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Update audio_processing module

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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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52
webrtc/common_audio/resampler/include/push_resampler.h Normal file
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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_

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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.
*/
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include <string.h>
#include "webrtc/common_audio/include/audio_util.h"
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
namespace webrtc {
template <typename T>
PushResampler<T>::PushResampler()
: src_sample_rate_hz_(0),
dst_sample_rate_hz_(0),
num_channels_(0) {
}
template <typename T>
PushResampler<T>::~PushResampler() {
}
template <typename T>
int PushResampler<T>::InitializeIfNeeded(int src_sample_rate_hz,
int dst_sample_rate_hz,
int num_channels) {
if (src_sample_rate_hz == src_sample_rate_hz_ &&
dst_sample_rate_hz == dst_sample_rate_hz_ &&
num_channels == num_channels_)
// No-op if settings haven't changed.
return 0;
if (src_sample_rate_hz <= 0 || dst_sample_rate_hz <= 0 ||
num_channels <= 0 || num_channels > 2)
return -1;
src_sample_rate_hz_ = src_sample_rate_hz;
dst_sample_rate_hz_ = dst_sample_rate_hz;
num_channels_ = num_channels;
const size_t src_size_10ms_mono =
static_cast<size_t>(src_sample_rate_hz / 100);
const size_t dst_size_10ms_mono =
static_cast<size_t>(dst_sample_rate_hz / 100);
sinc_resampler_.reset(new PushSincResampler(src_size_10ms_mono,
dst_size_10ms_mono));
if (num_channels_ == 2) {
src_left_.reset(new T[src_size_10ms_mono]);
src_right_.reset(new T[src_size_10ms_mono]);
dst_left_.reset(new T[dst_size_10ms_mono]);
dst_right_.reset(new T[dst_size_10ms_mono]);
sinc_resampler_right_.reset(new PushSincResampler(src_size_10ms_mono,
dst_size_10ms_mono));
}
return 0;
}
template <typename T>
int PushResampler<T>::Resample(const T* src, size_t src_length, T* dst,
size_t dst_capacity) {
const size_t src_size_10ms =
static_cast<size_t>(src_sample_rate_hz_ * num_channels_ / 100);
const size_t dst_size_10ms =
static_cast<size_t>(dst_sample_rate_hz_ * num_channels_ / 100);
if (src_length != src_size_10ms || dst_capacity < dst_size_10ms)
return -1;
if (src_sample_rate_hz_ == dst_sample_rate_hz_) {
// The old resampler provides this memcpy facility in the case of matching
// sample rates, so reproduce it here for the sinc resampler.
memcpy(dst, src, src_length * sizeof(T));
return static_cast<int>(src_length);
}
if (num_channels_ == 2) {
const size_t src_length_mono = src_length / num_channels_;
const size_t dst_capacity_mono = dst_capacity / num_channels_;
T* deinterleaved[] = {src_left_.get(), src_right_.get()};
Deinterleave(src, src_length_mono, num_channels_, deinterleaved);
size_t dst_length_mono =
sinc_resampler_->Resample(src_left_.get(), src_length_mono,
dst_left_.get(), dst_capacity_mono);
sinc_resampler_right_->Resample(src_right_.get(), src_length_mono,
dst_right_.get(), dst_capacity_mono);
deinterleaved[0] = dst_left_.get();
deinterleaved[1] = dst_right_.get();
Interleave(deinterleaved, dst_length_mono, num_channels_, dst);
return static_cast<int>(dst_length_mono * num_channels_);
} else {
return static_cast<int>(
sinc_resampler_->Resample(src, src_length, dst, dst_capacity));
}
}
// Explictly generate required instantiations.
template class PushResampler<int16_t>;
template class PushResampler<float>;
} // namespace webrtc

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webrtc/common_audio/resampler/push_sinc_resampler.cc Normal file
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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.
*/
#include "webrtc/common_audio/resampler/push_sinc_resampler.h"
#include <cstring>
#include "webrtc/base/checks.h"
#include "webrtc/common_audio/include/audio_util.h"
namespace webrtc {
PushSincResampler::PushSincResampler(size_t source_frames,
size_t destination_frames)
: resampler_(new SincResampler(source_frames * 1.0 / destination_frames,
source_frames,
this)),
source_ptr_(nullptr),
source_ptr_int_(nullptr),
destination_frames_(destination_frames),
first_pass_(true),
source_available_(0) {}
PushSincResampler::~PushSincResampler() {
}
size_t PushSincResampler::Resample(const int16_t* source,
size_t source_length,
int16_t* destination,
size_t destination_capacity) {
if (!float_buffer_.get())
float_buffer_.reset(new float[destination_frames_]);
source_ptr_int_ = source;
// Pass nullptr as the float source to have Run() read from the int16 source.
Resample(nullptr, source_length, float_buffer_.get(), destination_frames_);
FloatS16ToS16(float_buffer_.get(), destination_frames_, destination);
source_ptr_int_ = nullptr;
return destination_frames_;
}
size_t PushSincResampler::Resample(const float* source,
size_t source_length,
float* destination,
size_t destination_capacity) {
RTC_CHECK_EQ(source_length, resampler_->request_frames());
RTC_CHECK_GE(destination_capacity, destination_frames_);
// Cache the source pointer. Calling Resample() will immediately trigger
// the Run() callback whereupon we provide the cached value.
source_ptr_ = source;
source_available_ = source_length;
// On the first pass, we call Resample() twice. During the first call, we
// provide dummy input and discard the output. This is done to prime the
// SincResampler buffer with the correct delay (half the kernel size), thereby
// ensuring that all later Resample() calls will only result in one input
// request through Run().
//
// If this wasn't done, SincResampler would call Run() twice on the first
// pass, and we'd have to introduce an entire |source_frames| of delay, rather
// than the minimum half kernel.
//
// It works out that ChunkSize() is exactly the amount of output we need to
// request in order to prime the buffer with a single Run() request for
// |source_frames|.
if (first_pass_)
resampler_->Resample(resampler_->ChunkSize(), destination);
resampler_->Resample(destination_frames_, destination);
source_ptr_ = nullptr;
return destination_frames_;
}
void PushSincResampler::Run(size_t frames, float* destination) {
// Ensure we are only asked for the available samples. This would fail if
// Run() was triggered more than once per Resample() call.
RTC_CHECK_EQ(source_available_, frames);
if (first_pass_) {
// Provide dummy input on the first pass, the output of which will be
// discarded, as described in Resample().
std::memset(destination, 0, frames * sizeof(*destination));
first_pass_ = false;
return;
}
if (source_ptr_) {
std::memcpy(destination, source_ptr_, frames * sizeof(*destination));
} else {
for (size_t i = 0; i < frames; ++i)
destination[i] = static_cast<float>(source_ptr_int_[i]);
}
source_available_ -= frames;
}
} // namespace webrtc

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webrtc/common_audio/resampler/push_sinc_resampler.h Normal file
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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_PUSH_SINC_RESAMPLER_H_
#define WEBRTC_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"
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;
// 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(); }
rtc::scoped_ptr<SincResampler> resampler_;
rtc::scoped_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_;
RTC_DISALLOW_COPY_AND_ASSIGN(PushSincResampler);
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_PUSH_SINC_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.
