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openboard/src/podcast/ffmpeg/UBFFmpegVideoEncoder.cpp

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/*
* Copyright (C) 2015-2016 Département de l'Instruction Publique (DIP-SEM)
*
* This file is part of OpenBoard.
*
* OpenBoard is free software: you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation, version 3 of the License,
* with a specific linking exception for the OpenSSL project's
* "OpenSSL" library (or with modified versions of it that use the
* same license as the "OpenSSL" library).
*
* OpenBoard is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with OpenBoard. If not, see <http://www.gnu.org/licenses/>.
*/
#include "UBFFmpegVideoEncoder.h"
// Due to the whole FFmpeg / libAV silliness, we have to support libavresample instead
// of libswresapmle on some platforms, as well as now-obsolete function names
#if LIBAVFORMAT_VERSION_MICRO < 100
#define swr_alloc avresample_alloc_context
#define swr_init avresample_open
#define swr_get_out_samples avresample_get_out_samples
#define swr_free avresample_free
#define av_opt_set_sample_fmt av_opt_set_int
#define av_frame_alloc avcodec_alloc_frame
#define av_frame_free avcodec_free_frame
#define av_packet_unref av_free_packet
#define AV_ERROR_MAX_STRING_SIZE 64
#define AV_CODEC_FLAG_GLOBAL_HEADER (1 << 22)
uint8_t* audio_samples_buffer; // used by processAudio because av_frame_get_buffer doesn't exist in this version
int avformat_alloc_output_context2(AVFormatContext **avctx, AVOutputFormat *oformat,
const char *format, const char *filename)
{
AVFormatContext *s = avformat_alloc_context();
int ret = 0;
*avctx = NULL;
if (!s)
goto nomem;
if (!oformat) {
if (format) {
oformat = av_guess_format(format, NULL, NULL);
if (!oformat) {
av_log(s, AV_LOG_ERROR, "Requested output format '%s' is not a suitable output format\n", format);
ret = AVERROR(EINVAL);
goto error;
}
} else {
oformat = av_guess_format(NULL, filename, NULL);
if (!oformat) {
ret = AVERROR(EINVAL);
av_log(s, AV_LOG_ERROR, "Unable to find a suitable output format for '%s'\n",
filename);
goto error;
}
}
}
s->oformat = oformat;
if (s->oformat->priv_data_size > 0) {
s->priv_data = av_mallocz(s->oformat->priv_data_size);
if (!s->priv_data)
goto nomem;
if (s->oformat->priv_class) {
*(const AVClass**)s->priv_data= s->oformat->priv_class;
av_opt_set_defaults(s->priv_data);
}
} else
s->priv_data = NULL;
if (filename)
av_strlcpy(s->filename, filename, sizeof(s->filename));
*avctx = s;
return 0;
nomem:
av_log(s, AV_LOG_ERROR, "Out of memory\n");
ret = AVERROR(ENOMEM);
error:
avformat_free_context(s);
return ret;
}
int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels,
int nb_samples, enum AVSampleFormat sample_fmt, int align)
{
int ret, nb_planes = av_sample_fmt_is_planar(sample_fmt) ? nb_channels : 1;
*audio_data = (uint8_t**) av_malloc(sizeof(*audio_data) * nb_planes);
if (!*audio_data)
return AVERROR(ENOMEM);
ret = av_samples_alloc(*audio_data, linesize, nb_channels,
nb_samples, sample_fmt, align);
if (ret < 0)
av_freep(audio_data);
return ret;
}
int swr_convert(struct SwrContext *s, uint8_t **out, int out_count, const uint8_t **in, int in_count)
{
return avresample_convert(s, out, 0, out_count, const_cast<uint8_t **>(in), 0, in_count);
}
#endif
#if LIBAVCODEC_VERSION_INT < AV_VERSION_INT(55,55,0)
void av_packet_rescale_ts(AVPacket *pkt, AVRational src_tb, AVRational dst_tb)
{
if (pkt->pts != AV_NOPTS_VALUE)
pkt->pts = av_rescale_q(pkt->pts, src_tb, dst_tb);
if (pkt->dts != AV_NOPTS_VALUE)
