| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| audio_decoder_pcm.h | 21 explicit AudioDecoderPcmU(size_t num_channels) : num_channels_(num_channels) {
22 RTC_DCHECK_GE(num_channels, 1u);
42 explicit AudioDecoderPcmA(size_t num_channels) : num_channels_(num_channels) {
43 RTC_DCHECK_GE(num_channels, 1u);
|
| /external/webrtc/webrtc/common_audio/include/ |
| audio_util.h | 74 int num_channels,
76 for (int i = 0; i < num_channels; ++i) {
90 size_t num_channels,
92 for (size_t i = 0; i < num_channels; ++i) {
97 interleaved_idx += num_channels;
104 // (|samples_per_channel| * |num_channels|).
108 size_t num_channels,
110 for (size_t i = 0; i < num_channels; ++i) {
115 interleaved_idx += num_channels;
122 // |interleaved| (|samples_per_channel| * |num_channels|)
[all...] |
| /external/webrtc/webrtc/common_audio/ |
| audio_ring_buffer_unittest.cc | 27 const size_t num_channels = input.num_channels(); local
29 AudioRingBuffer buf(num_channels, buffer_frames);
30 rtc::scoped_ptr<float* []> slice(new float* [num_channels]);
37 buf.Write(input.Slice(slice.get(), input_pos), num_channels,
44 buf.Read(output->Slice(slice.get(), output_pos), num_channels,
52 buf.Write(input.Slice(slice.get(), input_pos), num_channels,
56 buf.Read(output->Slice(slice.get(), output_pos), num_channels,
64 const size_t num_channels = ::testing::get<3>(GetParam()); local
67 ChannelBuffer<float> input(kFrames, static_cast<int>(num_channels));
[all...] |
| channel_buffer.cc | 16 size_t num_channels,
19 ibuf_(num_frames, num_channels, num_bands),
21 fbuf_(num_frames, num_channels, num_bands) {}
50 for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
64 for (size_t i = 0; i < ibuf_.num_channels(); ++i) {
|
| wav_header.cc | 62 bool CheckWavParameters(size_t num_channels,
67 // num_channels, sample_rate, and bytes_per_sample must be positive, must fit
70 if (num_channels == 0 || sample_rate <= 0 || bytes_per_sample == 0)
74 if (num_channels > std::numeric_limits<uint16_t>::max())
79 if (static_cast<uint64_t>(sample_rate) * num_channels * bytes_per_sample >
108 if (num_samples % num_channels != 0)
138 static inline uint32_t ByteRate(size_t num_channels, int sample_rate,
140 return static_cast<uint32_t>(num_channels * sample_rate * bytes_per_sample);
143 static inline uint16_t BlockAlign(size_t num_channels,
145 return static_cast<uint16_t>(num_channels * bytes_per_sample)
[all...] |
| /external/adhd/cras/src/server/ |
| cras_audio_area.c | 13 struct cras_audio_area *cras_audio_area_create(int num_channels)
18 sz = sizeof(*area) + num_channels * sizeof(struct cras_channel_area);
20 area->num_channels = num_channels;
39 for (src_idx = 0; src_idx < src->num_channels; src_idx++) {
41 for (dst_idx = 0; dst_idx < dst->num_channels; dst_idx++) {
77 if ((fmt->num_channels == 1) &&
85 for (i = 0; i < fmt->num_channels; i++) {
102 for (i = 0 ; i < area->num_channels; i++) {
|
| cras_fmt_conv.c | 299 unsigned int num_in_ch = conv->in_fmt.num_channels;
300 unsigned int num_out_ch = conv->out_fmt.num_channels;
365 for (i = 0; i < conv->out_fmt.num_channels; i++)
369 conv->in_fmt.num_channels);
370 in_idx += conv->in_fmt.num_channels;
371 out_idx += conv->out_fmt.num_channels;
488 if (in->num_channels != out->num_channels) {
491 in->num_channels, out->num_channels);
[all...] |
| /external/libxaac/decoder/drc_src/ |
| impd_drc_peak_limiter.c | 37 FLOAT32 limit_threshold, UWORD32 num_channels,
55 peak_limiter->num_channels = num_channels;
74 peak_limiter->num_channels *
87 UWORD32 num_channels = peak_limiter->num_channels; local
101 for (j = 0; j < num_channels; j++) {
102 tmp = max(tmp, (FLOAT32)fabs(samples[i * num_channels + j]));
142 for (j = 0; j < num_channels; j++) {
143 tmp = delayed_input[delayed_input_index * num_channels + j]
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/ |
| splitting_filter.cc | 19 SplittingFilter::SplittingFilter(size_t num_channels,
25 two_bands_states_.resize(num_channels);
27 for (size_t i = 0; i < num_channels; ++i) {
36 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels());
49 RTC_DCHECK_EQ(data->num_channels(), bands->num_channels());
61 RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
74 RTC_DCHECK_EQ(two_bands_states_.size(), data->num_channels());
87 RTC_DCHECK_EQ(three_band_filter_banks_.size(), data->num_channels());
[all...] |
| /external/autotest/server/site_tests/brillo_PlaybackAudioTest/ |
| brillo_PlaybackAudioTest.py | 70 num_channels, play_file_path=None):
78 @param num_channels: Number of channels to test playback with.