*/
#include <stdlib.h>
#include <string.h>
#include "webrtc/common_audio/resampler/include/resampler.h"
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
namespace webrtc {
Resampler::Resampler()
: state1_(nullptr),
state2_(nullptr),
state3_(nullptr),
in_buffer_(nullptr),
out_buffer_(nullptr),
in_buffer_size_(0),
out_buffer_size_(0),
in_buffer_size_max_(0),
out_buffer_size_max_(0),
my_in_frequency_khz_(0),
my_out_frequency_khz_(0),
my_mode_(kResamplerMode1To1),
num_channels_(0),
slave_left_(nullptr),
slave_right_(nullptr) {
}
Resampler::Resampler(int inFreq, int outFreq, int num_channels)
: Resampler() {
Reset(inFreq, outFreq, num_channels);
}
Resampler::~Resampler()
{
if (state1_)
{
free(state1_);
}
if (state2_)
{
free(state2_);
}
if (state3_)
{
free(state3_);
}
if (in_buffer_)
{
free(in_buffer_);
}
if (out_buffer_)
{
free(out_buffer_);
}
if (slave_left_)
{
delete slave_left_;
}
if (slave_right_)
{
delete slave_right_;
}
}
int Resampler::ResetIfNeeded(int inFreq, int outFreq, int num_channels)
{
int tmpInFreq_kHz = inFreq / 1000;
int tmpOutFreq_kHz = outFreq / 1000;
if ((tmpInFreq_kHz != my_in_frequency_khz_) || (tmpOutFreq_kHz != my_out_frequency_khz_)
|| (num_channels != num_channels_))
{
return Reset(inFreq, outFreq, num_channels);
} else
{
return 0;
}
}
int Resampler::Reset(int inFreq, int outFreq, int num_channels)
{
if (num_channels != 1 && num_channels != 2) {
return -1;
}
num_channels_ = num_channels;
if (state1_)
{
free(state1_);
state1_ = NULL;
}
if (state2_)
{
free(state2_);
state2_ = NULL;
}
if (state3_)
{
free(state3_);
state3_ = NULL;
}
if (in_buffer_)
{
free(in_buffer_);
in_buffer_ = NULL;
}
if (out_buffer_)
{
free(out_buffer_);
out_buffer_ = NULL;
}
if (slave_left_)
{
delete slave_left_;
slave_left_ = NULL;
}
if (slave_right_)
{
delete slave_right_;
slave_right_ = NULL;
}
in_buffer_size_ = 0;
out_buffer_size_ = 0;
in_buffer_size_max_ = 0;
out_buffer_size_max_ = 0;
// Start with a math exercise, Euclid's algorithm to find the gcd:
int a = inFreq;
int b = outFreq;
int c = a % b;
while (c != 0)
{
a = b;
b = c;
c = a % b;
}
// b is now the gcd;
// We need to track what domain we're in.
my_in_frequency_khz_ = inFreq / 1000;
my_out_frequency_khz_ = outFreq / 1000;
// Scale with GCD
inFreq = inFreq / b;
outFreq = outFreq / b;
if (num_channels_ == 2)
{
// Create two mono resamplers.
slave_left_ = new Resampler(inFreq, outFreq, 1);
slave_right_ = new Resampler(inFreq, outFreq, 1);
}
if (inFreq == outFreq)
{
my_mode_ = kResamplerMode1To1;
} else if (inFreq == 1)
{
switch (outFreq)
{
case 2:
my_mode_ = kResamplerMode1To2;
break;
case 3:
my_mode_ = kResamplerMode1To3;
break;
case 4:
my_mode_ = kResamplerMode1To4;
break;
case 6:
my_mode_ = kResamplerMode1To6;
break;
case 12:
my_mode_ = kResamplerMode1To12;
break;
default:
return -1;
}
} else if (outFreq == 1)
{
switch (inFreq)
{
case 2:
my_mode_ = kResamplerMode2To1;
break;
case 3:
my_mode_ = kResamplerMode3To1;
break;
case 4:
my_mode_ = kResamplerMode4To1;
break;
case 6:
my_mode_ = kResamplerMode6To1;
break;
case 12:
my_mode_ = kResamplerMode12To1;
break;
default:
return -1;
}
} else if ((inFreq == 2) && (outFreq == 3))
{
my_mode_ = kResamplerMode2To3;
} else if ((inFreq == 2) && (outFreq == 11))
{
my_mode_ = kResamplerMode2To11;
} else if ((inFreq == 4) && (outFreq == 11))
{
my_mode_ = kResamplerMode4To11;
} else if ((inFreq == 8) && (outFreq == 11))
{
my_mode_ = kResamplerMode8To11;
} else if ((inFreq == 3) && (outFreq == 2))
{
my_mode_ = kResamplerMode3To2;
} else if ((inFreq == 11) && (outFreq == 2))
{
my_mode_ = kResamplerMode11To2;
} else if ((inFreq == 11) && (outFreq == 4))
{
my_mode_ = kResamplerMode11To4;
} else if ((inFreq == 11) && (outFreq == 16))
{
my_mode_ = kResamplerMode11To16;
} else if ((inFreq == 11) && (outFreq == 32))
{
my_mode_ = kResamplerMode11To32;
} else if ((inFreq == 11) && (outFreq == 8))
{
my_mode_ = kResamplerMode11To8;
} else
{
return -1;
}
// Now create the states we need
switch (my_mode_)
{
case kResamplerMode1To1:
// No state needed;
break;
case kResamplerMode1To2:
state1_ = malloc(8 * sizeof(int32_t));
memset(state1_, 0, 8 * sizeof(int32_t));
break;
case kResamplerMode1To3:
state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state1_);
break;
case kResamplerMode1To4:
// 1:2
state1_ = malloc(8 * sizeof(int32_t));
memset(state1_, 0, 8 * sizeof(int32_t));
// 2:4
state2_ = malloc(8 * sizeof(int32_t));
memset(state2_, 0, 8 * sizeof(int32_t));
break;
case kResamplerMode1To6:
// 1:2
state1_ = malloc(8 * sizeof(int32_t));
memset(state1_, 0, 8 * sizeof(int32_t));
// 2:6
state2_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state2_);
break;
case kResamplerMode1To12:
// 1:2
state1_ = malloc(8 * sizeof(int32_t));
memset(state1_, 0, 8 * sizeof(int32_t));
// 2:4
state2_ = malloc(8 * sizeof(int32_t));
memset(state2_, 0, 8 * sizeof(int32_t));
// 4:12
state3_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
WebRtcSpl_ResetResample16khzTo48khz(
(WebRtcSpl_State16khzTo48khz*) state3_);
break;
case kResamplerMode2To3:
// 2:6
state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state1_);
// 6:3
state2_ = malloc(8 * sizeof(int32_t));
memset(state2_, 0, 8 * sizeof(int32_t));
break;
case kResamplerMode2To11:
state1_ = malloc(8 * sizeof(int32_t));
memset(state1_, 0, 8 * sizeof(int32_t));
state2_ = malloc(sizeof(WebRtcSpl_State8khzTo22khz));
WebRtcSpl_ResetResample8khzTo22khz((WebRtcSpl_State8khzTo22khz *)state2_);
break;
case kResamplerMode4To11:
state1_ = malloc(sizeof(WebRtcSpl_State8khzTo22khz));
WebRtcSpl_ResetResample8khzTo22khz((WebRtcSpl_State8khzTo22khz *)state1_);
break;
case kResamplerMode8To11:
state1_ = malloc(sizeof(WebRtcSpl_State16khzTo22khz));
WebRtcSpl_ResetResample16khzTo22khz((WebRtcSpl_State16khzTo22khz *)state1_);
break;
case kResamplerMode11To16:
state1_ = malloc(8 * sizeof(int32_t));
memset(state1_, 0, 8 * sizeof(int32_t));
state2_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state2_);
break;
case kResamplerMode11To32:
// 11 -> 22