pkt->dts = av_rescale_q(pkt->dts, src_tb, dst_tb);
if (pkt->duration > 0)
pkt->duration = av_rescale_q(pkt->duration, src_tb, dst_tb);
if (pkt->convergence_duration > 0)
pkt->convergence_duration = av_rescale_q(pkt->convergence_duration, src_tb, dst_tb);
}
#endif
#if defined(LIBAVRESAMPLE_VERSION_INT) && LIBAVRESAMPLE_VERSION_INT < AV_VERSION_INT(1,3,0)
//#include <libavresample/internal.h>
int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples)
{
int64_t samples = avresample_get_delay(avr) + (int64_t)in_nb_samples;
/*
if (avr->resample_needed) {
samples = av_rescale_rnd(samples,
avr->out_sample_rate,
avr->in_sample_rate,
AV_ROUND_UP);
}
*/
samples += avresample_available(avr);
if (samples > INT_MAX)
return AVERROR(EINVAL);
return samples;
}
#endif
//-------------------------------------------------------------------------
// Utility functions
//-------------------------------------------------------------------------
QString avErrorToQString(int errnum)
{
char error[AV_ERROR_MAX_STRING_SIZE];
av_strerror(errnum, error, AV_ERROR_MAX_STRING_SIZE);
return QString(error);
}
/**
* @brief Write a given frame to the audio stream or, if a null frame is passed, flush the stream.
*
* @param frame An AVFrame to be written to the stream, or NULL to flush the stream
* @param packet A (reusable) packet, used to temporarily store frame data
* @param stream The stream to write to
* @param outputFormatContext The output format context
*/
void writeFrame(AVFrame *frame, AVPacket *packet, AVStream *stream, AVFormatContext *outputFormatContext)
{
int gotOutput, ret;
av_init_packet(packet);
do {
if (stream->codec->codec_type == AVMEDIA_TYPE_AUDIO)
ret = avcodec_encode_audio2(stream->codec, packet, frame, &gotOutput);
else
ret = avcodec_encode_video2(stream->codec, packet, frame, &gotOutput);
if (ret < 0)
qWarning() << "Couldn't encode audio frame: " << avErrorToQString(ret);
else if (gotOutput) {
AVRational codecTimebase = stream->codec->time_base;
AVRational streamVideoTimebase = stream->time_base;
av_packet_rescale_ts(packet, codecTimebase, streamVideoTimebase);
packet->stream_index = stream->index;
av_interleaved_write_frame(outputFormatContext, packet);
av_packet_unref(packet);
}
} while (gotOutput && !frame);
}
void flushStream(AVPacket *packet, AVStream *stream, AVFormatContext *outputFormatContext)
{
writeFrame(NULL, packet, stream, outputFormatContext);
}
//-------------------------------------------------------------------------
// UBFFmpegVideoEncoder
//-------------------------------------------------------------------------
UBFFmpegVideoEncoder::UBFFmpegVideoEncoder(QObject* parent)
: UBAbstractVideoEncoder(parent)
, mOutputFormatContext(NULL)
, mSwsContext(NULL)
, mShouldRecordAudio(true)
, mAudioInput(NULL)
, mSwrContext(NULL)
, mAudioOutBuffer(NULL)
, mAudioSampleRate(44100)
, mAudioFrameCount(0)
{
mVideoTimebase = 100 * framesPerSecond();
mVideoEncoderThread = new QThread;
mVideoWorker = new UBFFmpegVideoEncoderWorker(this);
mVideoWorker->moveToThread(mVideoEncoderThread);
connect(mVideoWorker, SIGNAL(error(QString)),
this, SLOT(setLastErrorMessage(QString)));
connect(mVideoEncoderThread, SIGNAL(started()),
mVideoWorker, SLOT(runEncoding()));
connect(mVideoWorker, SIGNAL(encodingFinished()),
mVideoEncoderThread, SLOT(quit()));
connect(mVideoEncoderThread, SIGNAL(finished()),
this, SLOT(finishEncoding()));
}
UBFFmpegVideoEncoder::~UBFFmpegVideoEncoder()
{
if (mVideoWorker)
delete mVideoWorker;
if (mVideoEncoderThread)
delete mVideoEncoderThread;
if (mAudioInput)
delete mAudioInput;
