| /external/webrtc/common_audio/resampler/ |
| D | push_sinc_resampler.cc | 33 size_t PushSincResampler::Resample(const int16_t* source, in Resample() function in webrtc::PushSincResampler
42 Resample(nullptr, source_length, float_buffer_.get(), destination_frames_); in Resample()
48 size_t PushSincResampler::Resample(const float* source, in Resample() function in webrtc::PushSincResampler
73 resampler_->Resample(resampler_->ChunkSize(), destination); in Resample()
75 resampler_->Resample(destination_frames_, destination); in Resample()
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| D | sinc_resampler_unittest.cc | 72 resampler.Resample(resampler.ChunkSize(), resampled_destination.get()); in TEST()
79 resampler.Resample(max_chunk_size, resampled_destination.get()); in TEST()
92 resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get()); in TEST()
99 resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get()); in TEST()
242 TEST_P(SincResamplerTest, Resample) { in TEST_P() argument
279 resampler.Resample(output_samples, resampled_destination.get()); in TEST_P()
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| D | push_sinc_resampler_unittest.cc | 91 sinc_resampler.Resample(output_samples, resampled_destination.get()); in ResampleBenchmarkTest()
103 resampler.Resample(source_int.get(), input_samples, in ResampleBenchmarkTest()
108 EXPECT_EQ(output_samples, resampler.Resample(source.get(), input_samples, in ResampleBenchmarkTest()
180 resampler.Resample(source_int.get(), input_block_size, in ResampleTest()
189 resampler.Resample(&source[i * input_block_size], input_block_size, in ResampleTest()
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| D | push_sinc_resampler.h | 43 size_t Resample(const int16_t* source,
47 size_t Resample(const float* source,
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| D | push_resampler.cc | 76 int PushResampler<T>::Resample(const T* src, in Resample() function in webrtc::PushResampler
106 dst_length_mono = resampler.resampler->Resample( in Resample()
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| D | sinc_resampler.h | 70 void Resample(size_t frames, float* destination);
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| D | sinc_resampler.cc | 257 void SincResampler::Resample(size_t frames, float* destination) { in Resample() function in webrtc::SincResampler
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| /external/webrtc/modules/audio_processing/ |
| D | audio_buffer.cc | 131 input_resamplers_[0]->Resample(downmixed_data, input_num_frames_, in CopyFrom()
140 input_resamplers_[i]->Resample(stacked_data[i], input_num_frames_, in CopyFrom()
164 output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_, in CopyTo()
186 output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_, in CopyTo()
235 input_resamplers_[0]->Resample(float_buffer.data(), input_num_frames_, in CopyFrom()
261 input_resamplers_[0]->Resample(downmixed_data, input_num_frames_, in CopyFrom()
281 input_resamplers_[i]->Resample(float_buffer.data(), input_num_frames_, in CopyFrom()
308 output_resamplers_[0]->Resample(data_->channels()[0], buffer_num_frames_, in CopyTo()
339 output_resamplers_[i]->Resample(data_->channels()[i], in CopyTo()
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| /external/webrtc/audio/ |
| D | audio_transport_impl.cc | 70 int Resample(const AudioFrame& frame, in Resample() function
84 return resampler->Resample( in Resample()
232 nSamplesOut = Resample(mixed_frame_, samplesPerSec, &render_resampler_, in NeedMorePlayData()
263 auto output_samples = Resample(mixed_frame_, sample_rate, &render_resampler_, in PullRenderData()
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| D | remix_resample.cc | 71 resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(), in RemixAndResample()
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| /external/webrtc/modules/audio_processing/agc2/rnn_vad/ |
| D | rnn_vad_unittest.cc | 92 decimator.Resample(samples_48k.data(), samples_48k.size(), in TEST_P()
128 decimator.Resample(samples.data(), samples.size(), in TEST_P()
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| D | rnn_vad_tool.cc | 81 resampler.Resample(samples_10ms.data(), samples_10ms.size(), in main()
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| /external/crosvm/rutabaga_gfx/src/cross_domain/ |
| D | mod.rs | 22 Resample, enumerator
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| D | cross_domain.rs | 324 CrossDomainToken::Resample => { in handle_fence()
390 .add(CrossDomainToken::Resample, &thread_resample_evt)?; in run()
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| /external/webrtc/common_audio/resampler/include/ |
| D | push_resampler.h | 37 int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity);
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| /external/webrtc/modules/audio_coding/acm2/ |
| D | acm_resampler.cc | 49 resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples); in Resample10Msec()
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| /external/crosvm/devices/src/ |
| D | bat.rs | 153 Resample, enumerator
167 (irq_evt.get_resample(), Token::Resample), in command_monitor()
286 Token::Resample => { in command_monitor()
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| /external/webrtc/modules/audio_processing/agc2/ |
| D | vad_wrapper.cc | 100 resampler_.Resample(frame.channel(0).data(), frame_size_, in Analyze()
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| /external/rappor/analysis/R/ |
| D | decode.R | 271 Resample <- function(e) { function
381 e <- Resample(estimates_stds_filtered)
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| /external/speex/ |
| D | configure.ac | 205 AC_ARG_ENABLE(resample-full-sinc-table, [ --enable-resample-full-sinc-table Resample full SINC tab…
207 AC_DEFINE([RESAMPLE_FULL_SINC_TABLE], , [Resample with full SINC table (no interpolation)])
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| D | config.h.in | 87 /* Resample with full SINC table (no interpolation) */
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| /external/pdfium/core/fxcodec/ |
| D | progressive_decoder.h | 212 void Resample(const RetainPtr<CFX_DIBitmap>& pDeviceBitmap,
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| /external/webrtc/common_audio/ |
| D | audio_converter.cc | 112 resamplers_[i]->Resample(src[i], src_frames(), dst[i], dst_frames()); in Convert()
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| /external/webrtc/rtc_base/ |
| D | virtual_socket_server.h | 410 static std::unique_ptr<Function> Resample(std::unique_ptr<Function> f,
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| D | virtual_socket_server.cc | 1179 return Resample(Invert(Accumulate(std::move(f))), 0, 1, samples); in CreateDistribution()
1231 std::unique_ptr<VirtualSocketServer::Function> VirtualSocketServer::Resample( in Resample() function in rtc::VirtualSocketServer
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