Bug 901633 - Part 1 - Implement a generic audio packetizer. r=jesup

--HG--
extra : rebase_source : addc991e335f661b83a2dc0224da26a4eefa2a0d
This commit is contained in:
Paul Adenot 2015-07-30 13:51:57 +02:00
Родитель d31516a8ec
Коммит a7ae94ef7e
4 изменённых файлов: 361 добавлений и 1 удалений

186
dom/media/AudioPacketizer.h Normal file
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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,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#ifndef AudioPacketizer_h_
#define AudioPacketizer_h_
#include <mozilla/PodOperations.h>
#include <mozilla/Assertions.h>
#include <nsAutoPtr.h>
#include <AudioSampleFormat.h>
// Enable this to warn when `Output` has been called but not enough data was
// buffered.
// #define LOG_PACKETIZER_UNDERRUN
namespace mozilla {
/**
* This class takes arbitrary input data, and returns packets of a specific
* size. In the process, it can convert audio samples from 16bit integers to
* float (or vice-versa).
*
* Input and output, as well as length units in the public interface are
* interleaved frames.
*
* Allocations of output buffer are performed by this class. Buffers can simply
* be delete-d. This is because packets are intended to be sent off to
* non-gecko code using normal pointers/length pairs
*
* The implementation uses a circular buffer using absolute virtual indices.
*/
template <typename InputType, typename OutputType>
class AudioPacketizer
{
public:
AudioPacketizer(uint32_t aPacketSize, uint32_t aChannels)
: mPacketSize(aPacketSize)
, mChannels(aChannels)
, mReadIndex(0)
, mWriteIndex(0)
// Start off with a single packet
, mStorage(new InputType[aPacketSize * aChannels])
, mLength(aPacketSize * aChannels)
{
MOZ_ASSERT(aPacketSize > 0 && aChannels > 0,
"The packet size and the number of channel should be strictly positive");
}
void Input(InputType* aFrames, uint32_t aFrameCount)
{
uint32_t inputSamples = aFrameCount * mChannels;
// Need to grow the storage. This should rarely happen, if at all, once the
// array has the right size.
if (inputSamples > EmptySlots()) {
// Calls to Input and Output are roughtly interleaved
// (Input,Output,Input,Output, etc.), or balanced
// (Input,Input,Input,Output,Output,Output), so we update the buffer to
// the exact right size in order to not waste space.
uint32_t newLength = AvailableSamples() + inputSamples;
uint32_t toCopy = AvailableSamples();
nsAutoPtr<InputType> oldStorage = mStorage;
mStorage = new InputType[newLength];
// Copy the old data at the beginning of the new storage.
if (WriteIndex() >= ReadIndex()) {
PodCopy(mStorage.get(),
oldStorage.get() + ReadIndex(),
AvailableSamples());
} else {
uint32_t firstPartLength = mLength - ReadIndex();