state1_ = malloc(8 * sizeof(int32_t));
memset(state1_, 0, 8 * sizeof(int32_t));
// 22 -> 16
state2_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state2_);
// 16 -> 32
state3_ = malloc(8 * sizeof(int32_t));
memset(state3_, 0, 8 * sizeof(int32_t));
break;
case kResamplerMode2To1:
state1_ = malloc(8 * sizeof(int32_t));
memset(state1_, 0, 8 * sizeof(int32_t));
break;
case kResamplerMode3To1:
state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state1_);
break;
case kResamplerMode4To1:
// 4:2
state1_ = malloc(8 * sizeof(int32_t));
memset(state1_, 0, 8 * sizeof(int32_t));
// 2:1
state2_ = malloc(8 * sizeof(int32_t));
memset(state2_, 0, 8 * sizeof(int32_t));
break;
case kResamplerMode6To1:
// 6:2
state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state1_);
// 2:1
state2_ = malloc(8 * sizeof(int32_t));
memset(state2_, 0, 8 * sizeof(int32_t));
break;
case kResamplerMode12To1:
// 12:4
state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
WebRtcSpl_ResetResample48khzTo16khz(
(WebRtcSpl_State48khzTo16khz*) state1_);
// 4:2
state2_ = malloc(8 * sizeof(int32_t));
memset(state2_, 0, 8 * sizeof(int32_t));
// 2:1
state3_ = malloc(8 * sizeof(int32_t));
memset(state3_, 0, 8 * sizeof(int32_t));
break;
case kResamplerMode3To2:
// 3:6
state1_ = malloc(8 * sizeof(int32_t));
memset(state1_, 0, 8 * sizeof(int32_t));
// 6:2
state2_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state2_);
break;
case kResamplerMode11To2:
state1_ = malloc(sizeof(WebRtcSpl_State22khzTo8khz));
WebRtcSpl_ResetResample22khzTo8khz((WebRtcSpl_State22khzTo8khz *)state1_);
state2_ = malloc(8 * sizeof(int32_t));
memset(state2_, 0, 8 * sizeof(int32_t));
break;
case kResamplerMode11To4:
state1_ = malloc(sizeof(WebRtcSpl_State22khzTo8khz));
WebRtcSpl_ResetResample22khzTo8khz((WebRtcSpl_State22khzTo8khz *)state1_);
break;
case kResamplerMode11To8:
state1_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state1_);
break;
}
return 0;
}
// Synchronous resampling, all output samples are written to samplesOut
int Resampler::Push(const int16_t * samplesIn, size_t lengthIn,
int16_t* samplesOut, size_t maxLen, size_t &outLen)
{
if (num_channels_ == 2)
{
// Split up the signal and call the slave object for each channel
int16_t* left = (int16_t*)malloc(lengthIn * sizeof(int16_t) / 2);
int16_t* right = (int16_t*)malloc(lengthIn * sizeof(int16_t) / 2);
int16_t* out_left = (int16_t*)malloc(maxLen / 2 * sizeof(int16_t));
int16_t* out_right =
(int16_t*)malloc(maxLen / 2 * sizeof(int16_t));
int res = 0;
for (size_t i = 0; i < lengthIn; i += 2)
{
left[i >> 1] = samplesIn[i];
right[i >> 1] = samplesIn[i + 1];
}
// It's OK to overwrite the local parameter, since it's just a copy
lengthIn = lengthIn / 2;
size_t actualOutLen_left = 0;
size_t actualOutLen_right = 0;
// Do resampling for right channel
res |= slave_left_->Push(left, lengthIn, out_left, maxLen / 2, actualOutLen_left);
res |= slave_right_->Push(right, lengthIn, out_right, maxLen / 2, actualOutLen_right);
if (res || (actualOutLen_left != actualOutLen_right))
{
free(left);
free(right);
free(out_left);
free(out_right);
return -1;
}
// Reassemble the signal
for (size_t i = 0; i < actualOutLen_left; i++)
{
samplesOut[i * 2] = out_left[i];
samplesOut[i * 2 + 1] = out_right[i];
}
outLen = 2 * actualOutLen_left;
free(left);
free(right);
free(out_left);
free(out_right);
return 0;
}
// Containers for temp samples
int16_t* tmp;
int16_t* tmp_2;
// tmp data for resampling routines
int32_t* tmp_mem;
switch (my_mode_)
{
case kResamplerMode1To1:
memcpy(samplesOut, samplesIn, lengthIn * sizeof(int16_t));
outLen = lengthIn;
break;
case kResamplerMode1To2:
if (maxLen < (lengthIn * 2))
{
return -1;
}
WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut, (int32_t*)state1_);
outLen = lengthIn * 2;
return 0;
case kResamplerMode1To3:
// We can only handle blocks of 160 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 160) != 0)
{
return -1;
}
if (maxLen < (lengthIn * 3))
{
return -1;
}
tmp_mem = (int32_t*)malloc(336 * sizeof(int32_t));
for (size_t i = 0; i < lengthIn; i += 160)
{
WebRtcSpl_Resample16khzTo48khz(samplesIn + i, samplesOut + i * 3,
(WebRtcSpl_State16khzTo48khz *)state1_,
tmp_mem);
}
outLen = lengthIn * 3;
free(tmp_mem);
return 0;
case kResamplerMode1To4:
if (maxLen < (lengthIn * 4))
{
return -1;
}
tmp = (int16_t*)malloc(sizeof(int16_t) * 2 * lengthIn);
// 1:2
WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (int32_t*)state1_);
// 2:4
WebRtcSpl_UpsampleBy2(tmp, lengthIn * 2, samplesOut, (int32_t*)state2_);
outLen = lengthIn * 4;
free(tmp);
return 0;
case kResamplerMode1To6:
// We can only handle blocks of 80 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 80) != 0)
{
return -1;
}
if (maxLen < (lengthIn * 6))
{
return -1;
}
//1:2
tmp_mem = (int32_t*)malloc(336 * sizeof(int32_t));
tmp = (int16_t*)malloc(sizeof(int16_t) * 2 * lengthIn);
WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (int32_t*)state1_);
outLen = lengthIn * 2;
for (size_t i = 0; i < outLen; i += 160)
{
WebRtcSpl_Resample16khzTo48khz(tmp + i, samplesOut + i * 3,
(WebRtcSpl_State16khzTo48khz *)state2_,
tmp_mem);
}
outLen = outLen * 3;
free(tmp_mem);
free(tmp);
return 0;
case kResamplerMode1To12:
// We can only handle blocks of 40 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 40) != 0) {
return -1;
}
if (maxLen < (lengthIn * 12)) {
return -1;
}
tmp_mem = (int32_t*) malloc(336 * sizeof(int32_t));
tmp = (int16_t*) malloc(sizeof(int16_t) * 4 * lengthIn);
//1:2
WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut,
(int32_t*) state1_);
outLen = lengthIn * 2;
//2:4
WebRtcSpl_UpsampleBy2(samplesOut, outLen, tmp, (int32_t*) state2_);
outLen = outLen * 2;
// 4:12
for (size_t i = 0; i < outLen; i += 160) {
// WebRtcSpl_Resample16khzTo48khz() takes a block of 160 samples
// as input and outputs a resampled block of 480 samples. The
// data is now actually in 32 kHz sampling rate, despite the
// function name, and with a resampling factor of three becomes
// 96 kHz.
