Mirror of roytam1's UXP fork just in case Moonchild and Tobin decide to go after him
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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 "OpusTrackEncoder.h"
#include "nsString.h"
#include "GeckoProfiler.h"
#include <opus/opus.h>
#undef LOG
#define LOG(args, ...)
namespace mozilla {
// The Opus format supports up to 8 channels, and supports multitrack audio up
// to 255 channels, but the current implementation supports only mono and
// stereo, and downmixes any more than that.
static const int MAX_SUPPORTED_AUDIO_CHANNELS = 8;
// http://www.opus-codec.org/docs/html_api-1.0.2/group__opus__encoder.html
// In section "opus_encoder_init", channels must be 1 or 2 of input signal.
static const int MAX_CHANNELS = 2;
// A maximum data bytes for Opus to encode.
static const int MAX_DATA_BYTES = 4096;
// http://tools.ietf.org/html/draft-ietf-codec-oggopus-00#section-4
// Second paragraph, " The granule position of an audio data page is in units
// of PCM audio samples at a fixed rate of 48 kHz."
static const int kOpusSamplingRate = 48000;
// The duration of an Opus frame, and it must be 2.5, 5, 10, 20, 40 or 60 ms.
static const int kFrameDurationMs = 20;
// The supported sampling rate of input signal (Hz),
// must be one of the following. Will resampled to 48kHz otherwise.
static const int kOpusSupportedInputSamplingRates[] =
{8000, 12000, 16000, 24000, 48000};
namespace {
// An endian-neutral serialization of integers. Serializing T in little endian
// format to aOutput, where T is a 16 bits or 32 bits integer.
template<typename T>
static void
SerializeToBuffer(T aValue, nsTArray<uint8_t>* aOutput)
{
for (uint32_t i = 0; i < sizeof(T); i++) {
aOutput->AppendElement((uint8_t)(0x000000ff & (aValue >> (i * 8))));
}
}
static inline void
SerializeToBuffer(const nsCString& aComment, nsTArray<uint8_t>* aOutput)
{
// Format of serializing a string to buffer is, the length of string (32 bits,
// little endian), and the string.
SerializeToBuffer((uint32_t)(aComment.Length()), aOutput);
aOutput->AppendElements(aComment.get(), aComment.Length());
}
static void
SerializeOpusIdHeader(uint8_t aChannelCount, uint16_t aPreskip,
uint32_t aInputSampleRate, nsTArray<uint8_t>* aOutput)
{
// The magic signature, null terminator has to be stripped off from strings.
static const uint8_t magic[] = "OpusHead";
aOutput->AppendElements(magic, sizeof(magic) - 1);
// The version must always be 1 (8 bits, unsigned).
aOutput->AppendElement(1);
// Number of output channels (8 bits, unsigned).
aOutput->AppendElement(aChannelCount);
// Number of samples (at 48 kHz) to discard from the decoder output when
// starting playback (16 bits, unsigned, little endian).
SerializeToBuffer(aPreskip, aOutput);
// The sampling rate of input source (32 bits, unsigned, little endian).
SerializeToBuffer(aInputSampleRate, aOutput);
// Output gain, an encoder should set this field to zero (16 bits, signed,
// little endian).
SerializeToBuffer((int16_t)0, aOutput);
// Channel mapping family. Family 0 allows only 1 or 2 channels (8 bits,
// unsigned).
aOutput->AppendElement(0);
}
static void
SerializeOpusCommentHeader(const nsCString& aVendor,
const nsTArray<nsCString>& aComments,
nsTArray<uint8_t>* aOutput)
{
// The magic signature, null terminator has to be stripped off.
static const uint8_t magic[] = "OpusTags";
aOutput->AppendElements(magic, sizeof(magic) - 1);
// The vendor; Should append in the following order:
// vendor string length (32 bits, unsigned, little endian)
// vendor string.
SerializeToBuffer(aVendor, aOutput);
// Add comments; Should append in the following order:
// comment list length (32 bits, unsigned, little endian)
// comment #0 string length (32 bits, unsigned, little endian)
// comment #0 string
// comment #1 string length (32 bits, unsigned, little endian)
// comment #1 string ...
