| /external/webrtc/webrtc/common_audio/resampler/ |
| D | push_sinc_resampler.cc | 22 : resampler_(new SincResampler(source_frames * 1.0 / destination_frames, in PushSincResampler()
53 RTC_CHECK_EQ(source_length, resampler_->request_frames()); in Resample()
74 resampler_->Resample(resampler_->ChunkSize(), destination); in Resample()
76 resampler_->Resample(destination_frames_, destination); in Resample()
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| D | push_sinc_resampler.h | 57 SincResampler* get_resampler_for_testing() { return resampler_.get(); } in get_resampler_for_testing()
59 rtc::scoped_ptr<SincResampler> resampler_; variable
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| /external/webrtc/webrtc/voice_engine/ |
| D | utility_unittest.cc | 39 PushResampler<int16_t> resampler_; member in webrtc::voe::__anon974ed41d0111::UtilityTest
181 RemixAndResample(src_frame_, &resampler_, &dst_frame_); in TEST_F()
187 RemixAndResample(src_frame_, &resampler_, &dst_frame_); in TEST_F()
196 RemixAndResample(src_frame_, &resampler_, &dst_frame_); in TEST_F()
203 RemixAndResample(src_frame_, &resampler_, &dst_frame_); in TEST_F()
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| D | output_mixer.h | 118 PushResampler<int16_t> resampler_; variable
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| D | transmit_mixer.h | 201 PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate variable
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| D | output_mixer.cc | 484 RemixAndResample(_audioFrame, &resampler_, frame); in GetMixedAudio()
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| D | transmit_mixer.cc | 1159 &resampler_, &_audioFrame); in GenerateAudioFrame()
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| /external/webrtc/webrtc/modules/audio_coding/acm2/ |
| D | acm_resampler.cc | 44 if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz, in Resample10Msec()
52 resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples); in Resample10Msec()
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| D | acm_resampler.h | 33 PushResampler<int16_t> resampler_;
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| D | acm_receiver.h | 288 ACMResampler resampler_ GUARDED_BY(crit_sect_);
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| D | acm_receiver.cc | 240 int samples_per_channel_int = resampler_.Resample10Msec( in GetAudio()
256 int samples_per_channel_int = resampler_.Resample10Msec( in GetAudio()
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| D | audio_coding_module_impl.h | 248 ACMResampler resampler_ GUARDED_BY(acm_crit_sect_);
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| D | audio_coding_module_impl.cc | 421 int samples_per_channel = resampler_.Resample10Msec( in PreprocessToAddData()
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| /external/webrtc/webrtc/modules/audio_coding/neteq/tools/ |
| D | resample_input_audio_file.cc | 28 resampler_.ResetIfNeeded(file_rate_hz_, output_rate_hz, 1); in Read()
30 RTC_CHECK_EQ(resampler_.Push(temp_destination.get(), samples_to_read, in Read()
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| D | resample_input_audio_file.h | 45 Resampler resampler_; variable
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| /external/webrtc/webrtc/modules/audio_processing/vad/ |
| D | voice_activity_detector.cc | 46 resampler_.ResetIfNeeded(sample_rate_hz, kSampleRateHz, kNumChannels), in ProcessChunk()
48 resampler_.Push(audio, length, resampled_, kLength10Ms, length); in ProcessChunk()
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| D | voice_activity_detector.h | 58 Resampler resampler_; variable
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| /external/webrtc/webrtc/modules/audio_coding/test/ |
| D | opus_test.h | 51 acm2::ACMResampler resampler_; variable
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| D | opus_test.cc | 261 resampler_.Resample10Msec(audio_frame.data_, in Run()
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