83 num_channels=num_channels)
95 num_channels):
103 @param num_channels: Number of channels to test playback with.
111 num_channels=num_channels)
117 self.host, num_channels, sample_rate, sample_width,
128 num_channels=num_channels
[all...] |
| /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);
|
| /hardware/qcom/msm8994/original-kernel-headers/linux/ |
| msm_audio_amrwbplus.h | 12 unsigned int num_channels; member in struct:msm_audio_amrwbplus_config_v2
|
| /hardware/qcom/msm8996/original-kernel-headers/linux/ |
| msm_audio_amrwbplus.h | 12 unsigned int num_channels; member in struct:msm_audio_amrwbplus_config_v2
|
| /hardware/qcom/msm8998/original-kernel-headers/linux/ |
| msm_audio_amrwbplus.h | 12 unsigned int num_channels; member in struct:msm_audio_amrwbplus_config_v2
|
| /hardware/qcom/msm8x84/original-kernel-headers/linux/ |
| msm_audio_amrwbplus.h | 12 unsigned int num_channels; member in struct:msm_audio_amrwbplus_config_v2
|
| /external/adhd/cras/src/common/ |
| cras_audio_format.h | 76 // TODO(hychao): use channel_layout to replace num_channels
77 size_t num_channels; member in struct:cras_audio_format
94 uint32_t num_channels; member in struct:cras_audio_format_packed
103 dest->num_channels = src->num_channels;
113 dest->num_channels = src->num_channels;
119 * This is bits per smaple / 8 * num_channels.
124 return (size_t)bytes * fmt->num_channels;
130 size_t num_channels);
[all...] |
| /external/autotest/server/site_tests/brillo_RecordingAudioTest/ |
| brillo_RecordingAudioTest.py | 41 sample_rate, num_channels, rec_file):
49 @param num_channels: Number of channels to use for recording.
59 '--duration_secs=%d --num_channels=%d --sample_rate=%d '
61 (duration_secs, num_channels, sample_rate, sample_width,
65 '--duration_secs=%d --num_channels=%d --sample_rate=%d '
67 (duration_secs, num_channels, sample_rate, rec_file))
70 (num_channels, duration_secs, sample_rate, sample_width,
77 sample_rate, num_channels, duration_secs):
85 @param num_channels: Number of channels to use for recording.
97 num_channels=num_channels
[all...] |
| /external/webrtc/webrtc/common_audio/resampler/include/ |
| resampler.h | 31 Resampler(int inFreq, int outFreq, size_t num_channels);
35 int Reset(int inFreq, int outFreq, size_t num_channels);
38 int ResetIfNeeded(int inFreq, int outFreq, size_t num_channels);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
| audio_decoder_pcm16b.cc | 18 AudioDecoderPcm16B::AudioDecoderPcm16B(size_t num_channels)
19 : num_channels_(num_channels) {
20 RTC_DCHECK_GE(num_channels, 1u);
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/mock/ |
| mock_expand.h | 27 size_t num_channels)
33 num_channels) {}
60 size_t num_channels));
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| neteq_external_decoder_test.cc | 51 size_t num_channels; local
56 &num_channels,
58 EXPECT_EQ(channels_, num_channels);
|
| /external/adhd/cras/src/tests/ |
| fmt_conv_unittest.cc | 51 in_fmt.num_channels = out_fmt.num_channels = 2;
71 in_fmt.num_channels = 1;
72 out_fmt.num_channels = 2;
122 in_fmt.num_channels = 2;
123 out_fmt.num_channels = 1;
180 in_fmt.num_channels = 2;
181 out_fmt.num_channels = 1;
233 in_fmt.num_channels = 6;
234 out_fmt.num_channels = 2
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/transient/ |
| transient_suppression_test.cc | 56 DEFINE_int32(num_channels, 1, "Number of channels.");
65 "num_channels and sample_rate_hz, the detection signal from the\n"
79 int num_channels,
88 if (num_channels > 1) {
89 tmpbuf.reset(new int16_t[num_channels * audio_buffer_size]);
94 num_channels * audio_buffer_size,
95 in_file) != num_channels * audio_buffer_size) {
99 if (num_channels > 1) {
100 for (int i = 0; i < num_channels; ++i) {
103 read_ptr[i + j * num_channels];
[all...] |
| /external/adhd/cras/src/libcras/ |
| cras_helpers.c | 116 unsigned int num_channels,
124 aud_format = cras_audio_format_create(format, frame_rate, num_channels);
152 unsigned int num_channels,
161 data->frame_bytes = num_channels * PCM_FORMAT_WIDTH(format) / 8;
167 frame_rate, num_channels, dev_idx, &stream_id);
|
| /external/webrtc/webrtc/common_audio/resampler/ |
| push_resampler.cc | 35 size_t num_channels) {
38 num_channels == num_channels_)
43 num_channels <= 0 || num_channels > 2)
48 num_channels_ = num_channels;
|
