| /external/webrtc/common_audio/resampler/ |
| D | push_sinc_resampler.cc | 22 : resampler_(new SincResampler(source_frames * 1.0 / destination_frames, in PushSincResampler()
52 RTC_CHECK_EQ(source_length, resampler_->request_frames()); in Resample()
73 resampler_->Resample(resampler_->ChunkSize(), destination); in Resample()
75 resampler_->Resample(destination_frames_, destination); in Resample()
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| D | push_sinc_resampler.h | 62 SincResampler* get_resampler_for_testing() { return resampler_.get(); } in get_resampler_for_testing()
64 std::unique_ptr<SincResampler> resampler_; variable
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| /external/openscreen/cast/standalone_sender/ |
| D | simulated_capturer.cc | 235 resampler_(MakeUniqueSwrContext()) { in SimulatedAudioCapturer()
241 if (swr_is_initialized(resampler_.get())) { in ~SimulatedAudioCapturer()
242 swr_close(resampler_.get()); in ~SimulatedAudioCapturer()
248 if (swr_is_initialized(resampler_.get())) { in EnsureResamplerIsInitializedFor()
261 swr_get_delay(resampler_.get(), std::micro::den)); in EnsureResamplerIsInitializedFor()
276 swr_config_frame(resampler_.get(), fake_output_frame.get(), &frame); in EnsureResamplerIsInitializedFor()
282 const int init_result = swr_init(resampler_.get()); in EnsureResamplerIsInitializedFor()
301 swr_get_delay(resampler_.get(), input_sample_rate_); in ProcessDecodedFrame()
314 resampler_.get(), output_argument, num_output_samples_desired, in ProcessDecodedFrame()
318 swr_close(resampler_.get()); in ProcessDecodedFrame()
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| D | simulated_capturer.h | 158 const SwrContextUniquePtr resampler_; variable
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| /external/webrtc/modules/audio_coding/acm2/ |
| D | acm_resampler.cc | 41 if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz, in Resample10Msec()
50 resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples); in Resample10Msec()
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| D | acm_resampler.h | 35 PushResampler<int16_t> resampler_;
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| D | acm_receiver.h | 217 ACMResampler resampler_ RTC_GUARDED_BY(mutex_);
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| D | acm_receiver.cc | 166 int samples_per_channel_int = resampler_.Resample10Msec( in GetAudio()
181 int samples_per_channel_int = resampler_.Resample10Msec( in GetAudio()
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| D | audio_coding_module.cc | 163 acm2::ACMResampler resampler_ RTC_GUARDED_BY(acm_mutex_);
496 int samples_per_channel = resampler_.Resample10Msec( in PreprocessToAddData()
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| /external/webrtc/audio/ |
| D | remix_resample_unittest.cc | 40 PushResampler<int16_t> resampler_; member in webrtc::voe::__anon2b0da77f0111::UtilityTest
228 RemixAndResample(src_frame_, &resampler_, &dst_frame_); in TEST_F()
234 RemixAndResample(src_frame_, &resampler_, &dst_frame_); in TEST_F()
243 RemixAndResample(src_frame_, &resampler_, &dst_frame_); in TEST_F()
250 RemixAndResample(src_frame_, &resampler_, &dst_frame_); in TEST_F()
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| /external/webrtc/modules/audio_coding/neteq/tools/ |
| D | resample_input_audio_file.cc | 29 resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1); in Read()
31 RTC_CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read, in Read()
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| D | resample_input_audio_file.h | 47 Resampler resampler_; variable
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| /external/webrtc/modules/audio_processing/agc2/ |
| D | vad_with_level.cc | 49 resampler_.Resample(frame.channel(0).data(), frame.samples_per_channel(), in AnalyzeFrame()
66 resampler_.InitializeIfNeeded(sample_rate_hz, rnn_vad::kSampleRate24kHz, in SetSampleRate()
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| D | vad_with_level.h | 43 PushResampler<float> resampler_; variable
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| /external/webrtc/modules/audio_processing/vad/ |
| D | voice_activity_detector.cc | 45 resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels), in ProcessChunk()
47 resampler_.Push(audio, length, resampled_, kLength10Ms, length); in ProcessChunk()
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| D | voice_activity_detector.h | 60 Resampler resampler_; variable
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| /external/webrtc/modules/audio_coding/test/ |
| D | opus_test.h | 50 acm2::ACMResampler resampler_; variable
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| D | opus_test.cc | 258 EXPECT_EQ(480, resampler_.Resample10Msec( in Run()
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