}
void UBFFmpegVideoEncoder::setLastErrorMessage(const QString& pMessage)
{
qWarning() << "FFmpeg video encoder:" << pMessage;
mLastErrorMessage = pMessage;
}
bool UBFFmpegVideoEncoder::start()
{
bool initialized = init();
if (initialized) {
mVideoEncoderThread->start();
if (mShouldRecordAudio)
mAudioInput->start();
}
return initialized;
}
bool UBFFmpegVideoEncoder::stop()
{
qDebug() << "Video encoder: stop requested";
mVideoWorker->stopEncoding();
if (mShouldRecordAudio)
mAudioInput->stop();
return true;
}
bool UBFFmpegVideoEncoder::init()
{
av_register_all();
avcodec_register_all();
AVDictionary * options = NULL;
int ret;
// Output format and context
// --------------------------------------
if (avformat_alloc_output_context2(&mOutputFormatContext, NULL,
"mp4", NULL) < 0)
{
setLastErrorMessage("Couldn't allocate video format context");
return false;
}
// The default codecs for mp4 are h264 and aac, we use those
// Video codec and context
// -------------------------------------
mVideoStream = avformat_new_stream(mOutputFormatContext, 0);
AVCodec * videoCodec = avcodec_find_encoder(mOutputFormatContext->oformat->video_codec);
if (!videoCodec) {
setLastErrorMessage("Video codec not found");
return false;
}
AVCodecContext* c = avcodec_alloc_context3(videoCodec);
c->bit_rate = videoBitsPerSecond();
c->width = videoSize().width();
c->height = videoSize().height();
c->time_base = {1, mVideoTimebase};
c->gop_size = 10;
c->max_b_frames = 0;
c->pix_fmt = AV_PIX_FMT_YUV420P;
if (mOutputFormatContext->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
/*
* Supported pixel formats for h264 are:
* AV_PIX_FMT_YUV420P
* AV_PIX_FMT_YUV422P
* AV_PIX_FMT_YUV444P
* AV_PIX_FMT_YUVJ420P
*/
av_dict_set(&options, "preset", "slow", 0);
av_dict_set(&options, "crf", "20", 0);
ret = avcodec_open2(c, videoCodec, &options);
if (ret < 0) {
setLastErrorMessage(QString("Couldn't open video codec: ") + avErrorToQString(ret));
return false;
}
mVideoStream->codec = c;
// Source images are RGB32, and should be converted to YUV for h264 video
mSwsContext = sws_getCachedContext(mSwsContext,
c->width, c->height, AV_PIX_FMT_RGB32,
c->width, c->height, c->pix_fmt,
SWS_BICUBIC, 0, 0, 0);
// Audio codec and context
// -------------------------------------
if (mShouldRecordAudio) {
// Microphone input
mAudioInput = new UBMicrophoneInput();
connect(mAudioInput, SIGNAL(audioLevelChanged(quint8)),
this, SIGNAL(audioLevelChanged(quint8)));
connect(mAudioInput, SIGNAL(dataAvailable(QByteArray)),
this, SLOT(onAudioAvailable(QByteArray)));
if (!mAudioInput->init()) {
setLastErrorMessage("Couldn't initialize audio input");
return false;
}
int inChannelCount = mAudioInput->channelCount();
int inSampleRate = mAudioInput->sampleRate();
// Codec
AVCodec * audioCodec = avcodec_find_encoder(mOutputFormatContext->oformat->audio_codec);
if (!audioCodec) {
setLastErrorMessage("Audio codec not found");
return false;
}
mAudioStream = avformat_new_stream(mOutputFormatContext, audioCodec);
mAudioStream->id = mOutputFormatContext->nb_streams-1;
c = mAudioStream->codec;
c->bit_rate = 96000;
c->sample_fmt = audioCodec->sample_fmts ? audioCodec->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;// FLTP by default for AAC
c->sample_rate = mAudioSampleRate;
c->channel_layout = AV_CH_LAYOUT_STEREO;
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
//https://trac.ffmpeg.org/wiki/Encode/H.264#Profile
//Omit this unless your target device only supports a certain profile
//(see https://trac.ffmpeg.org/wiki/Encode/H.264#Compatibility).
