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|
/**************************************************************************
* Karlyriceditor - a lyrics editor and CD+G / video export for Karaoke *
* songs. *
* Copyright (C) 2009-2013 George Yunaev, support@ulduzsoft.com *
* *
* This program 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, either version 3 of the License, or *
* (at your option) any later version. *
* *
* This program 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 this program. If not, see <http://www.gnu.org/licenses/>. *
**************************************************************************/
/**************************************************************************
* This class uses ideas from QTFFmpegWrapper - QT FFmpeg Wrapper Class *
* Copyright (C) 2009,2010: Daniel Roggen, droggen@gmail.com *
* All rights reserved. *
**************************************************************************/
#include "ffmpeg_headers.h"
#include "ffmpegvideoencoder.h"
#include "videoencodingprofiles.h"
#include "audioplayer.h"
#include "audioplayerprivate.h"
extern "C" {
#include "libavutil/mathematics.h"
}
#define MAX_AUDIO_FRAME_SIZE 192000 // 1 second of 48khz 32bit audio
class FFMpegVideoEncoderPriv
{
public:
FFMpegVideoEncoderPriv();
~FFMpegVideoEncoderPriv();
bool createFile( const QString& filename );
bool close();
int encodeImage( const QImage & img, qint64 time );
void flush();
public:
// Video output parameters
const VideoEncodingProfile * m_profile;
const VideoFormat * m_videoformat;
bool m_convertaudio;
unsigned int m_videobitrate;
unsigned int m_audiobitrate;
// Do we also have an audio source?
AudioPlayerPrivate * m_aplayer;
// Error message
QString m_errorMsg;
private:
bool convertImage_sws(const QImage &img);
// FFmpeg stuff
AVFormatContext * outputFormatCtx;
AVOutputFormat * outputFormat;
AVCodecContext * videoCodecCtx;
AVCodecContext * audioCodecCtx;
AVStream * videoStream;
AVStream * audioStream;
AVCodec * videoCodec;
AVCodec * audioCodec;
// Video frame data
AVFrame * videoFrame;
uint8_t * videoImageBuffer;
// Audio frame data
AVFrame * audioFrame;
uint8_t * audioSampleBuffer;
unsigned int audioSampleBuffer_size;
// Conversion
SwsContext * videoConvertCtx;
// FIXME: Audio resample
AVAudioResampleContext* audioResampleCtx;
// File has been outputFileOpened successfully
bool outputFileOpened;
// Video frame PTS
unsigned int videoFrameNumber;
unsigned int audioPTStracker;
// Total output size
unsigned int outputTotalSize;
};
static bool isAudioSampleFormatSupported( const enum AVSampleFormat * sample_formats, AVSampleFormat format )
{
while ( *sample_formats != AV_SAMPLE_FMT_NONE )
{
if ( *sample_formats == format )
return true;
sample_formats++;
}
return false;
}
FFMpegVideoEncoderPriv::FFMpegVideoEncoderPriv()
{
ffmpeg_init_once();
outputFormatCtx = 0;
outputFormat = 0;
videoCodecCtx = 0;
videoStream = 0;
audioStream = 0;
audioCodec = 0;
videoCodec = 0;
videoFrame = 0;
audioResampleCtx = 0;
audioFrame = 0;
audioCodecCtx = 0;
videoImageBuffer = 0;
audioSampleBuffer = 0;
videoConvertCtx = 0;
outputFileOpened = false;
}
FFMpegVideoEncoderPriv::~FFMpegVideoEncoderPriv()
{
close();
}
bool FFMpegVideoEncoderPriv::close()
{
if ( outputFormatCtx )
{
if ( outputFileOpened )
{
flush();
av_write_trailer( outputFormatCtx );
// close video and audio
avcodec_close( videoStream->codec );
avcodec_close( audioStream->codec );
// Close the file
avio_close( outputFormatCtx->pb );
}
// free the streams
for ( unsigned int i = 0; i < outputFormatCtx->nb_streams; i++ )
{
av_freep(&outputFormatCtx->streams[i]->codec);
av_freep(&outputFormatCtx->streams[i]);
}
// Free the format
av_free( outputFormatCtx );
}
delete[] videoImageBuffer;
delete[] audioSampleBuffer;
if ( videoFrame )
av_frame_free( &videoFrame );
if ( audioFrame )
av_frame_free( &audioFrame );
outputFormatCtx = 0;
outputFormat = 0;
videoCodecCtx = 0;
videoStream = 0;
audioStream = 0;
videoCodec = 0;
videoFrame = 0;
audioFrame = 0;
videoImageBuffer = 0;
audioSampleBuffer = 0;
videoConvertCtx = 0;
return true;
}
FFMpegVideoEncoder::FFMpegVideoEncoder()
{
d = new FFMpegVideoEncoderPriv();
}
FFMpegVideoEncoder::~FFMpegVideoEncoder()
{
delete d;
}
bool FFMpegVideoEncoder::close()
{
return d->close();
}
int FFMpegVideoEncoder::encodeImage( const QImage & img, qint64 time )
{
return d->encodeImage( img, time );
}
QString FFMpegVideoEncoder::createFile( const QString &filename,
const VideoEncodingProfile *profile,
const VideoFormat * videoformat,
unsigned int quality,
bool convert_audio,
AudioPlayer *audio )
{
d->m_aplayer = audio ? audio->impl() : 0;
d->m_profile = profile;
d->m_videoformat = videoformat;
d->m_convertaudio = convert_audio;
d->m_videobitrate = profile->bitratesVideo[quality] * 1000;
d->m_audiobitrate = profile->bitratesAudio[quality] * 1000;
if ( d->createFile( filename ) )
return QString();
return d->m_errorMsg;
}
void FFMpegVideoEncoderPriv::flush()
{
int err, got_output;
AVPacket pkt;
av_init_packet( &pkt );
// packet data will be allocated by the encoder - av_init_packet() does NOT do that!
