gecko/content/media/encoder/OpusTrackEncoder.cpp
Justin Lebar 6c46d0633d Bug 820686 - Remove code after MOZ_CRASH or MOZ_ASSUME_NOT_REACHED. r=(see below)
r=tbsaunde for accessible
r=jmuizelaar for gfx
r=waldo for js
r=roc for layout
r=glandium for mozglue
r=jduell for netwerk
r=khuey for everything else
2013-06-28 18:38:32 -07:00

332 lines
10 KiB
C++

/* -*- 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 <opus/opus.h>
#undef LOG
#ifdef MOZ_WIDGET_GONK
#include <android/log.h>
#define LOG(args...) __android_log_print(ANDROID_LOG_INFO, "MediaEncoder", ## args);
#else
#define LOG(args, ...)
#endif
namespace mozilla {
// 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;
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[9] = "OpusHead";
memcpy(aOutput->AppendElements(sizeof(magic) - 1), 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[9] = "OpusTags";
memcpy(aOutput->AppendElements(sizeof(magic) - 1), 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()
, mEncoderState(ID_HEADER)
, mEncoder(nullptr)
, mSourceSegment(new AudioSegment())
, mLookahead(0)
{
}
OpusTrackEncoder::~OpusTrackEncoder()
{
if (mEncoder) {
opus_encoder_destroy(mEncoder);
}
}
nsresult
OpusTrackEncoder::Init(int aChannels, int aSamplingRate)
{
// The track must have 1 or 2 channels.
if (aChannels <= 0 || aChannels > MAX_CHANNELS) {
LOG("[Opus] Fail to create the AudioTrackEncoder! The input has"
" %d channel(s), but expects no more than %d.", aChannels, MAX_CHANNELS);
return NS_ERROR_INVALID_ARG;
}
// This monitor is used to wake up other methods that are waiting for encoder
// to be completely initialized.
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
mChannels = aChannels;
// The granule position is required to be incremented at a rate of 48KHz, and
// it is simply calculated as |granulepos = samples * (48000/source_rate)|,
// that is, the source sampling rate must divide 48000 evenly.
if (!((aSamplingRate >= 8000) && (kOpusSamplingRate / aSamplingRate) *
aSamplingRate == kOpusSamplingRate)) {
LOG("[Opus] Error! The source sample rate should be greater than 8000 and"
" divides 48000 evenly.");
return NS_ERROR_FAILURE;
}
mSamplingRate = aSamplingRate;
int error = 0;
mEncoder = opus_encoder_create(mSamplingRate, mChannels,
OPUS_APPLICATION_AUDIO, &error);
mInitialized = (error == OPUS_OK);
mReentrantMonitor.NotifyAll();
return error == OPUS_OK ? NS_OK : NS_ERROR_FAILURE;
}
int
OpusTrackEncoder::GetPacketDuration()
{
return mSamplingRate * kFrameDurationMs / 1000;
}
nsresult
OpusTrackEncoder::GetHeader(nsTArray<uint8_t>* aOutput)
{
{
// Wait if mEncoder is not initialized.
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
while (!mCanceled && !mEncoder) {
mReentrantMonitor.Wait();
}
}
if (mCanceled) {
return NS_ERROR_FAILURE;
}
switch (mEncoderState) {
case ID_HEADER:
{
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/mSamplingRate),
mSamplingRate, aOutput);
mEncoderState = COMMENT_HEADER;
break;
}
case COMMENT_HEADER:
{
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, aOutput);
mEncoderState = DATA;
break;
}
case DATA:
// No more headers.
break;
default:
MOZ_CRASH("Invalid state");
}
return NS_OK;
}
nsresult
OpusTrackEncoder::GetEncodedTrack(nsTArray<uint8_t>* aOutput,
int &aOutputDuration)
{
{
// Move all the samples from mRawSegment to mSourceSegment. We only hold
// the monitor in this block.
ReentrantMonitorAutoEnter mon(mReentrantMonitor);
// Wait if mEncoder is not initialized, or when not enough raw data, but is
// not the end of stream nor is being canceled.
while (!mCanceled && (!mEncoder || (mRawSegment->GetDuration() +
mSourceSegment->GetDuration() < GetPacketDuration() &&
!mEndOfStream))) {
mReentrantMonitor.Wait();
}
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) {
mSourceSegment->AppendNullData(mLookahead);
}
}
// Start encoding data.
nsAutoTArray<AudioDataValue, 9600> pcm;
pcm.SetLength(GetPacketDuration() * mChannels);
AudioSegment::ChunkIterator iter(*mSourceSegment);
int frameCopied = 0;
while (!iter.IsEnded() && frameCopied < GetPacketDuration()) {
AudioChunk chunk = *iter;
// Chunk to the required frame size.
int frameToCopy = chunk.GetDuration();
if (frameCopied + frameToCopy > GetPacketDuration()) {
frameToCopy = GetPacketDuration() - frameCopied;
}
if (!chunk.IsNull()) {
// Append the interleaved data to the end of pcm buffer.
InterleaveTrackData(chunk, frameToCopy, mChannels,
pcm.Elements() + frameCopied);
} else {
for (int i = 0; i < frameToCopy * mChannels; i++) {
pcm.AppendElement(0);
}
}
frameCopied += frameToCopy;
iter.Next();
}
// The ogg time stamping and pre-skip is always timed at 48000.
aOutputDuration = 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 && mEndOfStream) {
mDoneEncoding = true;
LOG("[Opus] Done encoding.");
}
// Append null data to pcm buffer if the leftover data is not enough for
// opus encoder.
if (frameCopied < GetPacketDuration() && mEndOfStream) {
for (int i = frameCopied * mChannels; i < GetPacketDuration() * mChannels; i++) {
pcm.AppendElement(0);
}
}
// Encode the data with Opus Encoder.
aOutput->SetLength(MAX_DATA_BYTES);
// result is returned as opus error code if it is negative.
int result = 0;
#ifdef MOZ_SAMPLE_TYPE_S16
const opus_int16* pcmBuf = static_cast<opus_int16*>(pcm.Elements());
result = opus_encode(mEncoder, pcmBuf, GetPacketDuration(),
aOutput->Elements(), MAX_DATA_BYTES);
#else
const float* pcmBuf = static_cast<float*>(pcm.Elements());
result = opus_encode_float(mEncoder, pcmBuf, GetPacketDuration(),
aOutput->Elements(), MAX_DATA_BYTES);
#endif
aOutput->SetLength(result >= 0 ? result : 0);
if (result < 0) {
LOG("[Opus] Fail to encode data! Result: %s.", opus_strerror(result));
}
return result >= 0 ? NS_OK : NS_ERROR_FAILURE;
}
}