/external/chromium_org/third_party/webrtc/modules/audio_processing/ |
common.h | 41 ChannelBuffer(int samples_per_channel, int num_channels) 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) 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]), 93 int num_channels() const { return num_channels_; } function in class:webrtc::ChannelBuffer [all...] |
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/common_audio/ |
wav_header.h | 27 bool CheckWavParameters(int num_channels, 38 int num_channels,
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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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wav_header.cc | 31 bool CheckWavParameters(int num_channels, 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) [all...] |
/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/mock/ |
mock_expand.h | 26 size_t num_channels) 27 : Expand(background_noise, sync_buffer, random_vector, fs, num_channels) { 54 size_t num_channels));
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/frameworks/av/cmds/stagefright/ |
WaveWriter.h | 26 uint16_t num_channels, uint32_t sampling_rate) 30 write_u16(num_channels); 32 write_u32(sampling_rate * num_channels * 2); 33 write_u16(num_channels * 2);
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/hardware/qcom/msm8x84/kernel-headers/linux/ |
msm_audio_amrwbplus.h | 27 unsigned int num_channels; member in struct:msm_audio_amrwbplus_config_v2
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/hardware/qcom/msm8x84/original-kernel-headers/linux/ |
msm_audio_amrwbplus.h | 12 unsigned int num_channels; member in struct:msm_audio_amrwbplus_config_v2
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/external/chromium_org/third_party/webrtc/modules/audio_coding/neteq/ |
accelerate.h | 32 Accelerate(int sample_rate_hz, size_t num_channels, 34 : TimeStretch(sample_rate_hz, num_channels, background_noise) { 72 size_t num_channels,
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preemptive_expand.h | 33 size_t num_channels, 36 : TimeStretch(sample_rate_hz, num_channels, background_noise), 81 size_t num_channels,
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/external/chromium_org/third_party/webrtc/common_audio/include/ |
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;
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/external/chromium_org/media/audio/ |
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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/frameworks/av/media/libstagefright/codecs/mp3dec/src/ |
s_tmp3dec_file.h | 87 int32 num_channels; member in struct:__anon17988
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/frameworks/ex/variablespeed/jni/ |
sola_time_scaler.h | 45 // @param num_channels number of interleaved channels in the signal. 46 void Init(int sample_rate, int num_channels) { 48 num_channels_ = num_channels; 85 // @param num_channels number of channels of the signal to process 89 void Init(double sample_rate, int num_channels, double initial_speed, 107 // @param num_frames number of frames (num_samples / num_channels) 125 int num_channels() const { return num_channels_; } function in class:video_editing::SolaTimeScaler
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/external/chromium_org/third_party/webrtc/voice_engine/ |
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; [all...] |
utility.h | 46 int num_channels,
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/external/chromium_org/third_party/webrtc/common_audio/resampler/ |
push_resampler.cc | 35 int num_channels) { 38 num_channels == num_channels_) 43 num_channels <= 0 || num_channels > 2) 48 num_channels_ = num_channels;
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/external/chromium_org/third_party/libwebp/utils/ |
rescaler.h | 26 int num_channels; // bytes to jump between pixels member in struct:__anon17928 45 int num_channels,
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/external/chromium_org/third_party/mesa/src/src/gallium/auxiliary/vl/ |
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/ |
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) {
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/external/mesa3d/src/gallium/auxiliary/vl/ |
vl_zscan.h | 45 unsigned num_channels; member in struct:vl_zscan 78 unsigned num_channels);
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/external/webp/src/utils/ |
rescaler.h | 26 int num_channels; // bytes to jump between pixels member in struct:__anon17066 45 int num_channels,
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/external/chromium_org/third_party/webrtc/common_audio/resampler/include/ |
push_resampler.h | 32 int num_channels);
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/external/chromium_org/third_party/webrtc/voice_engine/include/mock/ |
fake_voe_external_media.h | 54 int num_channels) { 55 const int length = samples_per_channel * num_channels; 67 num_channels == 2 ? true : false);
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