/test/ |
H A D | test_utils.h | 64 int num_channels, 67 size_t length = num_channels * samples_per_channel; 69 Interleave(data, samples_per_channel, num_channels, buffer.get()); 107 int num_channels, 111 frame->num_channels_ = num_channels; 112 cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels)); 116 int num_channels) { 117 switch (num_channels) { 62 WriteFloatData(const float* const* data, size_t samples_per_channel, int num_channels, WavFile* wav_file, RawFile* raw_file) argument 106 SetContainerFormat(int sample_rate_hz, int num_channels, AudioFrame* frame, scoped_ptr<ChannelBuffer<T> >* cb) argument 115 LayoutFromChannels( int num_channels) argument
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/external/mesa3d/src/gallium/auxiliary/vl/ |
H A D | vl_zscan.h | 45 unsigned num_channels; member in struct:vl_zscan 78 unsigned num_channels);
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/external/webp/src/utils/ |
H A D | rescaler.h | 26 int num_channels; // bytes to jump between pixels member in struct:__anon33400 45 int num_channels,
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/external/chromium_org/third_party/webrtc/voice_engine/include/mock/ |
H A D | fake_voe_external_media.h | 54 int num_channels) { 55 const int length = samples_per_channel * num_channels; 67 num_channels == 2 ? true : false); 52 CallProcess(ProcessingTypes type, int16_t* audio, int samples_per_channel, int sample_rate_hz, int num_channels) argument
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/external/chromium_org/media/cast/sender/ |
H A D | audio_encoder.cc | 48 int num_channels, 53 num_channels_(num_channels), 204 int num_channels, 210 num_channels, 213 encoder_memory_(new uint8[opus_encoder_get_size(num_channels)]), 215 buffer_(new float[num_channels * samples_per_frame_]) { 220 num_channels, 295 int num_channels, 300 num_channels, 303 buffer_(new int16[num_channels * samples_per_frame 46 ImplBase(const scoped_refptr<CastEnvironment>& cast_environment, Codec codec, int num_channels, int sampling_rate, const FrameEncodedCallback& callback) argument 203 OpusImpl(const scoped_refptr<CastEnvironment>& cast_environment, int num_channels, int sampling_rate, int bitrate, const FrameEncodedCallback& callback) argument 294 Pcm16Impl(const scoped_refptr<CastEnvironment>& cast_environment, int num_channels, int sampling_rate, const FrameEncodedCallback& callback) argument 340 AudioEncoder( const scoped_refptr<CastEnvironment>& cast_environment, int num_channels, int sampling_rate, int bitrate, Codec codec, const FrameEncodedCallback& frame_encoded_callback) argument [all...] |
/external/chromium_org/media/cast/test/utility/ |
H A D | audio_utility.h | 25 TestAudioBusFactory(int num_channels,
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/external/chromium_org/third_party/webrtc/common_audio/resampler/include/ |
H A D | push_resampler.h | 32 int num_channels);
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/external/chromium_org/media/cast/receiver/ |
H A D | audio_decoder.cc | 27 int num_channels, 31 num_channels_(num_channels), 96 int num_channels, 100 num_channels, 102 decoder_memory_(new uint8[opus_decoder_get_size(num_channels)]), 106 buffer_(new float[max_samples_per_frame_ * num_channels]) { 109 if (opus_decoder_init(opus_decoder_, sampling_rate, num_channels) != 166 int num_channels, 170 num_channels, 25 ImplBase(const scoped_refptr<CastEnvironment>& cast_environment, Codec codec, int num_channels, int sampling_rate) argument 95 OpusImpl(const scoped_refptr<CastEnvironment>& cast_environment, int num_channels, int sampling_rate) argument 165 Pcm16Impl(const scoped_refptr<CastEnvironment>& cast_environment, int num_channels, int sampling_rate) argument
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