WebRtcSpl_Resample16khzTo48khz(tmp + i, samplesOut + i * 3,
(WebRtcSpl_State16khzTo48khz*) state3_,
tmp_mem);
}
outLen = outLen * 3;
free(tmp_mem);
free(tmp);
return 0;
case kResamplerMode2To3:
if (maxLen < (lengthIn * 3 / 2))
{
return -1;
}
// 2:6
// We can only handle blocks of 160 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 160) != 0)
{
return -1;
}
tmp = static_cast<int16_t*> (malloc(sizeof(int16_t) * lengthIn * 3));
tmp_mem = (int32_t*)malloc(336 * sizeof(int32_t));
for (size_t i = 0; i < lengthIn; i += 160)
{
WebRtcSpl_Resample16khzTo48khz(samplesIn + i, tmp + i * 3,
(WebRtcSpl_State16khzTo48khz *)state1_,
tmp_mem);
}
lengthIn = lengthIn * 3;
// 6:3
WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut, (int32_t*)state2_);
outLen = lengthIn / 2;
free(tmp);
free(tmp_mem);
return 0;
case kResamplerMode2To11:
// We can only handle blocks of 80 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 80) != 0)
{
return -1;
}
if (maxLen < ((lengthIn * 11) / 2))
{
return -1;
}
tmp = (int16_t*)malloc(sizeof(int16_t) * 2 * lengthIn);
// 1:2
WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (int32_t*)state1_);
lengthIn *= 2;
tmp_mem = (int32_t*)malloc(98 * sizeof(int32_t));
for (size_t i = 0; i < lengthIn; i += 80)
{
WebRtcSpl_Resample8khzTo22khz(tmp + i, samplesOut + (i * 11) / 4,
(WebRtcSpl_State8khzTo22khz *)state2_,
tmp_mem);
}
outLen = (lengthIn * 11) / 4;
free(tmp_mem);
free(tmp);
return 0;
case kResamplerMode4To11:
// We can only handle blocks of 80 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 80) != 0)
{
return -1;
}
if (maxLen < ((lengthIn * 11) / 4))
{
return -1;
}
tmp_mem = (int32_t*)malloc(98 * sizeof(int32_t));
for (size_t i = 0; i < lengthIn; i += 80)
{
WebRtcSpl_Resample8khzTo22khz(samplesIn + i, samplesOut + (i * 11) / 4,
(WebRtcSpl_State8khzTo22khz *)state1_,
tmp_mem);
}
outLen = (lengthIn * 11) / 4;
free(tmp_mem);
return 0;
case kResamplerMode8To11:
// We can only handle blocks of 160 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 160) != 0)
{
return -1;
}
if (maxLen < ((lengthIn * 11) / 8))
{
return -1;
}
tmp_mem = (int32_t*)malloc(88 * sizeof(int32_t));
for (size_t i = 0; i < lengthIn; i += 160)
{
WebRtcSpl_Resample16khzTo22khz(samplesIn + i, samplesOut + (i * 11) / 8,
(WebRtcSpl_State16khzTo22khz *)state1_,
tmp_mem);
}
outLen = (lengthIn * 11) / 8;
free(tmp_mem);
return 0;
case kResamplerMode11To16:
// We can only handle blocks of 110 samples
if ((lengthIn % 110) != 0)
{
return -1;
}
if (maxLen < ((lengthIn * 16) / 11))
{
return -1;
}
tmp_mem = (int32_t*)malloc(104 * sizeof(int32_t));
tmp = (int16_t*)malloc((sizeof(int16_t) * lengthIn * 2));
WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (int32_t*)state1_);
for (size_t i = 0; i < (lengthIn * 2); i += 220)
{
WebRtcSpl_Resample22khzTo16khz(tmp + i, samplesOut + (i / 220) * 160,
(WebRtcSpl_State22khzTo16khz *)state2_,
tmp_mem);
}
outLen = (lengthIn * 16) / 11;
free(tmp_mem);
free(tmp);
return 0;
case kResamplerMode11To32:
// We can only handle blocks of 110 samples
if ((lengthIn % 110) != 0)
{
return -1;
}
if (maxLen < ((lengthIn * 32) / 11))
{
return -1;
}
tmp_mem = (int32_t*)malloc(104 * sizeof(int32_t));
tmp = (int16_t*)malloc((sizeof(int16_t) * lengthIn * 2));
// 11 -> 22 kHz in samplesOut
WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut, (int32_t*)state1_);
// 22 -> 16 in tmp
for (size_t i = 0; i < (lengthIn * 2); i += 220)
{
WebRtcSpl_Resample22khzTo16khz(samplesOut + i, tmp + (i / 220) * 160,
(WebRtcSpl_State22khzTo16khz *)state2_,
tmp_mem);
}
// 16 -> 32 in samplesOut
WebRtcSpl_UpsampleBy2(tmp, (lengthIn * 16) / 11, samplesOut,
(int32_t*)state3_);
outLen = (lengthIn * 32) / 11;
free(tmp_mem);
free(tmp);
return 0;
case kResamplerMode2To1:
if (maxLen < (lengthIn / 2))
{
return -1;
}
WebRtcSpl_DownsampleBy2(samplesIn, lengthIn, samplesOut, (int32_t*)state1_);
outLen = lengthIn / 2;
return 0;
case kResamplerMode3To1:
// We can only handle blocks of 480 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 480) != 0)
{
return -1;
}
if (maxLen < (lengthIn / 3))
{
return -1;
}
tmp_mem = (int32_t*)malloc(496 * sizeof(int32_t));
for (size_t i = 0; i < lengthIn; i += 480)
{
WebRtcSpl_Resample48khzTo16khz(samplesIn + i, samplesOut + i / 3,
(WebRtcSpl_State48khzTo16khz *)state1_,
tmp_mem);
}
outLen = lengthIn / 3;
free(tmp_mem);
return 0;
case kResamplerMode4To1:
if (maxLen < (lengthIn / 4))
{
return -1;
}
tmp = (int16_t*)malloc(sizeof(int16_t) * lengthIn / 2);
// 4:2
WebRtcSpl_DownsampleBy2(samplesIn, lengthIn, tmp, (int32_t*)state1_);
// 2:1
WebRtcSpl_DownsampleBy2(tmp, lengthIn / 2, samplesOut, (int32_t*)state2_);
outLen = lengthIn / 4;
free(tmp);
return 0;
case kResamplerMode6To1:
// We can only handle blocks of 480 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 480) != 0)
{
return -1;
}
if (maxLen < (lengthIn / 6))
{
return -1;
}
tmp_mem = (int32_t*)malloc(496 * sizeof(int32_t));
tmp = (int16_t*)malloc((sizeof(int16_t) * lengthIn) / 3);
for (size_t i = 0; i < lengthIn; i += 480)
{
WebRtcSpl_Resample48khzTo16khz(samplesIn + i, tmp + i / 3,
(WebRtcSpl_State48khzTo16khz *)state1_,
tmp_mem);
}
outLen = lengthIn / 3;
free(tmp_mem);
WebRtcSpl_DownsampleBy2(tmp, outLen, samplesOut, (int32_t*)state2_);
free(tmp);
outLen = outLen / 2;
return 0;
case kResamplerMode12To1:
// We can only handle blocks of 480 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 480) != 0) {
return -1;
}
if (maxLen < (lengthIn / 12)) {
return -1;
}
tmp_mem = (int32_t*) malloc(496 * sizeof(int32_t));
tmp = (int16_t*) malloc((sizeof(int16_t) * lengthIn) / 3);
tmp_2 = (int16_t*) malloc((sizeof(int16_t) * lengthIn) / 6);
// 12:4
for (size_t i = 0; i < lengthIn; i += 480) {
// WebRtcSpl_Resample48khzTo16khz() takes a block of 480 samples
// as input and outputs a resampled block of 160 samples. The
// data is now actually in 96 kHz sampling rate, despite the
// function name, and with a resampling factor of 1/3 becomes
// 32 kHz.
