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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 <mozilla/UniquePtr.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 can be 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
*
* Alternatively, consumers can pass in a buffer in which the output is copied.
* The buffer needs to be large enough to store a packet worth of audio.
*
* 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(const 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();
UniquePtr<InputType[]> oldStorage = mozilla::Move(mStorage);
mStorage = mozilla::MakeUnique<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];
Output(out);
return out;
}
void Output(OutputType* aOutputBuffer)
{
uint32_t samplesNeeded = mPacketSize * mChannels;
// 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(aOutputBuffer + AvailableSamples(), zeros);
samplesNeeded -= zeros;
}
if (ReadIndex() + samplesNeeded <= mLength) {
ConvertAudioSamples<InputType,OutputType>(mStorage.get() + ReadIndex(),
aOutputBuffer,
samplesNeeded);
} else {
uint32_t firstPartLength = mLength - ReadIndex();
uint32_t secondPartLength = samplesNeeded - firstPartLength;
ConvertAudioSamples<InputType, OutputType>(mStorage.get() + ReadIndex(),
aOutputBuffer,
firstPartLength);
ConvertAudioSamples<InputType, OutputType>(mStorage.get(),
aOutputBuffer + firstPartLength,
secondPartLength);
}
mReadIndex += samplesNeeded;
}
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
mozilla::UniquePtr<InputType[]> mStorage;
// Length of the buffer, in samples
uint32_t mLength;
};
} // mozilla
#endif // AudioPacketizer_h_
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