SerializeToBuffer((uint32_t)aComments.Length(), aOutput);
for (uint32_t i = 0; i < aComments.Length(); ++i) {
SerializeToBuffer(aComments[i], aOutput);
}
}
} // Anonymous namespace.
OpusTrackEncoder::OpusTrackEncoder()
: AudioTrackEncoder()
, mEncoder(nullptr)
, mLookahead(0)
, mResampler(nullptr)
, mOutputTimeStamp(0)
{
}
OpusTrackEncoder::~OpusTrackEncoder()
{
if (mEncoder) {
opus_encoder_destroy(mEncoder);
}
if (mResampler) {
speex_resampler_destroy(mResampler);
mResampler = nullptr;
}
}
nsresult
OpusTrackEncoder::Init(int aChannels, int aSamplingRate)
{
// This monitor is used to wake up other methods that are waiting for encoder
// to be completely initialized.
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
NS_ENSURE_TRUE((aChannels <= MAX_SUPPORTED_AUDIO_CHANNELS) && (aChannels > 0),
NS_ERROR_FAILURE);
// This version of encoder API only support 1 or 2 channels,
// So set the mChannels less or equal 2 and
// let InterleaveTrackData downmix pcm data.
mChannels = aChannels > MAX_CHANNELS ? MAX_CHANNELS : aChannels;
// Reject non-audio sample rates.
NS_ENSURE_TRUE(aSamplingRate >= 8000, NS_ERROR_INVALID_ARG);
NS_ENSURE_TRUE(aSamplingRate <= 192000, NS_ERROR_INVALID_ARG);
// According to www.opus-codec.org, creating an opus encoder requires the
// sampling rate of source signal be one of 8000, 12000, 16000, 24000, or
// 48000. If this constraint is not satisfied, we resample the input to 48kHz.
nsTArray<int> supportedSamplingRates;
supportedSamplingRates.AppendElements(kOpusSupportedInputSamplingRates,
ArrayLength(kOpusSupportedInputSamplingRates));
if (!supportedSamplingRates.Contains(aSamplingRate)) {
int error;
mResampler = speex_resampler_init(mChannels,
aSamplingRate,
kOpusSamplingRate,
SPEEX_RESAMPLER_QUALITY_DEFAULT,
&error);
if (error != RESAMPLER_ERR_SUCCESS) {
return NS_ERROR_FAILURE;
}
}
mSamplingRate = aSamplingRate;
NS_ENSURE_TRUE(mSamplingRate > 0, NS_ERROR_FAILURE);
int error = 0;
mEncoder = opus_encoder_create(GetOutputSampleRate(), mChannels,
OPUS_APPLICATION_AUDIO, &error);
mInitialized = (error == OPUS_OK);
if (mAudioBitrate) {
opus_encoder_ctl(mEncoder, OPUS_SET_BITRATE(static_cast<int>(mAudioBitrate)));
}
mReentrantMonitor.NotifyAll();
return error == OPUS_OK ? NS_OK : NS_ERROR_FAILURE;
}
int
OpusTrackEncoder::GetOutputSampleRate()
{
return mResampler ? kOpusSamplingRate : mSamplingRate;
}
int
OpusTrackEncoder::GetPacketDuration()
{
return GetOutputSampleRate() * kFrameDurationMs / 1000;
}
already_AddRefed<TrackMetadataBase>
OpusTrackEncoder::GetMetadata()
{
PROFILER_LABEL("OpusTrackEncoder", "GetMetadata",
js::ProfileEntry::Category::OTHER);
{
// Wait if mEncoder is not initialized.
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
while (!mCanceled && !mInitialized) {
mReentrantMonitor.Wait();
}
}
if (mCanceled || mEncodingComplete) {
return nullptr;
}
RefPtr<OpusMetadata> meta = new OpusMetadata();
meta->mChannels = mChannels;
meta->mSamplingFrequency = mSamplingRate;
mLookahead = 0;
int error = opus_encoder_ctl(mEncoder, OPUS_GET_LOOKAHEAD(&mLookahead));
if (error != OPUS_OK) {
mLookahead = 0;
}
// The ogg time stamping and pre-skip is always timed at 48000.