//c->profile = FF_PROFILE_AAC_MAIN;
c->time_base = { 1, c->sample_rate };
if (mOutputFormatContext->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
ret = avcodec_open2(c, audioCodec, NULL);
if (ret < 0) {
setLastErrorMessage(QString("Couldn't open audio codec: ") + avErrorToQString(ret));
return false;
}
// The input (raw sound from the microphone) may not match the codec's sampling rate,
// sample format or number of channels; we use libswresample to convert and resample it
mSwrContext = swr_alloc();
if (!mSwrContext) {
setLastErrorMessage("Could not allocate resampler context");
return false;
}
av_opt_set_int(mSwrContext, "in_channel_count", inChannelCount, 0);
av_opt_set_int(mSwrContext, "in_channel_layout", av_get_default_channel_layout(inChannelCount), 0);
av_opt_set_int(mSwrContext, "in_sample_rate", inSampleRate, 0);
av_opt_set_sample_fmt(mSwrContext, "in_sample_fmt", (AVSampleFormat)mAudioInput->sampleFormat(), 0);
av_opt_set_int(mSwrContext, "out_channel_count", c->channels, 0);
av_opt_set_int(mSwrContext, "out_channel_layout", c->channel_layout, 0);
av_opt_set_int(mSwrContext, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(mSwrContext, "out_sample_fmt", c->sample_fmt, 0);
ret = swr_init(mSwrContext);
if (ret < 0) {
setLastErrorMessage(QString("Couldn't initialize the resampling context: ") + avErrorToQString(ret));
return false;
}
// Buffer for resampled/converted audio
mAudioOutBuffer = av_audio_fifo_alloc(c->sample_fmt, c->channels, c->frame_size);
}
// Open the output file
ret = avio_open(&(mOutputFormatContext->pb), videoFileName().toStdString().c_str(), AVIO_FLAG_WRITE);
if (ret < 0) {
setLastErrorMessage(QString("Couldn't open video file for writing: ") + avErrorToQString(ret));
return false;
}
// Write stream header
ret = avformat_write_header(mOutputFormatContext, NULL);
if (ret < 0) {
setLastErrorMessage(QString("Couldn't write header to file: ") + avErrorToQString(ret));
return false;
}
return true;
}
/**
* This function should be called every time a new "screenshot" is ready.
* The image is converted to the right format and sent to the encoder.
