/external/webrtc/webrtc/modules/audio_coding/neteq/ |
H A D | background_noise_unittest.cc | 20 size_t channels = 1; local 21 BackgroundNoise bgn(channels);
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H A D | merge_unittest.cc | 28 size_t channels = 1; local 29 BackgroundNoise bgn(channels); 33 Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels); 34 Merge merge(fs, channels, &expand, &sync_buffer);
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H A D | normal_unittest.cc | 36 size_t channels = 1; local 37 BackgroundNoise bgn(channels); 41 Expand expand(&bgn, &sync_buffer, &random_vector, &statistics, fs, channels); 50 size_t channels = 1; local 51 BackgroundNoise bgn(channels); 56 channels); 60 rtc::scoped_ptr<int16_t[]> mute_factor_array(new int16_t[channels]); 61 for (size_t i = 0; i < channels; ++i) { 64 AudioMultiVector output(channels); 96 size_t channels local [all...] |
H A D | audio_classifier.cc | 42 int channels) { 44 assert((input_length / channels) == kDefaultFrameSizeSamples); 47 assert(channels == 1 || channels == 2); 61 channels, 40 Analysis(const int16_t* input, int input_length, int channels) argument
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H A D | audio_classifier_unittest.cc | 40 size_t channels) { 42 rtc::scoped_ptr<int16_t[]> in(new int16_t[channels * kFrameSize]); 51 while (fread(in.get(), sizeof(int16_t), channels * kFrameSize, audio_file) == 52 channels * kFrameSize) { 54 classifier.Analysis(in.get(), channels * kFrameSize, channels); 38 RunAnalysisTest(const std::string& audio_filename, const std::string& data_filename, size_t channels) argument
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H A D | sync_buffer.h | 22 SyncBuffer(size_t channels, size_t length) argument 23 : AudioMultiVector(channels, length),
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/external/toybox/toys/other/ |
H A D | mix.c | 14 List OSS sound channels (module snd-mixer-oss), or set volume(s). 35 const char *channels[SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES; local 45 if (!strcmp(channels[channel], TT.chan)) break; 47 else printf("%s\n", channels[channel]); 58 TT.dev, channels[channel], level>>8, level & 0xFF); 59 else xprintf("%s:%s = %d\n", TT.dev, channels[channel], level);
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/external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
H A D | opus_inst.h | 20 size_t channels; member in struct:WebRtcOpusEncInst 33 size_t channels; member in struct:WebRtcOpusDecInst
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/external/webrtc/webrtc/test/fuzzers/ |
H A D | audio_decoder_opus_fuzzer.cc | 16 const size_t channels = (size % 2) + 1; // 1 or 2 channels. local 17 AudioDecoderOpus dec(channels);
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/external/ImageMagick/coders/ |
H A D | psd-private.h | 38 channels, member in struct:_PSDInfo
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/external/flac/libFLAC/include/protected/ |
H A D | stream_decoder.h | 44 unsigned channels; member in struct:FLAC__StreamDecoderProtected
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/external/mp4parser/isoparser/src/main/java/com/googlecode/mp4parser/authoring/adaptivestreaming/ |
H A D | AudioQuality.java | 24 int channels; field in class:AudioQuality
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/external/webrtc/webrtc/audio/ |
H A D | audio_sink.h | 33 size_t channels, 38 channels(channels), 44 size_t channels; // Number of channels in the audio data. member in struct:webrtc::AudioSinkInterface::Data 30 Data(int16_t* data, size_t samples_per_channel, int sample_rate, size_t channels, uint32_t timestamp) argument
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
H A D | acm_receiver.h | 47 size_t channels; member in struct:webrtc::acm2::AcmReceiver::Decoder 119 size_t channels,
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H A D | audio_coding_module.cc | 59 size_t channels) { 61 payload_name, sampling_freq_hz, channels); 79 size_t channels) { 82 channels); 56 Codec(const char* payload_name, CodecInst* codec, int sampling_freq_hz, size_t channels) argument 77 Codec(const char* payload_name, int sampling_freq_hz, size_t channels) argument
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/external/libpng/contrib/tools/ |