SerializeOpusIdHeader(mChannels, mLookahead * (kOpusSamplingRate /
GetOutputSampleRate()), mSamplingRate,
&meta->mIdHeader);
nsCString vendor;
vendor.AppendASCII(opus_get_version_string());
nsTArray<nsCString> comments;
comments.AppendElement(NS_LITERAL_CSTRING("ENCODER=Mozilla" MOZ_APP_UA_VERSION));
SerializeOpusCommentHeader(vendor, comments,
&meta->mCommentHeader);
return meta.forget();
}
nsresult
OpusTrackEncoder::GetEncodedTrack(EncodedFrameContainer& aData)
{
PROFILER_LABEL("OpusTrackEncoder", "GetEncodedTrack",
js::ProfileEntry::Category::OTHER);
{
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
// Wait until initialized or cancelled.
while (!mCanceled && !mInitialized) {
mReentrantMonitor.Wait();
}
if (mCanceled || mEncodingComplete) {
return NS_ERROR_FAILURE;
}
}
// calculation below depends on the truth that mInitialized is true.
MOZ_ASSERT(mInitialized);
bool wait = true;
int result = 0;
// Only wait once, then loop until we run out of packets of input data
while (result >= 0 && !mEncodingComplete) {
// re-sampled frames left last time which didn't fit into an Opus packet duration.
const int framesLeft = mResampledLeftover.Length() / mChannels;
// When framesLeft is 0, (GetPacketDuration() - framesLeft) is a multiple
// of kOpusSamplingRate. There is not precision loss in the integer division
// in computing framesToFetch. If frameLeft > 0, we need to add 1 to
// framesToFetch to ensure there will be at least n frames after re-sampling.
const int frameRoundUp = framesLeft ? 1 : 0;
MOZ_ASSERT(GetPacketDuration() >= framesLeft);
// Try to fetch m frames such that there will be n frames
// where (n + frameLeft) >= GetPacketDuration() after re-sampling.
const int framesToFetch = !mResampler ? GetPacketDuration()
: (GetPacketDuration() - framesLeft) * mSamplingRate / kOpusSamplingRate
+ frameRoundUp;
{
// Move all the samples from mRawSegment to mSourceSegment. We only hold
// the monitor in this block.
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
// Wait until enough raw data, end of stream or cancelled.
while (!mCanceled && mRawSegment.GetDuration() +
mSourceSegment.GetDuration() < framesToFetch &&
!mEndOfStream) {
if (wait) {
mReentrantMonitor.Wait();
wait = false;
} else {
goto done; // nested while's...
}
}
if (mCanceled) {
return NS_ERROR_FAILURE;
}
mSourceSegment.AppendFrom(&mRawSegment);
// Pad |mLookahead| samples to the end of source stream to prevent lost of
// original data, the pcm duration will be calculated at rate 48K later.
if (mEndOfStream && !mEosSetInEncoder) {
mEosSetInEncoder = true;
mSourceSegment.AppendNullData(mLookahead);
}
}
// Start encoding data.
AutoTArray<AudioDataValue, 9600> pcm;
pcm.SetLength(GetPacketDuration() * mChannels);
AudioSegment::ChunkIterator iter(mSourceSegment);
int frameCopied = 0;
while (!iter.IsEnded() && frameCopied < framesToFetch) {
AudioChunk chunk = *iter;
// Chunk to the required frame size.
int frameToCopy = chunk.GetDuration();
if (frameCopied + frameToCopy > framesToFetch) {
frameToCopy = framesToFetch - frameCopied;
}
if (!chunk.IsNull()) {
// Append the interleaved data to the end of pcm buffer.
AudioTrackEncoder::InterleaveTrackData(chunk, frameToCopy, mChannels,
pcm.Elements() + frameCopied * mChannels);
} else {
memset(pcm.Elements() + frameCopied * mChannels, 0,
frameToCopy * mChannels * sizeof(AudioDataValue));
}
frameCopied += frameToCopy;
iter.Next();
}
RefPtr<EncodedFrame> audiodata = new EncodedFrame();
audiodata->SetFrameType(EncodedFrame::OPUS_AUDIO_FRAME);
int framesInPCM = frameCopied;
if (mResampler) {
AutoTArray<AudioDataValue, 9600> resamplingDest;
// We want to consume all the input data, so we slightly oversize the
// resampled data buffer so we can fit the output data in. We cannot really
// predict the output frame count at each call.