WebRtcSpl_Resample48khzTo16khz(samplesIn + i, tmp + i / 3,
(WebRtcSpl_State48khzTo16khz*) state1_,
tmp_mem);
}
outLen = lengthIn / 3;
free(tmp_mem);
// 4:2
WebRtcSpl_DownsampleBy2(tmp, outLen, tmp_2, (int32_t*) state2_);
outLen = outLen / 2;
free(tmp);
// 2:1
WebRtcSpl_DownsampleBy2(tmp_2, outLen, samplesOut,
(int32_t*) state3_);
free(tmp_2);
outLen = outLen / 2;
return 0;
case kResamplerMode3To2:
if (maxLen < (lengthIn * 2 / 3))
{
return -1;
}
// 3:6
tmp = static_cast<int16_t*> (malloc(sizeof(int16_t) * lengthIn * 2));
WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (int32_t*)state1_);
lengthIn *= 2;
// 6:2
// We can only handle blocks of 480 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 480) != 0)
{
free(tmp);
return -1;
}
tmp_mem = (int32_t*)malloc(496 * sizeof(int32_t));
for (size_t i = 0; i < lengthIn; i += 480)
{
WebRtcSpl_Resample48khzTo16khz(tmp + i, samplesOut + i / 3,
(WebRtcSpl_State48khzTo16khz *)state2_,
tmp_mem);
}
outLen = lengthIn / 3;
free(tmp);
free(tmp_mem);
return 0;
case kResamplerMode11To2:
// We can only handle blocks of 220 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 220) != 0)
{
return -1;
}
if (maxLen < ((lengthIn * 2) / 11))
{
return -1;
}
tmp_mem = (int32_t*)malloc(126 * sizeof(int32_t));
tmp = (int16_t*)malloc((lengthIn * 4) / 11 * sizeof(int16_t));
for (size_t i = 0; i < lengthIn; i += 220)
{
WebRtcSpl_Resample22khzTo8khz(samplesIn + i, tmp + (i * 4) / 11,
(WebRtcSpl_State22khzTo8khz *)state1_,
tmp_mem);
}
lengthIn = (lengthIn * 4) / 11;
WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut,
(int32_t*)state2_);
outLen = lengthIn / 2;
free(tmp_mem);
free(tmp);
return 0;
case kResamplerMode11To4:
// We can only handle blocks of 220 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 220) != 0)
{
return -1;
}
if (maxLen < ((lengthIn * 4) / 11))
{
return -1;
}
tmp_mem = (int32_t*)malloc(126 * sizeof(int32_t));
for (size_t i = 0; i < lengthIn; i += 220)
{
WebRtcSpl_Resample22khzTo8khz(samplesIn + i, samplesOut + (i * 4) / 11,
(WebRtcSpl_State22khzTo8khz *)state1_,
tmp_mem);
}
outLen = (lengthIn * 4) / 11;
free(tmp_mem);
return 0;
case kResamplerMode11To8:
// We can only handle blocks of 160 samples
// Can be fixed, but I don't think it's needed
if ((lengthIn % 220) != 0)
{
return -1;
}
if (maxLen < ((lengthIn * 8) / 11))
{
return -1;
}
tmp_mem = (int32_t*)malloc(104 * sizeof(int32_t));
for (size_t i = 0; i < lengthIn; i += 220)
{
WebRtcSpl_Resample22khzTo16khz(samplesIn + i, samplesOut + (i * 8) / 11,
(WebRtcSpl_State22khzTo16khz *)state1_,
tmp_mem);
}
outLen = (lengthIn * 8) / 11;
free(tmp_mem);
return 0;
break;
}
return 0;
}
} // namespace webrtc

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webrtc/common_audio/resampler/sinc_resampler.cc Normal file
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@ -0,0 +1,378 @@
/*
* 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.
*/
// Modified from the Chromium original:
// src/media/base/sinc_resampler.cc
// Initial input buffer layout, dividing into regions r0_ to r4_ (note: r0_, r3_
// and r4_ will move after the first load):
//
// |----------------|-----------------------------------------|----------------|
//
// request_frames_
// <--------------------------------------------------------->
// r0_ (during first load)
//
// kKernelSize / 2 kKernelSize / 2 kKernelSize / 2 kKernelSize / 2
// <---------------> <---------------> <---------------> <--------------->
// r1_ r2_ r3_ r4_
//
// block_size_ == r4_ - r2_
// <--------------------------------------->
//
// request_frames_
// <------------------ ... ----------------->
// r0_ (during second load)
//
// On the second request r0_ slides to the right by kKernelSize / 2 and r3_, r4_
// and block_size_ are reinitialized via step (3) in the algorithm below.
//
// These new regions remain constant until a Flush() occurs. While complicated,
// this allows us to reduce jitter by always requesting the same amount from the
// provided callback.
//
// The algorithm:
//
// 1) Allocate input_buffer of size: request_frames_ + kKernelSize; this ensures
// there's enough room to read request_frames_ from the callback into region
// r0_ (which will move between the first and subsequent passes).
//
// 2) Let r1_, r2_ each represent half the kernel centered around r0_:
//
// r0_ = input_buffer_ + kKernelSize / 2
// r1_ = input_buffer_
// r2_ = r0_
//
// r0_ is always request_frames_ in size. r1_, r2_ are kKernelSize / 2 in
// size. r1_ must be zero initialized to avoid convolution with garbage (see
// step (5) for why).
//
// 3) Let r3_, r4_ each represent half the kernel right aligned with the end of
// r0_ and choose block_size_ as the distance in frames between r4_ and r2_:
//
// r3_ = r0_ + request_frames_ - kKernelSize
// r4_ = r0_ + request_frames_ - kKernelSize / 2
// block_size_ = r4_ - r2_ = request_frames_ - kKernelSize / 2
//
// 4) Consume request_frames_ frames into r0_.
//
// 5) Position kernel centered at start of r2_ and generate output frames until
// the kernel is centered at the start of r4_ or we've finished generating
// all the output frames.
//
// 6) Wrap left over data from the r3_ to r1_ and r4_ to r2_.
//
// 7) If we're on the second load, in order to avoid overwriting the frames we
// just wrapped from r4_ we need to slide r0_ to the right by the size of
// r4_, which is kKernelSize / 2:
//
// r0_ = r0_ + kKernelSize / 2 = input_buffer_ + kKernelSize
//
// r3_, r4_, and block_size_ then need to be reinitialized, so goto (3).
//
// 8) Else, if we're not on the second load, goto (4).
//
// Note: we're glossing over how the sub-sample handling works with
// |virtual_source_idx_|, etc.
// MSVC++ requires this to be set before any other includes to get M_PI.
#define _USE_MATH_DEFINES
#include "webrtc/common_audio/resampler/sinc_resampler.h"
#include <assert.h>
#include <math.h>
#include <string.h>
#include <limits>
#include "webrtc/system_wrappers/interface/cpu_features_wrapper.h"
#include "webrtc/typedefs.h"
namespace webrtc {
namespace {
double SincScaleFactor(double io_ratio) {
// |sinc_scale_factor| is basically the normalized cutoff frequency of the
// low-pass filter.
double sinc_scale_factor = io_ratio > 1.0 ? 1.0 / io_ratio : 1.0;
// The sinc function is an idealized brick-wall filter, but since we're
// windowing it the transition from pass to stop does not happen right away.
// So we should adjust the low pass filter cutoff slightly downward to avoid
// some aliasing at the very high-end.
// TODO(crogers): this value is empirical and to be more exact should vary
// depending on kKernelSize.
sinc_scale_factor *= 0.9;
return sinc_scale_factor;
}
} // namespace
// If we know the minimum architecture at compile time, avoid CPU detection.
