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/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*-*/
/* This Source Code Form is subject to the terms of the Mozilla Public
* License, v. 2.0. If a copy of the MPL was not distributed with this file,
#ifndef DOM_MEDIA_DRIFTCONTROL_DYNAMICRESAMPLER_H_
#define DOM_MEDIA_DRIFTCONTROL_DYNAMICRESAMPLER_H_
#include "AudioRingBuffer.h"
#include "AudioSegment.h"
#include "TimeUnits.h"
#include "WavDumper.h"
#include <speex/speex_resampler.h>
namespace mozilla {
const uint32_t STEREO = 2;
/**
* DynamicResampler allows updating on the fly the output sample rate and the
* number of channels. In addition to that, it maintains an internal buffer for
* the input data and allows pre-buffering as well. The Resample() method
* strives to provide the requested number of output frames by using the input
* data including any pre-buffering. If there are fewer frames in the internal
* buffer than is requested, the internal buffer is padded with enough silence
* to allow the requested to be resampled and returned.
*
* Input data buffering makes use of the AudioRingBuffer. The capacity of the
* buffer is initially 100ms of audio and it is pre-allocated during
* SetSampleFormat(). Should the input data grow beyond that, the input buffer
* is re-allocated on the fly. In addition to that, due to special feature of
* AudioRingBuffer, no extra copies take place when the input data is fed to the
* resampler.
*
* The sample format must be set before using any method.
*
* The DynamicResampler is not thread-safe, so all the methods appart from the
* constructor must be called on the same thread.
*/
class DynamicResampler final {
public:
/**
* Provide the initial input and output rate and the amount of pre-buffering.
* The channel count will be set to stereo. Memory allocation will take
* place. The input buffer is non-interleaved.
*/
DynamicResampler(uint32_t aInRate, uint32_t aOutRate,
uint32_t aInputPreBufferFrameCount = 0);
~DynamicResampler();
/**
* Set the sample format type to float or short.
*/
void SetSampleFormat(AudioSampleFormat aFormat);
uint32_t GetInRate() const { return mInRate; }
uint32_t GetChannels() const { return mChannels; }
/**
* Append `aInFrames` number of frames from `aInBuffer` to the internal input
* buffer. Memory copy/move takes place.
*/
void AppendInput(Span<const float* const> aInBuffer, uint32_t aInFrames);
void AppendInput(Span<const int16_t* const> aInBuffer, uint32_t aInFrames);
/**
* Append `aInFrames` number of frames of silence to the internal input
* buffer. Memory copy/move takes place.
*/
void AppendInputSilence(const uint32_t aInFrames);
/**
* Return the number of frames the internal input buffer can store.
*/
uint32_t InFramesBufferSize() const;
/**
* Return the number of frames stored in the internal input buffer.
*/
uint32_t InFramesBuffered(uint32_t aChannelIndex) const;
/**
* Prepends existing input data with a silent pre-buffer if not already done.
* Data will be prepended so that after resampling aDuration of data,
* the buffering level will be as close as possible to
* mInputPreBufferFrameCount, which is the desired buffering level.
*/
void EnsurePreBuffer(media::TimeUnit aDuration);
/**
* Set the number of frames that should be used for input pre-buffering.
*/
void SetInputPreBufferFrameCount(uint32_t aInputPreBufferFrameCount);
/*
* Resample as much frames as needed from the internal input buffer to the
* `aOutBuffer` in order to provide all `aOutFrames`.
*
* On first call, prepends the input buffer with silence so that after
* resampling aOutFrames frames of data, the input buffer holds data as close
* as possible to the configured pre-buffer size.
*
* If there are not enough input frames to provide the requested output
* frames, the input buffer is padded with enough silence to allow the
* requested frames to be resampled, and the pre-buffer is reset so that the
* next call will be treated as the first.
*
* Returns true if the internal input buffer underran and had to be padded
* with silence, otherwise false.