uint32_t secondPartLength = AvailableSamples() - firstPartLength;
PodCopy(mStorage.get(),
oldStorage.get() + ReadIndex(),
firstPartLength);
PodCopy(mStorage.get() + firstPartLength,
oldStorage.get(),
secondPartLength);
}
mWriteIndex = toCopy;
mReadIndex = 0;
mLength = newLength;
}
if (WriteIndex() + inputSamples <= mLength) {
PodCopy(mStorage.get() + WriteIndex(), aFrames, aFrameCount * mChannels);
} else {
uint32_t firstPartLength = mLength - WriteIndex();
uint32_t secondPartLength = inputSamples - firstPartLength;
PodCopy(mStorage.get() + WriteIndex(), aFrames, firstPartLength);
PodCopy(mStorage.get(), aFrames + firstPartLength, secondPartLength);
}
mWriteIndex += inputSamples;
}
OutputType* Output()
{
uint32_t samplesNeeded = mPacketSize * mChannels;
OutputType* out = new OutputType[samplesNeeded];
// Under-run. Pad the end of the buffer with silence.
if (AvailableSamples() < samplesNeeded) {
#ifdef LOG_PACKETIZER_UNDERRUN
char buf[256];
snprintf(buf, 256,
"AudioPacketizer %p underrun: available: %u, needed: %u\n",
this, AvailableSamples(), samplesNeeded);
NS_WARNING(buf);
#endif
uint32_t zeros = samplesNeeded - AvailableSamples();
PodZero(out + AvailableSamples(), zeros);
samplesNeeded -= zeros;
}
if (ReadIndex() + samplesNeeded <= mLength) {
ConvertAudioSamples<InputType,OutputType>(mStorage.get() + ReadIndex(),
out,
samplesNeeded);
} else {
uint32_t firstPartLength = mLength - ReadIndex();
uint32_t secondPartLength = samplesNeeded - firstPartLength;
ConvertAudioSamples<InputType, OutputType>(mStorage.get() + ReadIndex(),
out,
firstPartLength);
ConvertAudioSamples<InputType, OutputType>(mStorage.get(),
out + firstPartLength,
secondPartLength);
}
mReadIndex += samplesNeeded;
return out;
}
uint32_t PacketsAvailable() const {
return AvailableSamples() / mChannels / mPacketSize;
}
bool Empty() const {
return mWriteIndex == mReadIndex;
}
bool Full() const {
return mWriteIndex - mReadIndex == mLength;
}
uint32_t PacketSize() const {
return mPacketSize;
}
uint32_t Channels() const {
return mChannels;
}
private:
uint32_t ReadIndex() const {
return mReadIndex % mLength;
}
uint32_t WriteIndex() const {
return mWriteIndex % mLength;
}
uint32_t AvailableSamples() const {
return mWriteIndex - mReadIndex;
}
uint32_t EmptySlots() const {
return mLength - AvailableSamples();
}
// Size of one packet of audio, in frames
uint32_t mPacketSize;
// Number of channels of the stream flowing through this packetizer
uint32_t mChannels;
// Two virtual index into the buffer: the read position and the write
// position.
uint64_t mReadIndex;
uint64_t mWriteIndex;
// Storage for the samples
nsAutoPtr<InputType> mStorage;
// Length of the buffer, in samples
uint32_t mLength;
};
} // mozilla
#endif // AudioPacketizer_h_