*/
void UBFFmpegVideoEncoder::newPixmap(const QImage &pImage, long timestamp)
{
if (!mVideoWorker->isRunning()) {
qDebug() << "Encoder worker thread not running. Queuing frame.";
mPendingFrames.enqueue({pImage, timestamp});
}
else {
// First send any queued frames, then the latest one
while (!mPendingFrames.isEmpty()) {
AVFrame* avFrame = convertImageFrame(mPendingFrames.dequeue());
if (avFrame)
mVideoWorker->queueVideoFrame(avFrame);
}
// note: if converting the frame turns out to be too slow to do here, it
// can always be done from the worker thread (in that case,
// the worker's queue would contain ImageFrames rather than AVFrames)
AVFrame* avFrame = convertImageFrame({pImage, timestamp});
if (avFrame)
mVideoWorker->queueVideoFrame(avFrame);
// signal the worker that frames are available
mVideoWorker->mWaitCondition.wakeAll();
}
}
/**
* Convert a frame consisting of a QImage and timestamp to an AVFrame
* with the right pixel format and PTS
*/
AVFrame* UBFFmpegVideoEncoder::convertImageFrame(ImageFrame frame)
{
AVFrame* avFrame = av_frame_alloc();
avFrame->format = mVideoStream->codec->pix_fmt;
avFrame->width = mVideoStream->codec->width;
avFrame->height = mVideoStream->codec->height;
avFrame->pts = mVideoTimebase * frame.timestamp / 1000;
const uchar * rgbImage = frame.image.bits();
const int in_linesize[1] = { frame.image.bytesPerLine() };
// Allocate the output image
if (av_image_alloc(avFrame->data, avFrame->linesize, mVideoStream->codec->width,
mVideoStream->codec->height, mVideoStream->codec->pix_fmt, 32) < 0)
{
qWarning() << "Couldn't allocate image";
return NULL;
}
sws_scale(mSwsContext,
(const uint8_t* const*)&rgbImage,
in_linesize,
0,
mVideoStream->codec->height,
avFrame->data,
avFrame->linesize);
return avFrame;
}
void UBFFmpegVideoEncoder::onAudioAvailable(QByteArray data)
{
if (!data.isEmpty())
processAudio(data);
}
/**
* Resample and convert audio to match the encoder's settings and queue the
* output. If enough output data is available, it is packaged into AVFrames and
* sent to the encoder thread.
*/
void UBFFmpegVideoEncoder::processAudio(QByteArray &data)
{
int ret;
AVCodecContext* codecContext = mAudioStream->codec;
const char * inSamples = data.constData();
// The number of samples (per channel) in the input
int inSamplesCount = data.size() / ((mAudioInput->sampleSize() / 8) * mAudioInput->channelCount());
// The number of samples we will get after conversion
int outSamplesCount = swr_get_out_samples(mSwrContext, inSamplesCount);
// Allocate output samples
uint8_t ** outSamples = NULL;
int outSamplesLineSize;
ret = av_samples_alloc_array_and_samples(&outSamples, &outSamplesLineSize,
codecContext->channels, outSamplesCount,
codecContext->sample_fmt, 0);
if (ret < 0) {
qWarning() << "Could not allocate audio samples" << avErrorToQString(ret);
return;
}
// Convert to destination format
ret = swr_convert(mSwrContext,
outSamples, outSamplesCount,
(const uint8_t **)&inSamples, inSamplesCount);
if (ret < 0) {
qWarning() << "Error converting audio samples: " << avErrorToQString(ret);
return;
}
// Append the converted samples to the out buffer.
ret = av_audio_fifo_write(mAudioOutBuffer, (void**)outSamples, outSamplesCount);
if (ret < 0) {
qWarning() << "Could not write to FIFO queue: " << avErrorToQString(ret);
return;
}
// Keep the data queued until next call if the encoder thread isn't running
if (!mVideoWorker->isRunning())
return;
bool framesAdded = false;
while (av_audio_fifo_size(mAudioOutBuffer) > codecContext->frame_size) {
AVFrame * avFrame = av_frame_alloc();
avFrame->nb_samples = codecContext->frame_size;
avFrame->channel_layout = codecContext->channel_layout;
avFrame->format = codecContext->sample_fmt;