*/
bool Resample(float* aOutBuffer, uint32_t aOutFrames, uint32_t aChannelIndex);
bool Resample(int16_t* aOutBuffer, uint32_t aOutFrames,
uint32_t aChannelIndex);
/**
* Update the output rate or/and the channel count. If a value is not updated
* compared to the current one nothing happens. Changing the `aInRate`
* results in recalculation in the resampler. Changing `aChannels` results in
* the reallocation of the internal input buffer with the exception of
* changes between mono to stereo and vice versa where no reallocation takes
* place. A stereo internal input buffer is always maintained even if the
* sound is mono.
*/
void UpdateResampler(uint32_t aInRate, uint32_t aChannels);
private:
template <typename T>
void AppendInputInternal(Span<const T* const>& aInBuffer,
uint32_t aInFrames) {
MOZ_ASSERT(aInBuffer.Length() == (uint32_t)mChannels);
for (uint32_t i = 0; i < mChannels; ++i) {
PushInFrames(aInBuffer[i], aInFrames, i);
}
}
void ResampleInternal(const float* aInBuffer, uint32_t* aInFrames,
float* aOutBuffer, uint32_t* aOutFrames,
uint32_t aChannelIndex);
void ResampleInternal(const int16_t* aInBuffer, uint32_t* aInFrames,
int16_t* aOutBuffer, uint32_t* aOutFrames,
uint32_t aChannelIndex);
template <typename T>
bool ResampleInternal(T* aOutBuffer, uint32_t aOutFrames,
uint32_t aChannelIndex) {
MOZ_ASSERT(mInRate);
MOZ_ASSERT(mOutRate);
MOZ_ASSERT(mChannels);
MOZ_ASSERT(aChannelIndex < mChannels);
MOZ_ASSERT(aChannelIndex < mInternalInBuffer.Length());
MOZ_ASSERT(aOutFrames);
if (mInRate == mOutRate) {
bool underrun = false;
if (uint32_t buffered = mInternalInBuffer[aChannelIndex].AvailableRead();
buffered < aOutFrames) {
underrun = true;
mIsPreBufferSet = false;
mInternalInBuffer[aChannelIndex].WriteSilence(aOutFrames - buffered);
}
DebugOnly<uint32_t> numFramesRead =
mInternalInBuffer[aChannelIndex].Read(Span(aOutBuffer, aOutFrames));
MOZ_ASSERT(numFramesRead == aOutFrames);
// Workaround to avoid discontinuity when the speex resampler operates
// again. Feed it with the last 20 frames to warm up the internal memory
// of the resampler and then skip memory equals to resampler's input
// latency.
mInputTail[aChannelIndex].StoreTail<T>(aOutBuffer, aOutFrames);
if (aChannelIndex == 0 && !mIsWarmingUp) {
mInputStreamFile.Write(aOutBuffer, aOutFrames);
mOutputStreamFile.Write(aOutBuffer, aOutFrames);
}
return underrun;
}
uint32_t totalOutFramesNeeded = aOutFrames;
auto resample = [&](const T* aInBuffer, uint32_t aInLength) -> uint32_t {
uint32_t outFramesResampled = totalOutFramesNeeded;
uint32_t inFrames = aInLength;
ResampleInternal(aInBuffer, &inFrames, aOutBuffer, &outFramesResampled,
aChannelIndex);
aOutBuffer += outFramesResampled;
totalOutFramesNeeded -= outFramesResampled;
mInputTail[aChannelIndex].StoreTail<T>(aInBuffer, inFrames);
return inFrames;
};
mInternalInBuffer[aChannelIndex].ReadNoCopy(
[&](const Span<const T>& aInBuffer) -> uint32_t {
if (!totalOutFramesNeeded) {
return 0;
}
return resample(aInBuffer.Elements(), aInBuffer.Length());
});
if (totalOutFramesNeeded == 0) {
return false;
}
while (totalOutFramesNeeded > 0) {
MOZ_ASSERT(mInternalInBuffer[aChannelIndex].AvailableRead() == 0);
// Round up.