#if defined(WEBRTC_ARCH_X86_FAMILY)
#if defined(__SSE2__)
#define CONVOLVE_FUNC Convolve_SSE
void SincResampler::InitializeCPUSpecificFeatures() {}
#else
// x86 CPU detection required. Function will be set by
// InitializeCPUSpecificFeatures().
// TODO(dalecurtis): Once Chrome moves to an SSE baseline this can be removed.
#define CONVOLVE_FUNC convolve_proc_
void SincResampler::InitializeCPUSpecificFeatures() {
convolve_proc_ = WebRtc_GetCPUInfo(kSSE2) ? Convolve_SSE : Convolve_C;
}
#endif
#elif defined(WEBRTC_HAS_NEON)
#define CONVOLVE_FUNC Convolve_NEON
void SincResampler::InitializeCPUSpecificFeatures() {}
#elif defined(WEBRTC_DETECT_NEON)
#define CONVOLVE_FUNC convolve_proc_
void SincResampler::InitializeCPUSpecificFeatures() {
convolve_proc_ = WebRtc_GetCPUFeaturesARM() & kCPUFeatureNEON ?
Convolve_NEON : Convolve_C;
}
#else
// Unknown architecture.
#define CONVOLVE_FUNC Convolve_C
void SincResampler::InitializeCPUSpecificFeatures() {}
#endif
SincResampler::SincResampler(double io_sample_rate_ratio,
size_t request_frames,
SincResamplerCallback* read_cb)
: io_sample_rate_ratio_(io_sample_rate_ratio),
read_cb_(read_cb),
request_frames_(request_frames),
input_buffer_size_(request_frames_ + kKernelSize),
// Create input buffers with a 16-byte alignment for SSE optimizations.
kernel_storage_(static_cast<float*>(
AlignedMalloc(sizeof(float) * kKernelStorageSize, 16))),
kernel_pre_sinc_storage_(static_cast<float*>(
AlignedMalloc(sizeof(float) * kKernelStorageSize, 16))),
kernel_window_storage_(static_cast<float*>(
AlignedMalloc(sizeof(float) * kKernelStorageSize, 16))),
input_buffer_(static_cast<float*>(
AlignedMalloc(sizeof(float) * input_buffer_size_, 16))),
#if defined(WEBRTC_CPU_DETECTION)
convolve_proc_(NULL),
#endif
r1_(input_buffer_.get()),
r2_(input_buffer_.get() + kKernelSize / 2) {
#if defined(WEBRTC_CPU_DETECTION)
InitializeCPUSpecificFeatures();
assert(convolve_proc_);
#endif
assert(request_frames_ > 0);
Flush();
assert(block_size_ > kKernelSize);
memset(kernel_storage_.get(), 0,
sizeof(*kernel_storage_.get()) * kKernelStorageSize);
memset(kernel_pre_sinc_storage_.get(), 0,
sizeof(*kernel_pre_sinc_storage_.get()) * kKernelStorageSize);
memset(kernel_window_storage_.get(), 0,
sizeof(*kernel_window_storage_.get()) * kKernelStorageSize);
InitializeKernel();
}
SincResampler::~SincResampler() {}
void SincResampler::UpdateRegions(bool second_load) {
// Setup various region pointers in the buffer (see diagram above). If we're
// on the second load we need to slide r0_ to the right by kKernelSize / 2.
r0_ = input_buffer_.get() + (second_load ? kKernelSize : kKernelSize / 2);
r3_ = r0_ + request_frames_ - kKernelSize;
r4_ = r0_ + request_frames_ - kKernelSize / 2;
block_size_ = r4_ - r2_;
// r1_ at the beginning of the buffer.
assert(r1_ == input_buffer_.get());
// r1_ left of r2_, r4_ left of r3_ and size correct.
assert(r2_ - r1_ == r4_ - r3_);
// r2_ left of r3.
assert(r2_ < r3_);
}
void SincResampler::InitializeKernel() {
// Blackman window parameters.
static const double kAlpha = 0.16;
static const double kA0 = 0.5 * (1.0 - kAlpha);
static const double kA1 = 0.5;
static const double kA2 = 0.5 * kAlpha;
// Generates a set of windowed sinc() kernels.
// We generate a range of sub-sample offsets from 0.0 to 1.0.
const double sinc_scale_factor = SincScaleFactor(io_sample_rate_ratio_);
for (size_t offset_idx = 0; offset_idx <= kKernelOffsetCount; ++offset_idx) {
const float subsample_offset =
static_cast<float>(offset_idx) / kKernelOffsetCount;
for (size_t i = 0; i < kKernelSize; ++i) {
const size_t idx = i + offset_idx * kKernelSize;
const float pre_sinc = static_cast<float>(M_PI *
(static_cast<int>(i) - static_cast<int>(kKernelSize / 2) -
subsample_offset));
kernel_pre_sinc_storage_[idx] = pre_sinc;
// Compute Blackman window, matching the offset of the sinc().
const float x = (i - subsample_offset) / kKernelSize;
const float window = static_cast<float>(kA0 - kA1 * cos(2.0 * M_PI * x) +
kA2 * cos(4.0 * M_PI * x));
kernel_window_storage_[idx] = window;
// Compute the sinc with offset, then window the sinc() function and store
// at the correct offset.
kernel_storage_[idx] = static_cast<float>(window *
((pre_sinc == 0) ?
sinc_scale_factor :
(sin(sinc_scale_factor * pre_sinc) / pre_sinc)));
}
}
}
void SincResampler::SetRatio(double io_sample_rate_ratio) {
if (fabs(io_sample_rate_ratio_ - io_sample_rate_ratio) <
std::numeric_limits<double>::epsilon()) {
return;
}
io_sample_rate_ratio_ = io_sample_rate_ratio;
// Optimize reinitialization by reusing values which are independent of
// |sinc_scale_factor|. Provides a 3x speedup.
const double sinc_scale_factor = SincScaleFactor(io_sample_rate_ratio_);
for (size_t offset_idx = 0; offset_idx <= kKernelOffsetCount; ++offset_idx) {
for (size_t i = 0; i < kKernelSize; ++i) {
const size_t idx = i + offset_idx * kKernelSize;
const float window = kernel_window_storage_[idx];
const float pre_sinc = kernel_pre_sinc_storage_[idx];
kernel_storage_[idx] = static_cast<float>(window *
((pre_sinc == 0) ?
sinc_scale_factor :
(sin(sinc_scale_factor * pre_sinc) / pre_sinc)));
}
}
}
void SincResampler::Resample(size_t frames, float* destination) {
size_t remaining_frames = frames;
// Step (1) -- Prime the input buffer at the start of the input stream.
if (!buffer_primed_ && remaining_frames) {
read_cb_->Run(request_frames_, r0_);
buffer_primed_ = true;
}
// Step (2) -- Resample! const what we can outside of the loop for speed. It
// actually has an impact on ARM performance. See inner loop comment below.
const double current_io_ratio = io_sample_rate_ratio_;
const float* const kernel_ptr = kernel_storage_.get();
while (remaining_frames) {
// |i| may be negative if the last Resample() call ended on an iteration
// that put |virtual_source_idx_| over the limit.