H A D | cvtcolor.c | 53 int channels = 0; local 92 ++channels; 95 ++channels; 98 ++channels; 101 ++channels; 108 int components = channels; 114 if (components < channels) 123 if ((channels & 1) == 0) 125 double alpha = c[channels-1]; 128 for (i=0; i<channels [all...] |
/external/webrtc/webrtc/common_audio/ |
H A D | audio_ring_buffer.cc | 20 AudioRingBuffer::AudioRingBuffer(size_t channels, size_t max_frames) { argument 21 buffers_.reserve(channels); 22 for (size_t i = 0; i < channels; ++i) 31 void AudioRingBuffer::Write(const float* const* data, size_t channels, argument 33 RTC_DCHECK_EQ(buffers_.size(), channels); 34 for (size_t i = 0; i < channels; ++i) { 40 void AudioRingBuffer::Read(float* const* data, size_t channels, size_t frames) { argument 41 RTC_DCHECK_EQ(buffers_.size(), channels); 42 for (size_t i = 0; i < channels; ++i) {
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/external/webrtc/webrtc/modules/audio_coding/neteq/test/ |
H A D | audio_classifier_test.cc | 26 " channels output_type <input file name> <output file name> " 28 std::cout << "Where channels can be 1 (mono) or 2 (interleaved stereo),"; 38 int channels = atoi(argv[1]); local 39 if (channels < 1 || channels > 2) { 40 std::cout << "Disallowed number of channels " << channels << std::endl; 50 const int data_size = channels * kFrameSizeSamples; 76 bool is_music = classifier.Analysis(in.get(), data_size, channels);
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/external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
H A D | input_audio_file.cc | 64 size_t channels, 70 for (int j = static_cast<int>(channels - 1); j >= 0; --j) { 71 destination[i * channels + j] = source[i]; 63 DuplicateInterleaved(const int16_t* source, size_t samples, size_t channels, int16_t* destination) argument
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/external/webrtc/webrtc/sound/ |
H A D | soundsysteminterface.h | 52 // Number of channels in the PCM stream. 53 unsigned int channels; member in struct:rtc::SoundSystemInterface::OpenParams
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/external/kernel-headers/original/uapi/linux/hsi/ |
H A D | hsi_char.h | 54 __u32 channels; member in struct:hsc_rx_config 59 __u32 channels; member in struct:hsc_tx_config
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/external/libopus/celt/ |
H A D | opus_custom_demo.c | 51 opus_int32 frame_size, channels, rate; local 64 fprintf (stderr, "Usage: test_opus_custom <rate> <channels> <frame size> " 71 channels = atoi(argv[2]); 104 enc = opus_custom_encoder_create(mode, channels, &err); 112 dec = opus_custom_decoder_create(mode, channels, &err); 128 in = (opus_int16*)malloc(frame_size*channels*sizeof(opus_int16)); 129 out = (opus_int16*)malloc(frame_size*channels*sizeof(opus_int16)); 134 err = fread(in, sizeof(short), frame_size*channels, fin); 174 for (i=0;i<ret*channels;i++) 178 for (i=0;i<ret*channels; [all...] |
/external/libpng/contrib/gregbook/ |
H A D | readppm.c | 68 int bit_depth, color_type, channels; variable 97 channels = 3; 100 channels = 4; 103 channels = 1; 152 /* GRR WARNING: grayscale needs to be expanded and channels reset! */ 154 *pRowbytes = rowbytes = channels*width; 155 *pChannels = channels;
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/external/libpng/ |
H A D | pngwtran.c | 21 * row_info bit depth should be 8 (one pixel per byte). The channels 30 row_info->channels == 1) 156 row_info->pixel_depth = (png_byte)(bit_depth * row_info->channels); 180 int channels = 0; local 184 shift_start[channels] = row_info->bit_depth - bit_depth->red; 185 shift_dec[channels] = bit_depth->red; 186 channels++; 188 shift_start[channels] = row_info->bit_depth - bit_depth->green; 189 shift_dec[channels] = bit_depth->green; 190 channels [all...] |
/external/pdfium/third_party/libpng16/ |
H A D | pngwtran.c | 21 * row_info bit depth should be 8 (one pixel per byte). The channels 30 row_info->channels == 1) 156 row_info->pixel_depth = (png_byte)(bit_depth * row_info->channels); 180 int channels = 0; local 184 shift_start[channels] = row_info->bit_depth - bit_depth->red; 185 shift_dec[channels] = bit_depth->red; 186 channels++; 188 shift_start[channels] = row_info->bit_depth - bit_depth->green; 189 shift_dec[channels] = bit_depth->green; 190 channels [all...] |