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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,
* You can obtain one at http://mozilla.org/MPL/2.0/. */
#include <stdint.h>
#include <assert.h>
#include "../AudioPacketizer.h"
using namespace mozilla;
template<typename T>
class AutoBuffer
{
public:
AutoBuffer(size_t aLength)
{
mStorage = new T[aLength];
}
~AutoBuffer() {
delete [] mStorage;
}
T* Get() {
return mStorage;
}
private:
T* mStorage;
};
int16_t Sequence(int16_t* aBuffer, uint32_t aSize, uint32_t aStart = 0)
{
uint32_t i;
for (i = 0; i < aSize; i++) {
aBuffer[i] = aStart + i;
}
return aStart + i;
}
void IsSequence(int16_t* aBuffer, uint32_t aSize, uint32_t aStart = 0)
{
for (uint32_t i = 0; i < aSize; i++) {
if (aBuffer[i] != static_cast<int64_t>(aStart + i)) {
fprintf(stderr, "Buffer is not a sequence at offset %u\n", i);
assert(false);
}
}
assert("Buffer is a sequence.");
}
void Zero(int16_t* aBuffer, uint32_t aSize)
{
for (uint32_t i = 0; i < aSize; i++) {
if (aBuffer[i] != 0) {
fprintf(stderr, "Buffer is not null at offset %u\n", i);
assert(false);
}
}
}
double sine(uint32_t aPhase) {
return sin(aPhase * 2 * M_PI * 440 / 44100);
}
int main() {
for (int16_t channels = 1; channels < 2; channels++) {
// Test that the packetizer returns zero on underrun
{
AudioPacketizer<int16_t, int16_t> ap(441, channels);
for (int16_t i = 0; i < 10; i++) {
int16_t* out = ap.Output();
Zero(out, 441);
delete out;
}
}
// Simple test, with input/output buffer size aligned on the packet size,
// alternating Input and Output calls.
{
AudioPacketizer<int16_t, int16_t> ap(441, channels);
int16_t seqEnd = 0;
for (int16_t i = 0; i < 10; i++) {
AutoBuffer<int16_t> b(441 * channels);
int16_t prevEnd = seqEnd;
seqEnd = Sequence(b.Get(), channels * 441, prevEnd);
ap.Input(b.Get(), 441);
int16_t* out = ap.Output();
IsSequence(out, 441 * channels, prevEnd);
delete out;
}
}
// Simple test, with input/output buffer size aligned on the packet size,
// alternating two Input and Output calls.
{
AudioPacketizer<int16_t, int16_t> ap(441, channels);
int16_t seqEnd = 0;
for (int16_t i = 0; i < 10; i++) {
AutoBuffer<int16_t> b(441 * channels);
AutoBuffer<int16_t> b1(441 * channels);
int16_t prevEnd0 = seqEnd;
seqEnd = Sequence(b.Get(), 441 * channels, prevEnd0);
int16_t prevEnd1 = seqEnd;
seqEnd = Sequence(b1.Get(), 441 * channels, seqEnd);
ap.Input(b.Get(), 441);
ap.Input(b1.Get(), 441);
int16_t* out = ap.Output();
int16_t* out2 = ap.Output();
IsSequence(out, 441 * channels, prevEnd0);
IsSequence(out2, 441 * channels, prevEnd1);
delete out;
delete out2;
}
}
// Input/output buffer size not aligned on the packet size,
// alternating two Input and Output calls.
{
AudioPacketizer<int16_t, int16_t> ap(441, channels);
int16_t prevEnd = 0;
int16_t prevSeq = 0;
for (int16_t i = 0; i < 10; i++) {
AutoBuffer<int16_t> b(480 * channels);
AutoBuffer<int16_t> b1(480 * channels);
prevSeq = Sequence(b.Get(), 480 * channels, prevSeq);
prevSeq = Sequence(b1.Get(), 480 * channels, prevSeq);
ap.Input(b.Get(), 480);
ap.Input(b1.Get(), 480);
int16_t* out = ap.Output();
int16_t* out2 = ap.Output();
IsSequence(out, 441 * channels, prevEnd);
prevEnd += 441 * channels;
IsSequence(out2, 441 * channels, prevEnd);
prevEnd += 441 * channels;
delete out;
delete out2;
}
printf("Available: %d\n", ap.PacketsAvailable());
}
// "Real-life" test case: streaming a sine wave through a packetizer, and
// checking that we have the right output.
// 128 is, for example, the size of a Web Audio API block, and 441 is the
// size of a webrtc.org packet when the sample rate is 44100 (10ms)
{
AudioPacketizer<int16_t, int16_t> ap(441, channels);
AutoBuffer<int16_t> b(128 * channels);
uint32_t phase = 0;
uint32_t outPhase = 0;
for (int16_t i = 0; i < 1000; i++) {
for (int32_t j = 0; j < 128; j++) {
for (int32_t c = 0; c < channels; c++) {
// int16_t sinewave at 440Hz/44100Hz sample rate
b.Get()[j * channels + c] = (2 << 14) * sine(phase);
}
phase++;
}
ap.Input(b.Get(), 128);
while (ap.PacketsAvailable()) {
int16_t* packet = ap.Output();
for (uint32_t k = 0; k < ap.PacketSize(); k++) {
for (int32_t c = 0; c < channels; c++) {
assert(packet[k * channels + c] ==
static_cast<int16_t>(((2 << 14) * sine(outPhase))));
}
outPhase++;
}
delete [] packet;
}
}
}
}
printf("OK\n");
return 0;
}

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@ -6,7 +6,8 @@
GeckoCppUnitTests([
'TestAudioBuffers',
'TestAudioMixer'
'TestAudioMixer',
'TestAudioPacketizer'
])
LOCAL_INCLUDES += [

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@ -101,6 +101,7 @@ EXPORTS += [
'AudioChannelFormat.h',
'AudioCompactor.h',
'AudioMixer.h',
'AudioPacketizer.h',
'AudioSampleFormat.h',
'AudioSegment.h',
'AudioStream.h',