| /external/mesa3d/src/gallium/auxiliary/vl/ |
| vl_zscan.c | 109 o_vtex = MALLOC(zscan->num_channels * sizeof(struct ureg_dst));
123 for (i = 0; i < zscan->num_channels; ++i)
148 for (i = 0; i < zscan->num_channels; ++i) {
151 * (i - (signed)zscan->num_channels / 2)));
186 vtex = MALLOC(zscan->num_channels * sizeof(struct ureg_src));
187 tmp = MALLOC(zscan->num_channels * sizeof(struct ureg_dst));
189 for (i = 0; i < zscan->num_channels; ++i)
196 for (i = 0; i < zscan->num_channels; ++i)
207 for (i = 0; i < zscan->num_channels; ++i)
210 for (i = 0; i < zscan->num_channels; ++i
[all...] |
| vl_zscan.h | 45 unsigned num_channels; member in struct:vl_zscan
78 unsigned num_channels);
|
| /external/webrtc/webrtc/common_audio/ |
| wav_header_unittest.cc | 94 size_t num_channels = 0; local
122 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
143 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
164 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
186 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
209 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
228 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
240 ReadWavHeader(&r, &num_channels, &sample_rate, &format,
271 size_t num_channels = 0; local
278 ReadWavHeader(&r, &num_channels, &sample_rate, &format
307 size_t num_channels = 0; local
[all...] |
| blocker_unittest.cc | 86 size_t num_channels,
88 for (size_t i = 0; i < num_channels; ++i) {
96 size_t num_channels,
99 for (size_t i = 0; i < num_channels; ++i) {
113 size_t num_channels,
116 for (size_t i = 0; i < num_channels; ++i) {
|
| /hardware/intel/common/libmix/mix_audio/src/ |
| mixacp.c | 48 self->num_channels = 0;
142 t->num_channels = s->num_channels;
195 (acp1->num_channels == acp2->num_channels) &&
310 mix_log(MIX_AUDIO_COMP, MIX_LOG_LEVEL_INFO, "num_channels: %d\n", obj->num_channels);
|
| /cts/tests/camera/src/android/hardware/camera2/cts/rs/ |
| BitmapUtils.java | 36 public static int NUM_CHANNELS = 4;
43 * @return an array containing NUM_CHANNELS * COLOR_BIT_DEPTH histogram bucket values, with
55 int[] output = new int[COLOR_BIT_DEPTH * NUM_CHANNELS];
|
| /external/webrtc/webrtc/modules/audio_processing/transient/ |
| transient_suppressor.h | 32 int Initialize(int sample_rate_hz, int detector_rate_hz, int num_channels);
38 // |num_channels| must be equal to |num_channels_|.
53 int num_channels,
|
| /external/autotest/site_utils/tester_feedback/ |
| audio_query_delegate_impl.py | 183 sample_rate=None, num_channels=None, peak_percent_min=1,
190 @param num_channels: The number of recorded channels.
199 captured_audio_file, num_channels=num_channels,
209 if num_channels is not None:
210 props.append('has %d recorded channels' % num_channels)
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/g711/ |
| audio_encoder_pcm.h | 28 size_t num_channels; member in struct:webrtc::AudioEncoderPcm::Config
33 : frame_size_ms(20), num_channels(1), payload_type(pt) {}
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| audio_buffer.cc | 32 return stream_config.num_channels();
108 assert(stream_config.num_channels() == num_input_channels_);
153 assert(stream_config.num_channels() == num_channels_ || num_channels_ == 1);
178 for (size_t i = num_channels_; i < stream_config.num_channels(); ++i) {
344 size_t AudioBuffer::num_channels() const { function in class:webrtc::AudioBuffer
348 void AudioBuffer::set_num_channels(size_t num_channels) {
349 num_channels_ = num_channels;
446 low_pass_reference_channels_->num_channels() != num_channels_) {
|
| audio_processing_impl.cc | 349 ? formats_.api_format.input_stream().num_channels()
350 : formats_.api_format.output_stream().num_channels();
355 if (formats_.api_format.reverse_input_stream().num_channels() > 0) {
358 formats_.api_format.reverse_input_stream().num_channels(),
360 formats_.rev_proc_format.num_channels(),
364 formats_.api_format.reverse_input_stream().num_channels(),
366 formats_.api_format.reverse_output_stream().num_channels(),
377 formats_.api_format.input_stream().num_channels(),
413 if (stream.num_channels() > 0 && stream.sample_rate_hz() <= 0) {
418 const size_t num_in_channels = config.input_stream().num_channels();
[all...] |
| level_estimator_impl.cc | 38 for (size_t i = 0; i < audio->num_channels(); i++) {
|
| /hardware/qcom/wlan/qcwcn/wifi_hal/ |
| gscancommand.h | 40 u32 num_channels; member in struct:__anon34293
125 virtual void setNumChannelsPtr(int *num_channels);
|
| /external/webrtc/webrtc/modules/audio_coding/neteq/ |
| neteq_unittest.cc | 428 size_t num_channels; local
430 &num_channels, &type));
611 size_t num_channels; local
614 &num_channels, &type));
656 size_t num_channels; local
659 &num_channels, &type));
687 size_t num_channels; local
690 &num_channels, &type));
712 size_t num_channels; local
731 &num_channels, &type))
951 size_t num_channels; local
985 size_t num_channels; local
1275 size_t num_channels; local
1351 size_t num_channels; local
1418 size_t num_channels; local
1625 size_t num_channels; local
[all...] |
| /external/autotest/server/site_tests/brillo_RecordingAudioTest/ |
| control.libmedia.mono | 44 num_channels=1,
|
| control.libmedia.stereo | 44 num_channels=2,
|
| /external/kernel-headers/original/uapi/sound/ |
| tlv.h | 25 * TLV length must match with num_channels
|
| /external/webrtc/webrtc/common_audio/resampler/include/ |
| push_resampler.h | 32 size_t num_channels);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/opus/ |
| audio_decoder_opus.h | 21 explicit AudioDecoderOpus(size_t num_channels);
|
| /external/webrtc/webrtc/modules/audio_coding/codecs/pcm16b/ |
| audio_encoder_pcm16b.cc | 32 config.num_channels = codec_inst.channels;
|
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| audio_file_processor.cc | 32 return StreamConfig(file.sample_rate(), file.num_channels());
39 file.num_channels());
135 RTC_CHECK_EQ(in_buf_->num_channels(),
161 RTC_CHECK_EQ(reverse_buf_->num_channels(),
|
| /external/webrtc/webrtc/voice_engine/test/auto_test/ |
| voe_output_test.cc | 156 const int num_channels = isStereo ? 2 : 1; local
158 for (int c = 0; c < num_channels; ++c) {
159 ASSERT_GE(audio10ms[i * num_channels + c], lower_bound_);
160 ASSERT_LE(audio10ms[i * num_channels + c], upper_bound_);
|
| /frameworks/av/media/libstagefright/ |
| XINGSeeker.cpp | 95 int num_channels; local
97 if (!GetMPEGAudioFrameSize(header, &xingframesize, &sampling_rate, &num_channels,
107 if (num_channels != 1) offset += 32;
110 if (num_channels != 1) offset += 17;
|
| /hardware/qcom/msm8994/kernel-headers/linux/ |
| msm_audio_amrwbplus.h | 27 unsigned int num_channels; member in struct:msm_audio_amrwbplus_config_v2
|
| /hardware/qcom/msm8996/kernel-headers/linux/ |
| msm_audio_amrwbplus.h | 27 unsigned int num_channels; member in struct:msm_audio_amrwbplus_config_v2
|
