| /external/webrtc/webrtc/common_audio/ |
| audio_converter.cc | 28 CopyConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
30 : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
37 for (size_t i = 0; i < src_channels(); ++i)
45 UpmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
47 : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {}
63 DownmixConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
65 : AudioConverter(src_channels, src_frames, dst_channels, dst_frames) {
75 for (size_t j = 0; j < src_channels(); ++j)
77 dst_mono[i] = sum / src_channels();
84 ResampleConverter(size_t src_channels, size_t src_frames, size_t dst_channels
[all...] |
| audio_converter.h | 21 // upmix from mono (i.e. |src_channels == 1|).
29 static rtc::scoped_ptr<AudioConverter> Create(size_t src_channels,
42 size_t src_channels() const { return src_channels_; } function in class:webrtc::AudioConverter
49 AudioConverter(size_t src_channels, size_t src_frames, size_t dst_channels,
|
| audio_converter_unittest.cc | 89 void RunAudioConverterTest(size_t src_channels,
104 if (src_channels == 2)
111 if (src_channels == 1)
118 if (src_channels == 1)
133 src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
136 src_channels, src_frames, dst_channels, dst_frames);
|
| /external/webrtc/webrtc/voice_engine/ |
| utility_unittest.cc | 34 void RunResampleTest(int src_channels,
126 void UtilityTest::RunResampleTest(int src_channels,
138 if (src_channels == 1)
145 if (src_channels == 1)
151 if (src_channels == 1)
165 src_channels, src_sample_rate_hz, dst_channels, dst_sample_rate_hz);
|
| /external/mesa3d/src/gallium/drivers/llvmpipe/ |
| lp_state_fs.c | 679 unsigned src_channels; local
683 src_channels = dst_channels < 3 ? dst_channels : 4;
684 src_count = num_fs * src_channels;
687 assert(num_fs * src_channels <= ARRAY_SIZE(src));
693 lp_build_transpose_aos_n(gallivm, type, &fs_src[i][0], src_channels, &src[i * src_channels]);
1756 unsigned src_channels = TGSI_NUM_CHANNELS; local
[all...] |
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| audio_processing_unittest.cc | 1571 auto src_channels = &src[0]; local
[all...] |
