| /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...] |
| wav_header.h | 35 bool CheckWavParameters(size_t num_channels,
46 size_t num_channels,
56 size_t* num_channels,
|
| channel_buffer.h | 43 size_t num_channels,
45 : data_(new T[num_frames * num_channels]()),
46 channels_(new T*[num_channels * num_bands]),
47 bands_(new T*[num_channels * num_bands]),
50 num_channels_(num_channels),
118 size_t num_channels() const { return num_channels_; } function in class:webrtc::ChannelBuffer
145 IFChannelBuffer(size_t num_frames, size_t num_channels, size_t num_bands = 1);
154 size_t num_channels() const { return ibuf_.num_channels(); } function in class:webrtc::IFChannelBuffer
|
| audio_util.cc | 45 int num_channels,
48 num_channels, deinterleaved);
|
| /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_RecordingAudioTest/ |
| brillo_RecordingAudioTest.py | 36 def _get_recording_cmd(self, recording_method, duration_secs, num_channels,
43 @param num_channels: Number of channels to use for recording.
54 '--duration_secs=%d --num_channels=%d --filename=%s' %
55 (duration_secs, num_channels, rec_file) )
58 '--duration_secs=%d --num_channels=%d --filename=%s' %
59 (duration_secs, num_channels, rec_file))
68 sample_rate, num_channels):
76 @param num_channels: Number of channels to use for recording.
85 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/msm8x84/original-kernel-headers/linux/ |
| msm_audio_amrwbplus.h | 12 unsigned int num_channels; member in struct:msm_audio_amrwbplus_config_v2
|
| /hardware/broadcom/wlan/bcmdhd/wifi_hal/ |
| link_layer_stats.cpp | 41 uint32_t num_channels; member in struct:__anon29740
92 if (radio_stat->num_channels > 11) {
93 ALOGE("Incorrect number of channels = %d", radio_stat->num_channels);
94 // dump data before num_channels
109 + radio_stat->num_channels * sizeof(wifi_channel_stat));
119 uint32_t channel_size = internal_stat_ptr->num_channels * sizeof(wifi_channel_stat);
135 external_stat_ptr->num_channels = internal_stat_ptr->num_channels;
136 if (internal_stat_ptr->num_channels) {
|
| /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/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/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;
|
| /external/autotest/server/site_tests/brillo_PlaybackAudioTest/ |
| brillo_PlaybackAudioTest.py | 65 duration_secs, num_channels, play_file_path=None):
73 @param num_channels: Number of channels to test playback with.
78 num_channels=num_channels)
92 num_channels=_DEFAULT_NUM_CHANNELS,
104 @param num_channels: Number of channels to test playback with.
115 num_channels=num_channels,
129 'sine %d vol 0.9' % (num_channels, sine_format,
150 num_channels=num_channels
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| test_utils.cc | 40 RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
48 Deinterleave(&interleaved_[0], buffer->num_frames(), buffer->num_channels(),
57 RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
59 Interleave(buffer.channels(), buffer.num_frames(), buffer.num_channels(),
79 size_t num_channels,
82 size_t length = num_channels * samples_per_channel;
84 Interleave(data, samples_per_channel, num_channels, buffer.get());
119 AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels) {
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/beamformer/ |
| nonlinear_beamformer_test.cc | 48 const size_t num_mics = in_file.num_channels();
57 FLAGS_i.c_str(), in_file.num_channels(), in_file.sample_rate());
59 FLAGS_o.c_str(), out_file.num_channels(), out_file.sample_rate());
63 in_file.num_channels());
66 out_file.num_channels());
73 in_buf.num_channels(), in_buf.channels());
78 out_buf.num_channels(), &interleaved[0]);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| audio_decoder_opus.cc | 17 AudioDecoderOpus::AudioDecoderOpus(size_t num_channels)
18 : channels_(num_channels) {
19 RTC_DCHECK(num_channels == 1 || num_channels == 2);
|
