| /external/webrtc/webrtc/common_audio/resampler/ |
| D | sinc_resampler_unittest.cc | 60 SincResampler resampler(kSampleRateRatio, SincResampler::kDefaultRequestSize, in TEST() local
64 size_t max_chunk_size = resampler.ChunkSize() * kChunks; in TEST()
70 resampler.Resample(resampler.ChunkSize(), resampled_destination.get()); in TEST()
76 resampler.Resample(max_chunk_size, resampled_destination.get()); in TEST()
82 SincResampler resampler(kSampleRateRatio, SincResampler::kDefaultRequestSize, in TEST() local
85 new float[resampler.ChunkSize()]); in TEST()
90 resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get()); in TEST()
94 resampler.Flush(); in TEST()
98 resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get()); in TEST()
99 for (size_t i = 0; i < resampler.ChunkSize() / 2; ++i) in TEST()
[all …]
|
| D | push_resampler_unittest.cc | 19 PushResampler<int16_t> resampler; in TEST() local
20 EXPECT_EQ(-1, resampler.InitializeIfNeeded(-1, 16000, 1)); in TEST()
21 EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, -1, 1)); in TEST()
22 EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, 16000, 0)); in TEST()
23 EXPECT_EQ(-1, resampler.InitializeIfNeeded(16000, 16000, 3)); in TEST()
24 EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 1)); in TEST()
25 EXPECT_EQ(0, resampler.InitializeIfNeeded(16000, 16000, 2)); in TEST()
|
| D | push_sinc_resampler_unittest.cc | 97 PushSincResampler resampler(input_samples, output_samples); in ResampleBenchmarkTest() local
102 resampler.Resample(source_int.get(), in ResampleBenchmarkTest()
110 resampler.Resample(source.get(), in ResampleBenchmarkTest()
152 PushSincResampler resampler(input_block_size, output_block_size); in ResampleTest() local
169 resampler.get_resampler_for_testing()->ChunkSize(); in ResampleTest()
180 resampler.Resample(source_int.get(), in ResampleTest()
191 resampler.Resample(&source[i * input_block_size], in ResampleTest()
|
| D | Android.mk | 23 resampler.cc \
|
| /external/webrtc/webrtc/common_audio/ |
| D | common_audio.gyp | 21 'resampler/include',
26 'resampler/include',
54 'resampler/include/push_resampler.h',
55 'resampler/include/resampler.h',
56 'resampler/push_resampler.cc',
57 'resampler/push_sinc_resampler.cc',
58 'resampler/push_sinc_resampler.h',
59 'resampler/resampler.cc',
60 'resampler/sinc_resampler.cc',
61 'resampler/sinc_resampler.h',
[all …]
|
| D | BUILD.gn | 14 "resampler/include",
44 "resampler/include/push_resampler.h",
45 "resampler/include/resampler.h",
46 "resampler/push_resampler.cc",
47 "resampler/push_sinc_resampler.cc",
48 "resampler/push_sinc_resampler.h",
49 "resampler/resampler.cc",
50 "resampler/sinc_resampler.cc",
51 "resampler/sinc_resampler.h",
197 "resampler/sinc_resampler_sse.cc",
[all …]
|
| /external/webrtc/webrtc/voice_engine/ |
| D | utility.cc | 25 PushResampler<int16_t>* resampler, in RemixAndResample() argument
29 resampler, dst_frame); in RemixAndResample()
39 PushResampler<int16_t>* resampler, in RemixAndResample() argument
53 if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_, in RemixAndResample()
63 int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_, in RemixAndResample()
|
| D | utility.h | 35 PushResampler<int16_t>* resampler,
45 PushResampler<int16_t>* resampler,
|
| D | utility_unittest.cc | 130 PushResampler<int16_t> resampler; // Create a new one with every test. in RunResampleTest() local
166 RemixAndResample(src_frame_, &resampler, &dst_frame_); in RunResampleTest()
|
| /external/webrtc/webrtc/modules/audio_processing/aec/ |
| D | echo_cancellation.c | 133 aecpc->resampler = WebRtcAec_CreateResampler(); in WebRtcAec_Create()
134 if (!aecpc->resampler) { in WebRtcAec_Create()
182 WebRtcAec_FreeResampler(aecpc->resampler); in WebRtcAec_Free()
208 if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) { in WebRtcAec_Init()
304 WebRtcAec_ResampleLinear(aecpc->resampler, in WebRtcAec_BufferFarend()
596 retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew); in ProcessNormal()
|
| D | echo_cancellation_internal.h | 51 void* resampler; member
|
| /external/webrtc/ |
| D | Android.mk | 113 include $(webrtc_path)/webrtc/common_audio/resampler/Android.mk
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| D | Android.mk | 148 ../../common_audio/resampler/push_resampler.cc \
|
| /external/libopus/ |
| D | silk_sources.mk | 63 silk/resampler.c \
|
| D | Android.bp | 100 "silk/resampler.c",
|
| /external/webrtc/webrtc/modules/audio_processing/test/ |
| D | audio_processing_unittest.cc | 2574 PushResampler<float> resampler; in TEST_P() local
2575 resampler.InitializeIfNeeded(out_rate, ref_rate, out_num); in TEST_P()
2606 static_cast<size_t>(resampler.Resample( in TEST_P()
|