/external/chromium_org/third_party/webrtc/common_audio/resampler/ |
H A D | push_sinc_resampler.cc | 21 : resampler_(new SincResampler(source_frames * 1.0 / destination_frames, 52 assert(source_length == resampler_->request_frames()); 73 resampler_->Resample(resampler_->ChunkSize(), destination); 75 resampler_->Resample(destination_frames_, destination);
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H A D | push_sinc_resampler.h | 46 SincResampler* get_resampler_for_testing() { return resampler_.get(); } 52 scoped_ptr<SincResampler> resampler_; member in class:webrtc::PushSincResampler
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/acm2/ |
H A D | acm_resampler.h | 33 PushResampler<int16_t> resampler_; member in class:webrtc::acm2::ACMResampler
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H A D | acm_resampler.cc | 45 if (resampler_.InitializeIfNeeded(in_freq_hz, out_freq_hz, 53 resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
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H A D | acm_receiver.h | 334 ACMResampler resampler_ GUARDED_BY(crit_sect_);
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H A D | acm_receiver.cc | 416 resampler_.Resample10Msec(audio_buffer_, 436 resampler_.Resample10Msec(audio_frame->data_,
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H A D | audio_coding_module_impl.h | 341 ACMResampler resampler_ GUARDED_BY(acm_crit_sect_);
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H A D | audio_coding_module_impl.cc | 1387 resampler_.Resample10Msec(src_ptr_audio,
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/external/chromium_org/media/base/ |
H A D | audio_converter.cc | 56 resampler_.reset(new MultiChannelResampler( 72 if (disable_fifo || resampler_) 109 if (resampler_) 110 resampler_->Flush(); 114 if (!resampler_) 116 return resampler_->ChunkSize(); 143 if (!resampler_ && !audio_fifo_) { 146 if (resampler_) 147 resampler_->Resample(temp_dest->frames(), temp_dest); 186 if (resampler_) { [all...] |
H A D | audio_converter.h | 117 scoped_ptr<MultiChannelResampler> resampler_; member in class:media::AudioConverter
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/external/chromium_org/third_party/webrtc/modules/audio_coding/main/test/ |
H A D | opus_test.h | 48 acm2::ACMResampler resampler_; member in class:webrtc::OpusTest
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H A D | opus_test.cc | 259 resampler_.Resample10Msec(audio_frame.data_,
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/external/chromium_org/media/audio/ |
H A D | audio_output_proxy_unittest.cc | 428 resampler_ = new AudioOutputResampler( 444 scoped_refptr<AudioOutputResampler> resampler_; member in class:media::AudioOutputResamplerTest 453 AudioOutputProxy* proxy = new AudioOutputProxy(resampler_.get()); 462 OpenAndClose(resampler_.get()); 474 CreateAndWait(resampler_.get()); 482 StartAndStop(resampler_.get()); 490 CloseAfterStop(resampler_.get()); 498 TwoStreams(resampler_.get()); 508 OneStream_TwoPlays(resampler_.get()); 517 TwoStreams_BothPlaying(resampler_ [all...] |
/external/chromium_org/remoting/codec/ |
H A D | audio_encoder_opus.h | 45 scoped_ptr<media::MultiChannelResampler> resampler_; member in class:remoting::AudioEncoderOpus
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H A D | audio_encoder_opus.cc | 76 resampler_.reset(new media::MultiChannelResampler( 99 resampler_.reset(); 159 resampler_.get() ? media::SincResampler::kDefaultRequestSize : 0; 177 if (resampler_.get()) { 181 resampler_->Resample(kFrameSamples, resampler_bus_.get());
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/external/chromium_org/third_party/webrtc/voice_engine/ |
H A D | utility_unittest.cc | 42 PushResampler<int16_t> resampler_; member in class:webrtc::voe::__anon16259::UtilityTest 197 RemixAndResample(src_frame_, &resampler_, &dst_frame_); 203 RemixAndResample(src_frame_, &resampler_, &dst_frame_); 212 RemixAndResample(src_frame_, &resampler_, &dst_frame_); 219 RemixAndResample(src_frame_, &resampler_, &dst_frame_);
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H A D | output_mixer.h | 133 PushResampler<int16_t> resampler_; member in class:webrtc::voe::OutputMixer
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H A D | transmit_mixer.h | 201 PushResampler<int16_t> resampler_; // ADM sample rate -> mixing rate member in class:webrtc::voe::TransmitMixer
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H A D | output_mixer.cc | 507 RemixAndResample(_audioFrame, &resampler_, frame);
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H A D | transmit_mixer.cc | 1167 &resampler_,
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/external/chromium_org/chrome/renderer/media/ |
H A D | cast_rtp_stream.cc | 364 if (resampler_) { 414 resampler_->Resample(output_bus->frames(), output_bus.get()); 427 resampler_.reset(new media::MultiChannelResampler( 463 scoped_ptr<media::MultiChannelResampler> resampler_; member in class:CastAudioSink
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