//
// Note: The loop construct here can severely impact performance on ARM
// or when built with clang. See https://codereview.chromium.org/18566009/
for (int i = static_cast<int>(
ceil((block_size_ - virtual_source_idx_) / current_io_ratio));
i > 0; --i) {
assert(virtual_source_idx_ < block_size_);
// |virtual_source_idx_| lies in between two kernel offsets so figure out
// what they are.
const int source_idx = static_cast<int>(virtual_source_idx_);
const double subsample_remainder = virtual_source_idx_ - source_idx;
const double virtual_offset_idx =
subsample_remainder * kKernelOffsetCount;
const int offset_idx = static_cast<int>(virtual_offset_idx);
// We'll compute "convolutions" for the two kernels which straddle
// |virtual_source_idx_|.
const float* const k1 = kernel_ptr + offset_idx * kKernelSize;
const float* const k2 = k1 + kKernelSize;
// Ensure |k1|, |k2| are 16-byte aligned for SIMD usage. Should always be
// true so long as kKernelSize is a multiple of 16.
assert(0u == (reinterpret_cast<uintptr_t>(k1) & 0x0F));
assert(0u == (reinterpret_cast<uintptr_t>(k2) & 0x0F));
// Initialize input pointer based on quantized |virtual_source_idx_|.
const float* const input_ptr = r1_ + source_idx;
// Figure out how much to weight each kernel's "convolution".
const double kernel_interpolation_factor =
virtual_offset_idx - offset_idx;
*destination++ = CONVOLVE_FUNC(
input_ptr, k1, k2, kernel_interpolation_factor);
// Advance the virtual index.
virtual_source_idx_ += current_io_ratio;
if (!--remaining_frames)
return;
}
// Wrap back around to the start.
virtual_source_idx_ -= block_size_;
// Step (3) -- Copy r3_, r4_ to r1_, r2_.
// This wraps the last input frames back to the start of the buffer.
memcpy(r1_, r3_, sizeof(*input_buffer_.get()) * kKernelSize);
// Step (4) -- Reinitialize regions if necessary.
if (r0_ == r2_)
UpdateRegions(true);
// Step (5) -- Refresh the buffer with more input.
read_cb_->Run(request_frames_, r0_);
}
}
#undef CONVOLVE_FUNC
size_t SincResampler::ChunkSize() const {
return static_cast<size_t>(block_size_ / io_sample_rate_ratio_);
}
void SincResampler::Flush() {
virtual_source_idx_ = 0;
buffer_primed_ = false;
memset(input_buffer_.get(), 0,
sizeof(*input_buffer_.get()) * input_buffer_size_);
UpdateRegions(false);
}
float SincResampler::Convolve_C(const float* input_ptr, const float* k1,
const float* k2,
double kernel_interpolation_factor) {
float sum1 = 0;
float sum2 = 0;
// Generate a single output sample. Unrolling this loop hurt performance in
// local testing.
size_t n = kKernelSize;
while (n--) {
sum1 += *input_ptr * *k1++;
sum2 += *input_ptr++ * *k2++;
}
// Linearly interpolate the two "convolutions".
return static_cast<float>((1.0 - kernel_interpolation_factor) * sum1 +
kernel_interpolation_factor * sum2);
}
} // namespace webrtc

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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.
*/
// Modified from the Chromium original here:
// src/media/base/sinc_resampler.h
#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
#define WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/system_wrappers/interface/aligned_malloc.h"
#include "webrtc/test/testsupport/gtest_prod_util.h"
#include "webrtc/typedefs.h"
namespace webrtc {
// Callback class for providing more data into the resampler. Expects |frames|
// of data to be rendered into |destination|; zero padded if not enough frames
// are available to satisfy the request.
class SincResamplerCallback {
public:
virtual ~SincResamplerCallback() {}
virtual void Run(size_t frames, float* destination) = 0;
};
// SincResampler is a high-quality single-channel sample-rate converter.
class SincResampler {
public:
// The kernel size can be adjusted for quality (higher is better) at the
// expense of performance. Must be a multiple of 32.
// TODO(dalecurtis): Test performance to see if we can jack this up to 64+.
static const size_t kKernelSize = 32;
// Default request size. Affects how often and for how much SincResampler
// calls back for input. Must be greater than kKernelSize.
static const size_t kDefaultRequestSize = 512;
// The kernel offset count is used for interpolation and is the number of
// sub-sample kernel shifts. Can be adjusted for quality (higher is better)
// at the expense of allocating more memory.
static const size_t kKernelOffsetCount = 32;
static const size_t kKernelStorageSize =
kKernelSize * (kKernelOffsetCount + 1);
// Constructs a SincResampler with the specified |read_cb|, which is used to
// acquire audio data for resampling. |io_sample_rate_ratio| is the ratio
// of input / output sample rates. |request_frames| controls the size in
// frames of the buffer requested by each |read_cb| call. The value must be
// greater than kKernelSize. Specify kDefaultRequestSize if there are no
// request size constraints.
SincResampler(double io_sample_rate_ratio,
size_t request_frames,
SincResamplerCallback* read_cb);
virtual ~SincResampler();
// Resample |frames| of data from |read_cb_| into |destination|.
void Resample(size_t frames, float* destination);
// The maximum size in frames that guarantees Resample() will only make a
// single call to |read_cb_| for more data.
size_t ChunkSize() const;
size_t request_frames() const { return request_frames_; }
// Flush all buffered data and reset internal indices. Not thread safe, do
// not call while Resample() is in progress.
void Flush();
// Update |io_sample_rate_ratio_|. SetRatio() will cause a reconstruction of
// the kernels used for resampling. Not thread safe, do not call while
// Resample() is in progress.
//
// TODO(ajm): Use this in PushSincResampler rather than reconstructing
// SincResampler. We would also need a way to update |request_frames_|.
void SetRatio(double io_sample_rate_ratio);
float* get_kernel_for_testing() { return kernel_storage_.get(); }
private:
FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, Convolve);
FRIEND_TEST_ALL_PREFIXES(SincResamplerTest, ConvolveBenchmark);
void InitializeKernel();
void UpdateRegions(bool second_load);
// Selects runtime specific CPU features like SSE. Must be called before
// using SincResampler.
// TODO(ajm): Currently managed by the class internally. See the note with
// |convolve_proc_| below.
void InitializeCPUSpecificFeatures();
// Compute convolution of |k1| and |k2| over |input_ptr|, resultant sums are
// linearly interpolated using |kernel_interpolation_factor|. On x86 and ARM
// the underlying implementation is chosen at run time.
static float Convolve_C(const float* input_ptr, const float* k1,
const float* k2, double kernel_interpolation_factor);
#if defined(WEBRTC_ARCH_X86_FAMILY)
static float Convolve_SSE(const float* input_ptr, const float* k1,
const float* k2,
double kernel_interpolation_factor);
#elif defined(WEBRTC_DETECT_NEON) || defined(WEBRTC_HAS_NEON)
static float Convolve_NEON(const float* input_ptr, const float* k1,
const float* k2,
double kernel_interpolation_factor);
#endif
// The ratio of input / output sample rates.
double io_sample_rate_ratio_;
// An index on the source input buffer with sub-sample precision. It must be
// double precision to avoid drift.
double virtual_source_idx_;
// The buffer is primed once at the very beginning of processing.
bool buffer_primed_;
// Source of data for resampling.
SincResamplerCallback* read_cb_;
// The size (in samples) to request from each |read_cb_| execution.
const size_t request_frames_;
// The number of source frames processed per pass.
size_t block_size_;
// The size (in samples) of the internal buffer used by the resampler.
const size_t input_buffer_size_;
// Contains kKernelOffsetCount kernels back-to-back, each of size kKernelSize.
// The kernel offsets are sub-sample shifts of a windowed sinc shifted from
// 0.0 to 1.0 sample.
rtc::scoped_ptr<float[], AlignedFreeDeleter> kernel_storage_;
rtc::scoped_ptr<float[], AlignedFreeDeleter> kernel_pre_sinc_storage_;
rtc::scoped_ptr<float[], AlignedFreeDeleter> kernel_window_storage_;
// Data from the source is copied into this buffer for each processing pass.
rtc::scoped_ptr<float[], AlignedFreeDeleter> input_buffer_;
// Stores the runtime selection of which Convolve function to use.
