| /external/webrtc/webrtc/common_audio/resampler/ |
| push_sinc_resampler.cc | 34 size_t PushSincResampler::Resample(const int16_t* source,
43 Resample(nullptr, source_length, float_buffer_.get(), destination_frames_);
49 size_t PushSincResampler::Resample(const float* source,
55 // Cache the source pointer. Calling Resample() will immediately trigger
60 // On the first pass, we call Resample() twice. During the first call, we
63 // ensuring that all later Resample() calls will only result in one input
74 resampler_->Resample(resampler_->ChunkSize(), destination);
76 resampler_->Resample(destination_frames_, destination);
83 // Run() was triggered more than once per Resample() call.
88 // discarded, as described in Resample()
[all...] |
| push_sinc_resampler.h | 23 // use SincResamplerCallback::Run() to request data upon a call to Resample().
24 // These Run() calls will happen on the same thread Resample() is called on.
38 size_t Resample(const int16_t* source, size_t source_frames,
40 size_t Resample(const float* source,
65 // True on the first call to Resample(), to prime the SincResampler buffer.
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| push_resampler.cc | 69 int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
89 sinc_resampler_->Resample(src_left_.get(), src_length_mono,
91 sinc_resampler_right_->Resample(src_right_.get(), src_length_mono,
100 sinc_resampler_->Resample(src, src_length, dst, dst_capacity));
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| sinc_resampler.h | 64 // Resample |frames| of data from |read_cb_| into |destination|.
65 void Resample(size_t frames, float* destination);
67 // The maximum size in frames that guarantees Resample() will only make a
74 // not call while Resample() is in progress.
79 // Resample() is in progress.
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| push_sinc_resampler_unittest.cc | 91 sinc_resampler.Resample(output_samples, resampled_destination.get());
102 resampler.Resample(source_int.get(),
110 resampler.Resample(source.get(),
166 // checked before the first call to Resample(), because ChunkSize() will
180 resampler.Resample(source_int.get(),
191 resampler.Resample(&source[i * input_block_size],
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| sinc_resampler_unittest.cc | 70 resampler.Resample(resampler.ChunkSize(), resampled_destination.get());
76 resampler.Resample(max_chunk_size, resampled_destination.get());
90 resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get());
98 resampler.Resample(resampler.ChunkSize() / 2, resampled_destination.get());
251 TEST_P(SincResamplerTest, Resample) {
288 resampler.Resample(output_samples, resampled_destination.get());
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| sinc_resampler.cc | 270 void SincResampler::Resample(size_t frames, float* destination) {
279 // Step (2) -- Resample! const what we can outside of the loop for speed. It
284 // |i| may be negative if the last Resample() call ended on an iteration
|
| /external/webrtc/webrtc/common_audio/resampler/include/ |
| 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/ |
| acm_resampler.cc | 52 resampler_.Resample(in_audio, in_length, out_audio, out_capacity_samples);
54 LOG(LS_ERROR) << "Resample(" << in_audio << ", " << in_length << ", "
|
| /external/webrtc/webrtc/modules/audio_processing/ |
| audio_buffer.cc | 131 // Resample.
134 input_resamplers_[i]->Resample(data_ptr[i],
167 // Resample.
170 output_resamplers_[i]->Resample(data_ptr[i],
361 // We don't resample the keyboard channel.
399 // Resample.
402 input_resamplers_[i]->Resample(input_buffer_->fbuf_const()->channels()[i],
419 // Resample if necessary.
427 output_resamplers_[i]->Resample(
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| /external/webrtc/webrtc/voice_engine/ |
| utility.cc | 63 int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
66 LOG(LS_ERROR) << "Resample failed: audio_ptr = " << audio_ptr
|
| /external/pdfium/core/src/fxcodec/codec/ |
| fx_codec_progress.h | 159 void Resample(CFX_DIBitmap* pDeviceBitmap,
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| fx_codec_progress.cpp | [all...] |
| /external/webrtc/webrtc/base/ |
| virtualsocketserver.h | 190 static Function* Resample(Function* f,
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| virtualsocketserver.cc | 993 return Resample(Invert(Accumulate(f)), 0, 1, samples);
1037 VirtualSocketServer::Function* VirtualSocketServer::Resample(Function* f,
[all...] |
| /external/webrtc/webrtc/common_audio/ |
| audio_converter.cc | 97 resamplers_[i]->Resample(src[i], src_frames(), dst[i], dst_frames());
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| /external/webrtc/webrtc/modules/audio_processing/test/ |
| audio_processing_unittest.cc | [all...] |
| /external/ImageMagick/PerlMagick/ |
| Magick.xs | 419 { "Resample", { {"density", StringReference}, {"x", RealReference},
[all...] |
| /external/ImageMagick/PerlMagick/quantum/ |
| quantum.xs.in | 419 { "Resample", { {"density", StringReference}, {"x", RealReference},
[all...] |
