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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37
webrtc/common_audio/audio_converter.h
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@ -8,11 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
#define WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
#ifndef COMMON_AUDIO_AUDIO_CONVERTER_H_
#define COMMON_AUDIO_AUDIO_CONVERTER_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include <stddef.h>
#include <memory>
#include "rtc_base/constructor_magic.h"
namespace webrtc {
@ -26,36 +29,40 @@ class AudioConverter {
public:
// Returns a new AudioConverter, which will use the supplied format for its
// lifetime. Caller is responsible for the memory.
static rtc::scoped_ptr<AudioConverter> Create(int src_channels,
static std::unique_ptr<AudioConverter> Create(size_t src_channels,
size_t src_frames,
int dst_channels,
size_t dst_channels,
size_t dst_frames);
virtual ~AudioConverter() {};
virtual ~AudioConverter() {}
// Convert |src|, containing |src_size| samples, to |dst|, having a sample
// capacity of |dst_capacity|. Both point to a series of buffers containing
// the samples for each channel. The sizes must correspond to the format
// passed to Create().
virtual void Convert(const float* const* src, size_t src_size,
float* const* dst, size_t dst_capacity) = 0;
virtual void Convert(const float* const* src,
size_t src_size,
float* const* dst,
size_t dst_capacity) = 0;
int src_channels() const { return src_channels_; }
size_t src_channels() const { return src_channels_; }
size_t src_frames() const { return src_frames_; }
int dst_channels() const { return dst_channels_; }
size_t dst_channels() const { return dst_channels_; }
size_t dst_frames() const { return dst_frames_; }
protected:
AudioConverter();
AudioConverter(int src_channels, size_t src_frames, int dst_channels,
AudioConverter(size_t src_channels,
size_t src_frames,
size_t dst_channels,
size_t dst_frames);
// Helper to RTC_CHECK that inputs are correctly sized.
void CheckSizes(size_t src_size, size_t dst_capacity) const;
private:
const int src_channels_;
const size_t src_channels_;
const size_t src_frames_;
const int dst_channels_;
const size_t dst_channels_;
const size_t dst_frames_;
RTC_DISALLOW_COPY_AND_ASSIGN(AudioConverter);
@ -63,4 +70,4 @@ class AudioConverter {
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_AUDIO_CONVERTER_H_
#endif // COMMON_AUDIO_AUDIO_CONVERTER_H_