// TODO(ajm): Move to using a global static which must only be initialized
// once by the user. We're not doing this initially, because we don't have
// e.g. a LazyInstance helper in webrtc.
#if defined(WEBRTC_CPU_DETECTION)
typedef float (*ConvolveProc)(const float*, const float*, const float*,
double);
ConvolveProc convolve_proc_;
#endif
// Pointers to the various regions inside |input_buffer_|. See the diagram at
// the top of the .cc file for more information.
float* r0_;
float* const r1_;
float* const r2_;
float* r3_;
float* r4_;
RTC_DISALLOW_COPY_AND_ASSIGN(SincResampler);
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_SINC_RESAMPLER_H_

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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.
*/
// Modified from the Chromium original:
// src/media/base/sinc_resampler.cc
#include "webrtc/common_audio/resampler/sinc_resampler.h"
#include <arm_neon.h>
namespace webrtc {
float SincResampler::Convolve_NEON(const float* input_ptr, const float* k1,
const float* k2,
double kernel_interpolation_factor) {
float32x4_t m_input;
float32x4_t m_sums1 = vmovq_n_f32(0);
float32x4_t m_sums2 = vmovq_n_f32(0);
const float* upper = input_ptr + kKernelSize;
for (; input_ptr < upper; ) {
m_input = vld1q_f32(input_ptr);
input_ptr += 4;
m_sums1 = vmlaq_f32(m_sums1, m_input, vld1q_f32(k1));
k1 += 4;
m_sums2 = vmlaq_f32(m_sums2, m_input, vld1q_f32(k2));
k2 += 4;
}
// Linearly interpolate the two "convolutions".
m_sums1 = vmlaq_f32(
vmulq_f32(m_sums1, vmovq_n_f32(1.0 - kernel_interpolation_factor)),
m_sums2, vmovq_n_f32(kernel_interpolation_factor));
// Sum components together.
float32x2_t m_half = vadd_f32(vget_high_f32(m_sums1), vget_low_f32(m_sums1));
return vget_lane_f32(vpadd_f32(m_half, m_half), 0);
}
} // namespace webrtc

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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.
*/
// Modified from the Chromium original:
// src/media/base/simd/sinc_resampler_sse.cc
#include "webrtc/common_audio/resampler/sinc_resampler.h"
#include <xmmintrin.h>
namespace webrtc {
float SincResampler::Convolve_SSE(const float* input_ptr, const float* k1,
const float* k2,
double kernel_interpolation_factor) {
__m128 m_input;
__m128 m_sums1 = _mm_setzero_ps();
__m128 m_sums2 = _mm_setzero_ps();
// Based on |input_ptr| alignment, we need to use loadu or load. Unrolling
// these loops hurt performance in local testing.
if (reinterpret_cast<uintptr_t>(input_ptr) & 0x0F) {
for (size_t i = 0; i < kKernelSize; i += 4) {
m_input = _mm_loadu_ps(input_ptr + i);
m_sums1 = _mm_add_ps(m_sums1, _mm_mul_ps(m_input, _mm_load_ps(k1 + i)));
m_sums2 = _mm_add_ps(m_sums2, _mm_mul_ps(m_input, _mm_load_ps(k2 + i)));
}
} else {
for (size_t i = 0; i < kKernelSize; i += 4) {
m_input = _mm_load_ps(input_ptr + i);
m_sums1 = _mm_add_ps(m_sums1, _mm_mul_ps(m_input, _mm_load_ps(k1 + i)));
m_sums2 = _mm_add_ps(m_sums2, _mm_mul_ps(m_input, _mm_load_ps(k2 + i)));
}
}
// Linearly interpolate the two "convolutions".
m_sums1 = _mm_mul_ps(m_sums1, _mm_set_ps1(
static_cast<float>(1.0 - kernel_interpolation_factor)));
m_sums2 = _mm_mul_ps(m_sums2, _mm_set_ps1(
static_cast<float>(kernel_interpolation_factor)));
m_sums1 = _mm_add_ps(m_sums1, m_sums2);
// Sum components together.
float result;
m_sums2 = _mm_add_ps(_mm_movehl_ps(m_sums1, m_sums1), m_sums1);
_mm_store_ss(&result, _mm_add_ss(m_sums2, _mm_shuffle_ps(
m_sums2, m_sums2, 1)));
return result;
}
} // namespace webrtc

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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.
*/
// MSVC++ requires this to be set before any other includes to get M_PI.
#define _USE_MATH_DEFINES
#include "webrtc/common_audio/resampler/sinusoidal_linear_chirp_source.h"
#include <math.h>
namespace webrtc {
SinusoidalLinearChirpSource::SinusoidalLinearChirpSource(int sample_rate,
size_t samples,
double max_frequency,
double delay_samples)
: sample_rate_(sample_rate),
total_samples_(samples),
max_frequency_(max_frequency),
current_index_(0),
delay_samples_(delay_samples) {
// Chirp rate.
double duration = static_cast<double>(total_samples_) / sample_rate_;
k_ = (max_frequency_ - kMinFrequency) / duration;
}
void SinusoidalLinearChirpSource::Run(size_t frames, float* destination) {
for (size_t i = 0; i < frames; ++i, ++current_index_) {
// Filter out frequencies higher than Nyquist.
if (Frequency(current_index_) > 0.5 * sample_rate_) {
destination[i] = 0;
} else {
// Calculate time in seconds.
if (current_index_ < delay_samples_) {
destination[i] = 0;
} else {
// Sinusoidal linear chirp.
double t = (current_index_ - delay_samples_) / sample_rate_;
destination[i] =
sin(2 * M_PI * (kMinFrequency * t + (k_ / 2) * t * t));
}
}
}
}
double SinusoidalLinearChirpSource::Frequency(size_t position) {
return kMinFrequency + (position - delay_samples_) *
(max_frequency_ - kMinFrequency) / total_samples_;
}
} // namespace webrtc

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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.
*/
// Modified from the Chromium original here:
// src/media/base/sinc_resampler_unittest.cc
#ifndef WEBRTC_COMMON_AUDIO_RESAMPLER_SINUSOIDAL_LINEAR_CHIRP_SOURCE_H_
#define WEBRTC_COMMON_AUDIO_RESAMPLER_SINUSOIDAL_LINEAR_CHIRP_SOURCE_H_
#include "webrtc/base/constructormagic.h"
#include "webrtc/common_audio/resampler/sinc_resampler.h"
namespace webrtc {
// Fake audio source for testing the resampler. Generates a sinusoidal linear
// chirp (http://en.wikipedia.org/wiki/Chirp) which can be tuned to stress the
// resampler for the specific sample rate conversion being used.
class SinusoidalLinearChirpSource : public SincResamplerCallback {
public:
// |delay_samples| can be used to insert a fractional sample delay into the
// source. It will produce zeros until non-negative time is reached.
SinusoidalLinearChirpSource(int sample_rate, size_t samples,
double max_frequency, double delay_samples);
virtual ~SinusoidalLinearChirpSource() {}
void Run(size_t frames, float* destination) override;
double Frequency(size_t position);
private:
enum {
kMinFrequency = 5
};
int sample_rate_;
size_t total_samples_;
double max_frequency_;
double k_;
size_t current_index_;
double delay_samples_;
RTC_DISALLOW_COPY_AND_ASSIGN(SinusoidalLinearChirpSource);
};
} // namespace webrtc
#endif // WEBRTC_COMMON_AUDIO_RESAMPLER_SINUSOIDAL_LINEAR_CHIRP_SOURCE_H_