Update audio_processing module

Corresponds to upstream commit 524e9b043e7e86fd72353b987c9d5f6a1ebf83e1

Update notes:

 * Pull in third party license file

 * Replace .gypi files with BUILD.gn to keep track of what changes
   upstream

 * Bunch of new filse pulled in as dependencies

 * Won't build yet due to changes needed on top of these
This commit is contained in:
Arun Raghavan
2015-10-13 17:25:22 +05:30
parent 5ae7a5d6cd
commit 753eada3aa
324 changed files with 52533 additions and 16117 deletions

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/*
* 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 WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
#define WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/typedefs.h"
namespace webrtc {
class PushSincResampler;
// Wraps PushSincResampler to provide stereo support.
// TODO(ajm): add support for an arbitrary number of channels.
template <typename T>
class PushResampler {
public:
PushResampler();
virtual ~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);
// 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_;
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_INCLUDE_PUSH_RESAMPLER_H_

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/*
* Copyright (c) 2011 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.
*/
/*
* A wrapper for resampling a numerous amount of sampling combinations.
*/
#ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
#define WEBRTC_RESAMPLER_RESAMPLER_H_
#include <stddef.h>
#include "webrtc/typedefs.h"
namespace webrtc {
// All methods return 0 on success and -1 on failure.
class Resampler
{
public:
Resampler();
Resampler(int inFreq, int outFreq, int num_channels);
~Resampler();
// 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, 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);
private:
enum ResamplerMode
{
kResamplerMode1To1,
kResamplerMode1To2,
kResamplerMode1To3,
kResamplerMode1To4,
kResamplerMode1To6,
kResamplerMode1To12,
kResamplerMode2To3,
kResamplerMode2To11,
kResamplerMode4To11,
kResamplerMode8To11,
kResamplerMode11To16,
kResamplerMode11To32,
kResamplerMode2To1,
kResamplerMode3To1,
kResamplerMode4To1,
kResamplerMode6To1,
kResamplerMode12To1,
kResamplerMode3To2,
kResamplerMode11To2,
kResamplerMode11To4,
kResamplerMode11To8
};
// 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_;
int my_in_frequency_khz_;
int my_out_frequency_khz_;
ResamplerMode my_mode_;
int num_channels_;
// Extra instance for stereo
Resampler* slave_left_;
Resampler* slave_right_;
};
} // namespace webrtc
#endif // WEBRTC_RESAMPLER_RESAMPLER_H_