Searched refs:num_channels (Results 1 - 25 of 138) sorted by relevance

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/external/chromium_org/third_party/webrtc/common_audio/
H A Dwav_header.h27 bool CheckWavParameters(int num_channels,
38 int num_channels,
H A Dwav_writer.h27 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);
H A Dwav_header.cc31 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
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H A Dwav_writer.cc27 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
/external/chromium_org/third_party/webrtc/modules/audio_processing/
H A Dcommon.h41 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
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H A Daudio_processing_impl.cc153 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
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/
H A Daccelerate.h32 Accelerate(int sample_rate_hz, size_t num_channels, argument
34 : TimeStretch(sample_rate_hz, num_channels, background_noise) {
72 size_t num_channels,
H A Dpreemptive_expand.h33 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
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/mock/
H A Dmock_expand.h26 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
/external/chromium_org/third_party/webrtc/common_audio/include/
H A Daudio_util.h65 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
/external/chromium_org/media/audio/
H A Daudio_power_monitor.cc35 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));
/external/chromium_org/third_party/webrtc/voice_engine/
H A Dutility.cc76 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
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H A Dutility.h46 int num_channels,
/external/chromium_org/third_party/webrtc/common_audio/resampler/
H A Dpush_resampler.cc35 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
/external/chromium_org/third_party/libwebp/utils/
H A Drescaler.h26 int num_channels; // bytes to jump between pixels member in struct:__anon13303
45 int num_channels,
H A Drescaler.c31 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
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/external/chromium_org/third_party/mesa/src/src/gallium/auxiliary/vl/
H A Dvl_zscan.h45 unsigned num_channels; member in struct:vl_zscan
78 unsigned num_channels);
/external/chromium_org/third_party/webrtc/modules/audio_processing/test/
H A Dtest_utils.h64 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
/external/mesa3d/src/gallium/auxiliary/vl/
H A Dvl_zscan.h45 unsigned num_channels; member in struct:vl_zscan
78 unsigned num_channels);
/external/webp/src/utils/
H A Drescaler.h26 int num_channels; // bytes to jump between pixels member in struct:__anon33400
45 int num_channels,
/external/chromium_org/third_party/webrtc/voice_engine/include/mock/
H A Dfake_voe_external_media.h54 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
/external/chromium_org/media/cast/sender/
H A Daudio_encoder.cc48 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
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/external/chromium_org/media/cast/test/utility/
H A Daudio_utility.h25 TestAudioBusFactory(int num_channels,
/external/chromium_org/third_party/webrtc/common_audio/resampler/include/
H A Dpush_resampler.h32 int num_channels);
/external/chromium_org/media/cast/receiver/
H A Daudio_decoder.cc27 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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