uint32_t totalInFramesNeeded =
((CheckedUint32(totalOutFramesNeeded) * mInRate + mOutRate - 1) /
mOutRate)
.value();
resample(nullptr, totalInFramesNeeded);
}
mIsPreBufferSet = false;
return true;
}
template <typename T>
void PushInFrames(const T* aInBuffer, const uint32_t aInFrames,
uint32_t aChannelIndex) {
MOZ_ASSERT(aInBuffer);
MOZ_ASSERT(aInFrames);
MOZ_ASSERT(mChannels);
MOZ_ASSERT(aChannelIndex < mChannels);
MOZ_ASSERT(aChannelIndex < mInternalInBuffer.Length());
EnsureInputBufferSizeInFrames(
mInternalInBuffer[aChannelIndex].AvailableRead() + aInFrames);
mInternalInBuffer[aChannelIndex].Write(Span(aInBuffer, aInFrames));
}
void WarmUpResampler(bool aSkipLatency);
bool EnsureInputBufferSizeInFrames(uint32_t aSizeInFrames) {
uint32_t sampleSize = 0;
if (mSampleFormat == AUDIO_FORMAT_FLOAT32) {
sampleSize = sizeof(float);
} else if (mSampleFormat == AUDIO_FORMAT_S16) {
sampleSize = sizeof(short);
}
if (sampleSize == 0) {
// No sample format set, we wouldn't know how many bytes to allocate.
return true;
}
uint32_t sizeInFrames = InFramesBufferSize();
if (aSizeInFrames <= sizeInFrames) {
// Buffer size is sufficient.
return true; // no reallocation necessary
}
// 5 second cap.
const uint32_t cap = 5 * mInRate;
if (sizeInFrames >= cap) {
// Already at the cap.
return false;
}
// As a backoff strategy, at least double the previous size.
sizeInFrames *= 2;
if (aSizeInFrames > sizeInFrames) {
// A larger buffer than the normal backoff strategy provides is needed, or
// this is the first time setting the buffer size. Add another 50ms, as
// some jitter is expected.
sizeInFrames = aSizeInFrames + mInRate / 20;
}
// mInputPreBufferFrameCount is an indication of the desired average
// buffering. Provide for at least twice this.
sizeInFrames = std::max(sizeInFrames, mInputPreBufferFrameCount * 2);
sizeInFrames = std::min(cap, sizeInFrames);
bool success = true;
for (auto& b : mInternalInBuffer) {
success = success && b.EnsureLengthBytes(sampleSize * sizeInFrames);
}
if (success) {
// All buffers have the new size.
return true;
}
// Allocating an input buffer failed. We stick with the old buffer size.
NS_WARNING(nsPrintfCString("Failed to allocate a buffer of %u bytes (%u "
"frames). Expect glitches.",
sampleSize * sizeInFrames, sizeInFrames)
.get());
return false;
}
public:
const uint32_t mOutRate;
private:
bool mIsPreBufferSet = false;
bool mIsWarmingUp = false;
uint32_t mInputPreBufferFrameCount;
uint32_t mChannels = 0;
uint32_t mInRate;
AutoTArray<AudioRingBuffer, STEREO> mInternalInBuffer;
SpeexResamplerState* mResampler = nullptr;
AudioSampleFormat mSampleFormat = AUDIO_FORMAT_SILENCE;
class TailBuffer {
public:
template <typename T>
T* Buffer() {
return reinterpret_cast<T*>(mBuffer);
}
/* Store the MAXSIZE last elements of the buffer. */
template <typename T>
void StoreTail(const Span<const T>& aInBuffer) {
StoreTail(aInBuffer.data(), aInBuffer.size());
}
template <typename T>
void StoreTail(const T* aInBuffer, uint32_t aInFrames) {
const T* inBuffer = aInBuffer;
mSize = std::min(aInFrames, MAXSIZE);
if (inBuffer) {
PodCopy(Buffer<T>(), inBuffer + aInFrames - mSize, mSize);
} else {
std::fill_n(Buffer<T>(), mSize, static_cast<T>(0));
}
}
uint32_t Length() { return mSize; }
static constexpr uint32_t MAXSIZE = 20;
private:
float mBuffer[MAXSIZE] = {};
uint32_t mSize = 0;
};
AutoTArray<TailBuffer, STEREO> mInputTail;
WavDumper mInputStreamFile;
WavDumper mOutputStreamFile;
};
} // namespace mozilla
#endif // DOM_MEDIA_DRIFTCONTROL_DYNAMICRESAMPLER_H_