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Diffstat (limited to 'third_party/libwebrtc/common_audio/resampler/push_sinc_resampler.h')
-rw-r--r-- | third_party/libwebrtc/common_audio/resampler/push_sinc_resampler.h | 81 |
1 files changed, 81 insertions, 0 deletions
diff --git a/third_party/libwebrtc/common_audio/resampler/push_sinc_resampler.h b/third_party/libwebrtc/common_audio/resampler/push_sinc_resampler.h new file mode 100644 index 0000000000..7946ef8f82 --- /dev/null +++ b/third_party/libwebrtc/common_audio/resampler/push_sinc_resampler.h @@ -0,0 +1,81 @@ +/* + * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#ifndef COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_ +#define COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_ + +#include <stddef.h> +#include <stdint.h> + +#include <memory> + +#include "common_audio/resampler/sinc_resampler.h" + +namespace webrtc { + +// A thin wrapper over SincResampler to provide a push-based interface as +// required by WebRTC. SincResampler uses a pull-based interface, and will +// use SincResamplerCallback::Run() to request data upon a call to Resample(). +// These Run() calls will happen on the same thread Resample() is called on. +class PushSincResampler : public SincResamplerCallback { + public: + // Provide the size of the source and destination blocks in samples. These + // must correspond to the same time duration (typically 10 ms) as the sample + // ratio is inferred from them. + PushSincResampler(size_t source_frames, size_t destination_frames); + ~PushSincResampler() override; + + PushSincResampler(const PushSincResampler&) = delete; + PushSincResampler& operator=(const PushSincResampler&) = delete; + + // Perform the resampling. `source_frames` must always equal the + // `source_frames` provided at construction. `destination_capacity` must be + // at least as large as `destination_frames`. Returns the number of samples + // provided in destination (for convenience, since this will always be equal + // to `destination_frames`). + size_t Resample(const int16_t* source, + size_t source_frames, + int16_t* destination, + size_t destination_capacity); + size_t Resample(const float* source, + size_t source_frames, + float* destination, + size_t destination_capacity); + + // Delay due to the filter kernel. Essentially, the time after which an input + // sample will appear in the resampled output. + static float AlgorithmicDelaySeconds(int source_rate_hz) { + return 1.f / source_rate_hz * SincResampler::kKernelSize / 2; + } + + protected: + // Implements SincResamplerCallback. + void Run(size_t frames, float* destination) override; + + private: + friend class PushSincResamplerTest; + SincResampler* get_resampler_for_testing() { return resampler_.get(); } + + std::unique_ptr<SincResampler> resampler_; + std::unique_ptr<float[]> float_buffer_; + const float* source_ptr_; + const int16_t* source_ptr_int_; + const size_t destination_frames_; + + // True on the first call to Resample(), to prime the SincResampler buffer. + bool first_pass_; + + // Used to assert we are only requested for as much data as is available. + size_t source_available_; +}; + +} // namespace webrtc + +#endif // COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_ |