uint32_t outframes = frameCopied * kOpusSamplingRate / mSamplingRate + 1;
uint32_t inframes = frameCopied;
resamplingDest.SetLength(outframes * mChannels);
#if MOZ_SAMPLE_TYPE_S16
short* in = reinterpret_cast<short*>(pcm.Elements());
short* out = reinterpret_cast<short*>(resamplingDest.Elements());
speex_resampler_process_interleaved_int(mResampler, in, &inframes,
out, &outframes);
#else
float* in = reinterpret_cast<float*>(pcm.Elements());
float* out = reinterpret_cast<float*>(resamplingDest.Elements());
speex_resampler_process_interleaved_float(mResampler, in, &inframes,
out, &outframes);
#endif
MOZ_ASSERT(pcm.Length() >= mResampledLeftover.Length());
PodCopy(pcm.Elements(), mResampledLeftover.Elements(),
mResampledLeftover.Length());
uint32_t outframesToCopy = std::min(outframes,
static_cast<uint32_t>(GetPacketDuration() - framesLeft));
MOZ_ASSERT(pcm.Length() - mResampledLeftover.Length() >=
outframesToCopy * mChannels);
PodCopy(pcm.Elements() + mResampledLeftover.Length(),
resamplingDest.Elements(), outframesToCopy * mChannels);
int frameLeftover = outframes - outframesToCopy;
mResampledLeftover.SetLength(frameLeftover * mChannels);
PodCopy(mResampledLeftover.Elements(),
resamplingDest.Elements() + outframesToCopy * mChannels,
mResampledLeftover.Length());
// This is always at 48000Hz.
framesInPCM = framesLeft + outframesToCopy;
audiodata->SetDuration(framesInPCM);
} else {
// The ogg time stamping and pre-skip is always timed at 48000.
audiodata->SetDuration(frameCopied * (kOpusSamplingRate / mSamplingRate));
}
// Remove the raw data which has been pulled to pcm buffer.
// The value of frameCopied should equal to (or smaller than, if eos)
// GetPacketDuration().
mSourceSegment.RemoveLeading(frameCopied);
// Has reached the end of input stream and all queued data has pulled for
// encoding.
if (mSourceSegment.GetDuration() == 0 && mEosSetInEncoder) {
mEncodingComplete = true;
LOG("[Opus] Done encoding.");
}
MOZ_ASSERT(mEosSetInEncoder || framesInPCM == GetPacketDuration());
// Append null data to pcm buffer if the leftover data is not enough for
// opus encoder.
if (framesInPCM < GetPacketDuration() && mEosSetInEncoder) {
PodZero(pcm.Elements() + framesInPCM * mChannels,
(GetPacketDuration() - framesInPCM) * mChannels);
}
nsTArray<uint8_t> frameData;
// Encode the data with Opus Encoder.
frameData.SetLength(MAX_DATA_BYTES);
// result is returned as opus error code if it is negative.
result = 0;
#ifdef MOZ_SAMPLE_TYPE_S16
const opus_int16* pcmBuf = static_cast<opus_int16*>(pcm.Elements());
result = opus_encode(mEncoder, pcmBuf, GetPacketDuration(),
frameData.Elements(), MAX_DATA_BYTES);
#else
const float* pcmBuf = static_cast<float*>(pcm.Elements());
result = opus_encode_float(mEncoder, pcmBuf, GetPacketDuration(),
frameData.Elements(), MAX_DATA_BYTES);
#endif
frameData.SetLength(result >= 0 ? result : 0);
if (result < 0) {
LOG("[Opus] Fail to encode data! Result: %s.", opus_strerror(result));
}
if (mEncodingComplete) {
if (mResampler) {
speex_resampler_destroy(mResampler);
mResampler = nullptr;
}
mResampledLeftover.SetLength(0);
}
audiodata->SwapInFrameData(frameData);
// timestamp should be the time of the first sample
audiodata->SetTimeStamp(mOutputTimeStamp);
mOutputTimeStamp += FramesToUsecs(GetPacketDuration(), kOpusSamplingRate).value();
LOG("[Opus] mOutputTimeStamp %lld.",mOutputTimeStamp);
aData.AppendEncodedFrame(audiodata);
}
done:
return result >= 0 ? NS_OK : NS_ERROR_FAILURE;
}
} // namespace mozilla