| /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()
|
| 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()
|
| D | sinc_resampler_unittest.cc | 73 resampler.Resample(resampler.ChunkSize(), resampled_destination.get()); in TEST()
80 resampler.Resample(max_chunk_size, resampled_destination.get()); in TEST()
93 resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get()); in TEST()
100 resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get()); in TEST()
253 TEST_P(SincResamplerTest, Resample) { in TEST_P() argument
290 resampler.Resample(output_samples, resampled_destination.get()); in TEST_P()
|
| D | push_sinc_resampler.h | 41 size_t Resample(const int16_t* source,
45 size_t Resample(const float* source,
|
| D | push_resampler.cc | 108 int PushResampler<T>::Resample(const T* src, in Resample() function in webrtc::PushResampler
134 dst_length_mono = resampler.resampler->Resample( in Resample()
|
| D | sinc_resampler.h | 68 void Resample(size_t frames, float* destination);
|
| D | sinc_resampler.cc | 267 void SincResampler::Resample(size_t frames, float* destination) { in Resample() function in webrtc::SincResampler
|
| /external/webrtc/modules/audio_processing/ |
| D | audio_buffer.cc | 142 input_resamplers_[0]->Resample(downmixed_data, input_num_frames_, in CopyFrom()
151 input_resamplers_[i]->Resample(stacked_data[i], input_num_frames_, in CopyFrom()
175 output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_, in CopyTo()
197 output_resamplers_[i]->Resample(data_->channels()[i], buffer_num_frames_, in CopyTo()
246 input_resamplers_[0]->Resample(float_buffer.data(), input_num_frames_, in CopyFrom()
272 input_resamplers_[0]->Resample(downmixed_data, input_num_frames_, in CopyFrom()
292 input_resamplers_[i]->Resample(float_buffer.data(), input_num_frames_, in CopyFrom()
319 output_resamplers_[0]->Resample(data_->channels()[0], buffer_num_frames_, in CopyTo()
350 output_resamplers_[i]->Resample(data_->channels()[i], in CopyTo()
|
| D | audio_processing_unittest.cc | 2142 static_cast<size_t>(resampler.Resample( in TEST_P()
|
| /external/webrtc/audio/ |
| D | audio_transport_impl.cc | 68 int Resample(const AudioFrame& frame, in Resample() function
80 return resampler->Resample( in Resample()
203 nSamplesOut = Resample(mixed_frame_, samplesPerSec, &render_resampler_, in NeedMorePlayData()
232 auto output_samples = Resample(mixed_frame_, sample_rate, &render_resampler_, in PullRenderData()
|
| D | remix_resample.cc | 70 resampler->Resample(audio_ptr, src_length, dst_frame->mutable_data(), in RemixAndResample()
|
| /external/webrtc/modules/audio_processing/agc2/rnn_vad/ |
| D | rnn_vad_unittest.cc | 90 decimator.Resample(samples_48k.data(), samples_48k.size(), in TEST()
125 decimator.Resample(samples.data(), samples.size(), in TEST()
|
| D | rnn_vad_tool.cc | 78 resampler.Resample(samples_10ms.data(), samples_10ms.size(), in main()
|
| /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);
|
| /external/webrtc/modules/audio_coding/acm2/ |
| D | acm_resampler.cc | 50 resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples); in Resample10Msec()
|
| /external/webrtc/modules/audio_processing/agc2/ |
| D | vad_with_level.cc | 49 resampler_.Resample(frame.channel(0).data(), frame.samples_per_channel(), in AnalyzeFrame()
|
| /external/crosvm/devices/src/ |
| D | bat.rs | 156 Token::Resample, in command_monitor()
187 Resample, in command_monitor() enumerator
283 Token::Resample => { in command_monitor()
|
| /external/speex/ |
| D | configure.ac | 231 AC_ARG_ENABLE(resample-full-sinc-table, [ --enable-resample-full-sinc-table Resample full SINC tab…
233 AC_DEFINE([RESAMPLE_FULL_SINC_TABLE], , [Resample with full SINC table (no interpolation)])
|
| D | config.h.in | 97 /* Resample with full SINC table (no interpolation) */
|
| /external/webrtc/common_audio/ |
| D | audio_converter.cc | 112 resamplers_[i]->Resample(src[i], src_frames(), dst[i], dst_frames()); in Convert()
|
| /external/pdfium/core/fxcodec/ |
| D | progressivedecoder.h | 235 void Resample(const RetainPtr<CFX_DIBitmap>& pDeviceBitmap,
|
| D | progressivedecoder.cpp | 1138 Resample(m_pDeviceBitmap, m_SrcRow, m_pDecodeBuf.get(), m_SrcFormat); in JpegContinueDecode()
2128 void ProgressiveDecoder::Resample(const RetainPtr<CFX_DIBitmap>& pDeviceBitmap, in Resample() function in fxcodec::ProgressiveDecoder
|
| /external/webrtc/rtc_base/ |
| D | virtual_socket_server.h | 228 static Function* Resample(Function* f,
|
| D | virtual_socket_server.cc | 1066 return Resample(Invert(Accumulate(f)), 0, 1, samples); in CreateDistribution()
1116 VirtualSocketServer::Function* VirtualSocketServer::Resample(Function* f, in Resample() function in rtc::VirtualSocketServer
|
| /external/ImageMagick/PerlMagick/quantum/ |
| D | quantum.xs.in | 418 { "Resample", { {"density", StringReference}, {"x", RealReference},
7543 Resample = 155
10120 case 78: /* Resample */
|