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/resampler/include/push_resampler.h
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@ -8,11 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
#define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
#ifndef COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
#define COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
#include <memory>
#include <vector>
namespace webrtc {
@ -28,25 +28,32 @@ class PushResampler {
// Must be called whenever the parameters change. Free to be called at any
// time as it is a no-op if parameters have not changed since the last call.
int InitializeIfNeeded(int src_sample_rate_hz, int dst_sample_rate_hz,
int num_channels);
int InitializeIfNeeded(int src_sample_rate_hz,
int dst_sample_rate_hz,
size_t num_channels);
// Returns the total number of samples provided in destination (e.g. 32 kHz,
// 2 channel audio gives 640 samples).
int Resample(const T* src, size_t src_length, T* dst, size_t dst_capacity);
private:
rtc::scoped_ptr<PushSincResampler> sinc_resampler_;
rtc::scoped_ptr<PushSincResampler> sinc_resampler_right_;
int src_sample_rate_hz_;
int dst_sample_rate_hz_;
int num_channels_;
rtc::scoped_ptr<T[]> src_left_;
rtc::scoped_ptr<T[]> src_right_;
rtc::scoped_ptr<T[]> dst_left_;
rtc::scoped_ptr<T[]> dst_right_;
};
size_t num_channels_;
// Vector that is needed to provide the proper inputs and outputs to the
// interleave/de-interleave methods used in Resample. This needs to be
// heap-allocated on the state to support an arbitrary number of channels
// without doing run-time heap-allocations in the Resample method.
std::vector<T*> channel_data_array_;
struct ChannelResampler {
std::unique_ptr<PushSincResampler> resampler;
std::vector<T> source;
std::vector<T> destination;
};
std::vector<ChannelResampler> channel_resamplers_;
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
#endif // COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_

128
webrtc/common_audio/resampler/include/resampler.h
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@ -8,88 +8,92 @@
* be found in the AUTHORS file in the root of the source tree.
*/
/*
* A wrapper for resampling a numerous amount of sampling combinations.
*/
#ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
#define WEBRTC_RESAMPLER_RESAMPLER_H_
#ifndef COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_
#define COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_
#include <stddef.h>
#include "webrtc/typedefs.h"
#include <stdint.h>
namespace webrtc {
// All methods return 0 on success and -1 on failure.
class Resampler
{
class Resampler {
public:
Resampler();
Resampler(int inFreq, int outFreq, size_t num_channels);
~Resampler();
public:
Resampler();
Resampler(int inFreq, int outFreq, int num_channels);
~Resampler();
// Reset all states
int Reset(int inFreq, int outFreq, size_t num_channels);
// Reset all states
int Reset(int inFreq, int outFreq, int num_channels);
// Reset all states if any parameter has changed
int ResetIfNeeded(int inFreq, int outFreq, size_t num_channels);
// Reset all states if any parameter has changed
int ResetIfNeeded(int inFreq, int outFreq, int num_channels);
// Resample samplesIn to samplesOut.
int Push(const int16_t* samplesIn,
size_t lengthIn,
int16_t* samplesOut,
size_t maxLen,
size_t& outLen); // NOLINT: to avoid changing APIs
// Resample samplesIn to samplesOut.
int Push(const int16_t* samplesIn, size_t lengthIn, int16_t* samplesOut,
size_t maxLen, size_t &outLen);
private:
enum ResamplerMode {
kResamplerMode1To1,
kResamplerMode1To2,
kResamplerMode1To3,
kResamplerMode1To4,
kResamplerMode1To6,
kResamplerMode1To12,
kResamplerMode2To3,
kResamplerMode2To11,
kResamplerMode4To11,
kResamplerMode8To11,
kResamplerMode11To16,
kResamplerMode11To32,
kResamplerMode2To1,
kResamplerMode3To1,
kResamplerMode4To1,
kResamplerMode6To1,
kResamplerMode12To1,
kResamplerMode3To2,
kResamplerMode11To2,
kResamplerMode11To4,
kResamplerMode11To8
};
private:
enum ResamplerMode
{
kResamplerMode1To1,
kResamplerMode1To2,
kResamplerMode1To3,
kResamplerMode1To4,
kResamplerMode1To6,
kResamplerMode1To12,
kResamplerMode2To3,
kResamplerMode2To11,
kResamplerMode4To11,
kResamplerMode8To11,
kResamplerMode11To16,
kResamplerMode11To32,
kResamplerMode2To1,
kResamplerMode3To1,
kResamplerMode4To1,
kResamplerMode6To1,
kResamplerMode12To1,
kResamplerMode3To2,
kResamplerMode11To2,
kResamplerMode11To4,
kResamplerMode11To8
};
// Computes the resampler mode for a given sampling frequency pair.
// Returns -1 for unsupported frequency pairs.
static int ComputeResamplerMode(int in_freq_hz,
int out_freq_hz,
ResamplerMode* mode);
// Generic pointers since we don't know what states we'll need
void* state1_;
void* state2_;
void* state3_;
// Generic pointers since we don't know what states we'll need
void* state1_;
void* state2_;
void* state3_;
// Storage if needed
int16_t* in_buffer_;
int16_t* out_buffer_;
size_t in_buffer_size_;
size_t out_buffer_size_;
size_t in_buffer_size_max_;
size_t out_buffer_size_max_;
// Storage if needed
int16_t* in_buffer_;
int16_t* out_buffer_;
size_t in_buffer_size_;
size_t out_buffer_size_;
size_t in_buffer_size_max_;
size_t out_buffer_size_max_;
int my_in_frequency_khz_;
int my_out_frequency_khz_;
ResamplerMode my_mode_;
int num_channels_;
int my_in_frequency_khz_;
int my_out_frequency_khz_;
ResamplerMode my_mode_;
size_t num_channels_;
// Extra instance for stereo
Resampler* slave_left_;
Resampler* slave_right_;
// Extra instance for stereo
Resampler* helper_left_;
Resampler* helper_right_;
};
} // namespace webrtc
#endif // WEBRTC_RESAMPLER_RESAMPLER_H_
#endif // COMMON_AUDIO_RESAMPLER_INCLUDE_RESAMPLER_H_