/external/webrtc/webrtc/common_audio/resampler/ |
D | push_sinc_resampler.cc | 34 size_t PushSincResampler::Resample(const int16_t* source, in Resample() function in webrtc::PushSincResampler 43 Resample(nullptr, source_length, float_buffer_.get(), destination_frames_); in Resample() 49 size_t PushSincResampler::Resample(const float* source, in Resample() function in webrtc::PushSincResampler 74 resampler_->Resample(resampler_->ChunkSize(), destination); in Resample() 76 resampler_->Resample(destination_frames_, destination); in Resample()
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D | push_resampler.cc | 69 int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst, in Resample() function in webrtc::PushResampler 89 sinc_resampler_->Resample(src_left_.get(), src_length_mono, in Resample() 91 sinc_resampler_right_->Resample(src_right_.get(), src_length_mono, in Resample() 100 sinc_resampler_->Resample(src, src_length, dst, dst_capacity)); in Resample()
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D | push_sinc_resampler_unittest.cc | 91 sinc_resampler.Resample(output_samples, resampled_destination.get()); in ResampleBenchmarkTest() 102 resampler.Resample(source_int.get(), in ResampleBenchmarkTest() 110 resampler.Resample(source.get(), in ResampleBenchmarkTest() 180 resampler.Resample(source_int.get(), in ResampleTest() 191 resampler.Resample(&source[i * input_block_size], in ResampleTest()
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D | sinc_resampler_unittest.cc | 70 resampler.Resample(resampler.ChunkSize(), resampled_destination.get()); in TEST() 76 resampler.Resample(max_chunk_size, resampled_destination.get()); in TEST() 90 resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get()); in TEST() 98 resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get()); in TEST() 251 TEST_P(SincResamplerTest, Resample) { in TEST_P() argument 288 resampler.Resample(output_samples, resampled_destination.get()); in TEST_P()
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D | push_sinc_resampler.h | 38 size_t Resample(const int16_t* source, size_t source_frames, 40 size_t Resample(const float* source,
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D | sinc_resampler.h | 65 void Resample(size_t frames, float* destination);
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D | sinc_resampler.cc | 270 void SincResampler::Resample(size_t frames, float* destination) { in Resample() function in webrtc::SincResampler
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/external/webrtc/webrtc/modules/audio_processing/ |
D | audio_buffer.cc | 134 input_resamplers_[i]->Resample(data_ptr[i], in CopyFrom() 170 output_resamplers_[i]->Resample(data_ptr[i], in CopyTo() 402 input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i], in DeinterleaveFrom() 427 output_resamplers_[i]->Resample( in InterleaveTo()
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/external/webrtc/webrtc/common_audio/resampler/include/ |
D | push_resampler.h | 36 int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity);
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/external/webrtc/webrtc/modules/audio_coding/acm2/ |
D | acm_resampler.cc | 52 resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples); in Resample10Msec()
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/external/webrtc/webrtc/voice_engine/ |
D | utility.cc | 63 int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_, in RemixAndResample()
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/external/pdfium/core/src/fxcodec/codec/ |
D | fx_codec_progress.h | 159 void Resample(CFX_DIBitmap* pDeviceBitmap,
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D | fx_codec_progress.cpp | 1786 void CCodec_ProgressiveDecoder::Resample(CFX_DIBitmap* pDeviceBitmap, in Resample() function in CCodec_ProgressiveDecoder 2106 Resample(m_pDeviceBitmap, m_SrcRow, m_pDecodeBuf, m_SrcFormat); in ContinueDecode()
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/external/webrtc/webrtc/common_audio/ |
D | audio_converter.cc | 97 resamplers_[i]->Resample(src[i], src_frames(), dst[i], dst_frames()); in Convert()
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/external/webrtc/webrtc/base/ |
D | virtualsocketserver.h | 190 static Function* Resample(Function* f,
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D | virtualsocketserver.cc | 993 return Resample(Invert(Accumulate(f)), 0, 1, samples); in CreateDistribution() 1037 VirtualSocketServer::Function* VirtualSocketServer::Resample(Function* f, in Resample() function in rtc::VirtualSocketServer
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/external/webrtc/webrtc/modules/audio_processing/test/ |
D | audio_processing_unittest.cc | 2606 static_cast<size_t>(resampler.Resample( in TEST_P()
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