avFrame->sample_rate = codecContext->sample_rate;
avFrame->pts = mAudioFrameCount;
#if LIBAVFORMAT_VERSION_MICRO < 100
int buffer_size = av_samples_get_buffer_size(NULL, codecContext->channels, codecContext->frame_size, codecContext->sample_fmt, 0);
audio_samples_buffer = (uint8_t*)av_malloc(buffer_size);
if (!audio_samples_buffer) {
qWarning() << "Couldn't allocate samples for audio frame: " << avErrorToQString(ret);
break;
}
ret = avcodec_fill_audio_frame(avFrame,
codecContext->channels,
codecContext->sample_fmt,
(const uint8_t*)audio_samples_buffer,
buffer_size,
0);
#else
ret = av_frame_get_buffer(avFrame, 0);
#endif
if (ret < 0) {
qWarning() << "Couldn't allocate frame: " << avErrorToQString(ret);
break;
}
ret = av_audio_fifo_read(mAudioOutBuffer, (void**)avFrame->data, codecContext->frame_size);
if (ret < 0)
qWarning() << "Could not read from FIFO queue: " << avErrorToQString(ret);
else {
mAudioFrameCount += codecContext->frame_size;
mVideoWorker->queueAudioFrame(avFrame);
framesAdded = true;
}
}
if (framesAdded)
mVideoWorker->mWaitCondition.wakeAll();
}
void UBFFmpegVideoEncoder::finishEncoding()
{
qDebug() << "VideoEncoder::finishEncoding called";
flushStream(mVideoWorker->mVideoPacket, mVideoStream, mOutputFormatContext);
if (mShouldRecordAudio)
flushStream(mVideoWorker->mAudioPacket, mAudioStream, mOutputFormatContext);
av_write_trailer(mOutputFormatContext);
avio_close(mOutputFormatContext->pb);
avcodec_close(mVideoStream->codec);
sws_freeContext(mSwsContext);
if (mShouldRecordAudio) {
avcodec_close(mAudioStream->codec);
swr_free(&mSwrContext);
}
avformat_free_context(mOutputFormatContext);
emit encodingFinished(true);
}
//-------------------------------------------------------------------------
// Worker
//-------------------------------------------------------------------------
UBFFmpegVideoEncoderWorker::UBFFmpegVideoEncoderWorker(UBFFmpegVideoEncoder* controller)
: mController(controller)
{
mStopRequested = false;
mIsRunning = false;
mVideoPacket = new AVPacket();
mAudioPacket = new AVPacket();
}
UBFFmpegVideoEncoderWorker::~UBFFmpegVideoEncoderWorker()
{
if (mVideoPacket)
delete mVideoPacket;
if (mAudioPacket)
delete mAudioPacket;
}
void UBFFmpegVideoEncoderWorker::stopEncoding()
{
qDebug() << "Video worker: stop requested";
mStopRequested = true;
mWaitCondition.wakeAll();
}
void UBFFmpegVideoEncoderWorker::queueVideoFrame(AVFrame* frame)
{
if (frame) {
mFrameQueueMutex.lock();
mImageQueue.enqueue(frame);
mFrameQueueMutex.unlock();
}
}
void UBFFmpegVideoEncoderWorker::queueAudioFrame(AVFrame* frame)
{
if (frame) {
mFrameQueueMutex.lock();
mAudioQueue.enqueue(frame);
mFrameQueueMutex.unlock();
}
}
/**
* The main encoding function. Takes the queued frames and
* writes them to the video and audio streams
*/
void UBFFmpegVideoEncoderWorker::runEncoding()
{
mIsRunning = true;
while (!mStopRequested) {
mFrameQueueMutex.lock();
mWaitCondition.wait(&mFrameQueueMutex);
while (!mImageQueue.isEmpty()) {
writeLatestVideoFrame();
}
while (!mAudioQueue.isEmpty()) {
writeLatestAudioFrame();
}
mFrameQueueMutex.unlock();
}
emit encodingFinished();
}
void UBFFmpegVideoEncoderWorker::writeLatestVideoFrame()
{
AVFrame* frame = mImageQueue.dequeue();
writeFrame(frame, mVideoPacket, mController->mVideoStream, mController->mOutputFormatContext);
av_frame_free(&frame);
}
void UBFFmpegVideoEncoderWorker::writeLatestAudioFrame()
{
AVFrame *frame = mAudioQueue.dequeue();
writeFrame(frame, mAudioPacket, mController->mAudioStream, mController->mOutputFormatContext);
av_frame_free(&frame);
#if LIBAVFORMAT_VERSION_MICRO < 100
if (audio_samples_buffer) {
av_free(audio_samples_buffer);
audio_samples_buffer = NULL;
}
#endif
}