| /external/webrtc/webrtc/voice_engine/ |
| utility.h | 39 // |samples_per_channel|, |num_channels| and |sample_rate_hz| of the data as
43 size_t num_channels,
|
| utility.cc | 37 size_t num_channels,
42 size_t audio_ptr_num_channels = num_channels;
46 if (num_channels == 2 && dst_frame->num_channels_ == 1) {
74 if (num_channels == 1 && dst_frame->num_channels_ == 2) {
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| /external/webrtc/webrtc/common_audio/ |
| wav_file.h | 30 virtual size_t num_channels() const = 0;
42 WavWriter(const std::string& filename, int sample_rate, size_t num_channels);
54 size_t num_channels() const override { return num_channels_; }
82 size_t num_channels() const override { return num_channels_; }
105 size_t num_channels);
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| blocker.cc | 25 size_t num_channels,
28 for (size_t i = 0; i < num_channels; ++i) {
40 size_t num_channels,
43 for (size_t i = 0; i < num_channels; ++i) {
54 size_t num_channels,
57 for (size_t i = 0; i < num_channels; ++i) {
67 size_t num_channels) {
68 for (size_t i = 0; i < num_channels; ++i) {
78 size_t num_channels,
80 for (size_t i = 0; i < num_channels; ++i)
[all...] |
| audio_converter_unittest.cc | 29 const size_t num_channels = data.size(); local
30 ScopedBuffer sb(new ChannelBuffer<float>(frames, num_channels));
31 for (size_t i = 0; i < num_channels; ++i)
39 EXPECT_EQ(ref.num_channels(), test.num_channels());
60 for (size_t i = 0; i < ref.num_channels(); ++i) {
69 const size_t length = ref.num_channels() * (ref.num_frames() - delay);
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| wav_file.cc | 43 s << "Sample rate: " << sample_rate() << " Hz, Channels: " << num_channels()
45 << (1.f * num_samples()) / (num_channels() * sample_rate()) << " s";
102 size_t num_channels)
104 num_channels_(num_channels),
156 size_t num_channels) {
158 new webrtc::WavWriter(filename, sample_rate, num_channels));
176 return reinterpret_cast<const webrtc::WavWriter*>(wf)->num_channels();
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| /external/webrtc/webrtc/modules/audio_processing/test/ |
| test_utils.h | 82 size_t num_channels,
96 size_t num_channels,
100 frame->num_channels_ = num_channels;
101 cb->reset(new ChannelBuffer<T>(frame->samples_per_channel_, num_channels));
104 AudioProcessing::ChannelLayout LayoutFromChannels(size_t num_channels);
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| /device/generic/goldfish/camera/fake-pipeline2/ |
| Scene.cpp | 31 #define G (Scene::GRASS * Scene::NUM_CHANNELS)
32 #define S (Scene::GRASS_SHADOW * Scene::NUM_CHANNELS)
33 #define H (Scene::HILL * Scene::NUM_CHANNELS)
34 #define W (Scene::WALL * Scene::NUM_CHANNELS)
35 #define R (Scene::ROOF * Scene::NUM_CHANNELS)
36 #define D (Scene::DOOR * Scene::NUM_CHANNELS)
37 #define C (Scene::CHIMNEY * Scene::NUM_CHANNELS)
38 #define I (Scene::WINDOW * Scene::NUM_CHANNELS)
39 #define U (Scene::SUN * Scene::NUM_CHANNELS)
40 #define K (Scene::SKY * Scene::NUM_CHANNELS)
[all...] |
| /system/extras/sound/ |
| playwav.c | 114 uint16_t num_channels; member in struct:wav_header
116 uint32_t byte_rate; /* sample_rate * num_channels * bps / 8 */
117 uint16_t block_align; /* num_channels * bps / 8 */
168 hdr.num_channels, hdr.sample_rate, hdr.bits_per_sample,
187 play_file(hdr.sample_rate, hdr.num_channels,
209 hdr.num_channels = channels;
211 hdr.byte_rate = hdr.sample_rate * hdr.num_channels * 2;
212 hdr.block_align = hdr.num_channels * 2;
237 cfg.channel_count = hdr.num_channels;
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| /external/webrtc/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) {
76 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),
82 size_t num_channels,
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| neteq_stereo_unittest.cc | 30 size_t num_channels; member in struct:webrtc::TestParameters
50 : num_channels_(GetParam().num_channels),
217 size_t num_channels; local
220 &samples_per_channel, &num_channels,
222 EXPECT_EQ(1u, num_channels);
228 &samples_per_channel, &num_channels,
230 EXPECT_EQ(num_channels_, num_channels);
389 p.num_channels = 2;
393 p.num_channels = 5;
404 ", num_channels = " << p.num_channels <
[all...] |
| neteq_impl_unittest.cc | 469 size_t num_channels; local
474 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
476 EXPECT_EQ(1u, num_channels);
548 size_t num_channels; local
553 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
555 EXPECT_EQ(1u, num_channels);
584 kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
586 EXPECT_EQ(1u, num_channels);
625 size_t num_channels; local
629 &num_channels, &type))
737 size_t num_channels; local
877 size_t num_channels; local
983 size_t num_channels; local
1080 size_t num_channels; local
1201 size_t num_channels; local
[all...] |
| /external/autotest/server/brillo/feedback/ |
| closed_loop_audio_client.py | 231 num_channels=_DEFAULT_NUM_CHANNELS,
237 @num_channels: Number of channels to record at.
