AnotherFoxGuy/webrtc-audio-processing
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Update to current webrtc library

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This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.
This commit is contained in:
Arun Raghavan
2020-10-12 18:08:02 -04:00
parent b1b02581d3
commit bcec8b0b21
859 changed files with 76187 additions and 49580 deletions
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27
webrtc/common_audio/resampler/push_sinc_resampler.h
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@ -8,13 +8,16 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
#define WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
#ifndef COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
#define COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/resampler/sinc_resampler.h"
#include "webrtc/typedefs.h"
#include <stddef.h>
#include <stdint.h>
#include <memory>
#include "common_audio/resampler/sinc_resampler.h"
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -35,8 +38,10 @@ class PushSincResampler : public SincResamplerCallback {
// 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 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,
@ -56,8 +61,8 @@ class PushSincResampler : public SincResamplerCallback {
friend class PushSincResamplerTest;
SincResampler* get_resampler_for_testing() { return resampler_.get(); }
rtc::scoped_ptr<SincResampler> resampler_;
rtc::scoped_ptr<float[]> float_buffer_;
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_;
@ -73,4 +78,4 @@ class PushSincResampler : public SincResamplerCallback {
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_
#endif // COMMON_AUDIO_RESAMPLER_PUSH_SINC_RESAMPLER_H_