| /external/webrtc/webrtc/common_audio/resampler/ |
| push_sinc_resampler.cc | 22 : resampler_(new SincResampler(source_frames * 1.0 / destination_frames,
53 RTC_CHECK_EQ(source_length, resampler_->request_frames());
74 resampler_->Resample(resampler_->ChunkSize(), destination);
76 resampler_->Resample(destination_frames_, destination);
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| push_sinc_resampler.h | 57 SincResampler* get_resampler_for_testing() { return resampler_.get(); }
59 rtc::scoped_ptr<SincResampler> resampler_; member in class:webrtc::PushSincResampler
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| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| acm_resampler.h | 33 PushResampler<int16_t> resampler_; member in class:webrtc::acm2::ACMResampler
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| acm_resampler.cc | 44 if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz,
52 resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
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| acm_receiver.h | 288 ACMResampler resampler_ GUARDED_BY(crit_sect_);
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| audio_coding_module_impl.h | 248 ACMResampler resampler_ GUARDED_BY(acm_crit_sect_);
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| acm_receiver.cc | 240 int samples_per_channel_int = resampler_.Resample10Msec(
256 int samples_per_channel_int = resampler_.Resample10Msec(
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| audio_coding_module_impl.cc | 421 int samples_per_channel = resampler_.Resample10Msec(
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| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| resample_input_audio_file.cc | 28 resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1);
30 RTC_CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read,
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| resample_input_audio_file.h | 45 Resampler resampler_; member in class:webrtc::test::ResampleInputAudioFile
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| /external/webrtc/webrtc/modules/audio_coding/test/ |
| opus_test.h | 51 acm2::ACMResampler resampler_; member in class:webrtc::OpusTest
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| opus_test.cc | 261 resampler_.Resample10Msec(audio_frame.data_,
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| /external/webrtc/webrtc/modules/audio_processing/vad/ |
| voice_activity_detector.h | 58 Resampler resampler_; member in class:webrtc::VoiceActivityDetector
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| voice_activity_detector.cc | 46 resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels),
48 resampler_.Push(audio, length, resampled_, kLength10Ms, length);
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| /external/webrtc/webrtc/voice_engine/ |
| utility_unittest.cc | 39 PushResampler<int16_t> resampler_; member in class:webrtc::voe::__anon28214::UtilityTest
181 RemixAndResample(src_frame_, &resampler_, &dst_frame_);
187 RemixAndResample(src_frame_, &resampler_, &dst_frame_);
196 RemixAndResample(src_frame_, &resampler_, &dst_frame_);
203 RemixAndResample(src_frame_, &resampler_, &dst_frame_);
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| output_mixer.h | 118 PushResampler<int16_t> resampler_; member in class:webrtc::voe::OutputMixer
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| transmit_mixer.h | 201 PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate member in class:webrtc::voe::TransmitMixer
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| output_mixer.cc | 484 RemixAndResample(_audioFrame, &resampler_, frame);
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| transmit_mixer.cc | [all...] |