240 self.num_channels = num_channels
250 (num_channels, duration_secs, sample_rate, sample_width,
271 num_channels=self.num_channels,
306 num_channels=self.num_channels,
377 num_channels = golden_file.getnchannels(
[all...] |
| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| audio_encoder_pcm.cc | 27 config.num_channels = codec_inst.channels;
35 return (frame_size_ms % 10 == 0) && (num_channels >= 1);
40 num_channels_(config.num_channels),
45 config.num_channels * config.frame_size_ms * sample_rate_hz / 1000),
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
| audio_decoder_pcm16b.h | 21 explicit AudioDecoderPcm16B(size_t num_channels);
|
| /hardware/qcom/msm8960/original-kernel-headers/linux/mfd/ |
| msm-adie-codec.h | 116 u32 num_channels,
120 u32 num_channels,
140 u32 num_channels, u32 vol_percentage /* in percentage */);
143 u32 num_channels, u32 volume /* in percentage */);
|
| /hardware/qcom/msm8994/original-kernel-headers/linux/mfd/ |
| msm-adie-codec.h | 116 u32 num_channels,
120 u32 num_channels,
140 u32 num_channels, u32 vol_percentage /* in percentage */);
143 u32 num_channels, u32 volume /* in percentage */);
|
| /hardware/qcom/msm8996/original-kernel-headers/linux/mfd/ |
| msm-adie-codec.h | 116 u32 num_channels,
120 u32 num_channels,
140 u32 num_channels, u32 vol_percentage /* in percentage */);
143 u32 num_channels, u32 volume /* in percentage */);
|
| /hardware/qcom/msm8x26/original-kernel-headers/linux/mfd/ |
| msm-adie-codec.h | 116 u32 num_channels,
120 u32 num_channels,
140 u32 num_channels, u32 vol_percentage /* in percentage */);
143 u32 num_channels, u32 volume /* in percentage */);
|
| /hardware/qcom/msm8x84/original-kernel-headers/linux/mfd/ |
| msm-adie-codec.h | 116 u32 num_channels,
120 u32 num_channels,
140 u32 num_channels, u32 vol_percentage /* in percentage */);
143 u32 num_channels, u32 volume /* in percentage */);
|
| /external/drm_gralloc/ |
| gralloc_drm_radeon.c | 62 int num_channels; member in struct:radeon_info
128 height_align = info->num_channels * 8;
154 base_align = MAX(info->num_banks * info->num_channels * 8 * 8 * bpe,
350 info->num_channels = 1;
353 info->num_channels = 2;
356 info->num_channels = 4;
359 info->num_channels = 8;
396 info->num_channels = 1;
399 info->num_channels = 2;
402 info->num_channels = 4
[all...] |
| /external/opencv3/3rdparty/libwebp/utils/ |
| rescaler.c | 29 int dst_stride, int num_channels, int x_add, int x_sub,
38 wrk->num_channels = num_channels;
51 wrk->frow = work + num_channels * dst_width;
56 const int x_stride = wrk->num_channels;
57 const int x_out_max = wrk->dst_width * wrk->num_channels;
104 const int x_out_max = wrk->dst_width * wrk->num_channels;
131 for (channel = 0; channel < wrk->num_channels; ++channel) {
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| audio_encoder_opus.cc | 29 config.num_channels = codec_inst.channels;
32 config.application = config.num_channels == 1 ? AudioEncoderOpus::kVoip
82 if (num_channels != 1 && num_channels != 2)
118 return config_.num_channels;
150 rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
220 return rtc::CheckedDivExact(kSampleRateHz, 100) * config_.num_channels;
233 RTC_CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, config.num_channels,
|
| /external/webp/src/utils/ |
| rescaler.c | 25 int num_channels, rescaler_t* const work) {
38 wrk->num_channels = num_channels;
67 wrk->frow = work + num_channels * dst_width;
68 memset(work, 0, 2 * dst_width * num_channels * sizeof(*work));
121 for (x = 0; x < wrk->num_channels * wrk->dst_width; ++x) {
|