pkt.data = 0;
pkt.size = 0;
// Get the delayed frames
for ( got_output = 1; got_output; )
{
if ( (err = avcodec_encode_audio2( audioCodecCtx, &pkt, 0, &got_output )) < 0 )
return;
if ( got_output )
{
// Set up the packet index
pkt.stream_index = audioStream->index;
// Newer ffmpeg versions do it anyway, but just in case
pkt.flags |= AV_PKT_FLAG_KEY;
if ( audioCodecCtx->coded_frame && audioCodecCtx->coded_frame->pts != AV_NOPTS_VALUE )
pkt.pts = av_rescale_q( audioCodecCtx->coded_frame->pts, audioCodecCtx->time_base, audioStream->time_base );
// And write the file
if ( (err = av_interleaved_write_frame( outputFormatCtx, &pkt )) < 0 )
{
qWarning("Failed to write the audio packet: error %d", err );
return;
}
outputTotalSize += pkt.size;
av_free_packet( &pkt );
}
}
}
bool FFMpegVideoEncoderPriv::createFile( const QString& fileName )
{
int err, size;
// If we had an open video, close it.
close();
av_log_set_level(AV_LOG_VERBOSE);
// Find the output container format
outputFormat = av_guess_format( qPrintable( m_profile->videoContainer ), qPrintable( m_profile->videoContainer ), 0 );
if ( !outputFormat )
{
m_errorMsg = QString("Could not guess the output format %1") .arg( m_profile->videoContainer );
goto cleanup;
}
// Allocate the output context
outputFormatCtx = avformat_alloc_context();
if ( !outputFormatCtx )
{
m_errorMsg = "Error allocating format context";
goto cleanup;
}
outputFormatCtx->oformat = outputFormat;
// Find the video encoder
videoCodec = avcodec_find_encoder_by_name( qPrintable(m_profile->videoCodec) );
if ( !videoCodec )
{
m_errorMsg = QString( "Video codec %1 is not found") .arg( m_profile->videoCodec );
goto cleanup;
}
// Allocate the video codec context
videoCodecCtx = avcodec_alloc_context3( videoCodec );
if ( !videoCodecCtx )
{
m_errorMsg = QString( "Context for video codec %1 cannot be allocated") .arg( m_profile->videoCodec );
goto cleanup;
}
// Set the video encoding parameters
videoCodecCtx->codec_type = AVMEDIA_TYPE_VIDEO;
videoCodecCtx->width = m_videoformat->width;
videoCodecCtx->height = m_videoformat->height;
videoCodecCtx->sample_aspect_ratio.den = m_videoformat->sample_aspect_den;
videoCodecCtx->sample_aspect_ratio.num = m_videoformat->sample_aspect_num;
videoCodecCtx->time_base.num = m_videoformat->frame_rate_num;
videoCodecCtx->time_base.den = m_videoformat->frame_rate_den;
videoCodecCtx->gop_size = (m_videoformat->frame_rate_den / m_videoformat->frame_rate_num) / 2; // GOP size is framerate / 2
videoCodecCtx->pix_fmt = AV_PIX_FMT_YUV420P;
videoCodecCtx->bit_rate = m_videobitrate;
videoCodecCtx->bit_rate_tolerance = m_videobitrate * av_q2d(videoCodecCtx->time_base);
// Set up the colorspace
if ( m_videoformat->colorspace == 601 )
videoCodecCtx->colorspace = ( 576 % videoCodecCtx->height ) ? AVCOL_SPC_SMPTE170M : AVCOL_SPC_BT470BG;
else if ( m_videoformat->colorspace == 709 )
videoCodecCtx->colorspace = AVCOL_SPC_BT709;
// Enable interlacing if needed
if ( m_videoformat->flags & VIFO_INTERLACED )
videoCodecCtx->flags |= CODEC_FLAG_INTERLACED_DCT;
// Enable multithreaded encoding: breaks FLV!
//videoCodecCtx->thread_count = 4;
// Video format-specific hacks
switch ( videoCodec->id )
{
case AV_CODEC_ID_H264:
av_opt_set( videoCodecCtx->priv_data, "preset", "slow", 0 );
break;
case AV_CODEC_ID_MPEG2VIDEO:
videoCodecCtx->max_b_frames = 2;
videoCodecCtx->bit_rate_tolerance = m_videobitrate * av_q2d(videoCodecCtx->time_base) * 2;
break;
case AV_CODEC_ID_MPEG1VIDEO:
// Needed to avoid using macroblocks in which some coeffs overflow.
videoCodecCtx->mb_decision = 2;
break;
default:
break;
}
// If we have a global header for the format, no need to duplicate the codec info in each keyframe
if ( outputFormatCtx->oformat->flags & AVFMT_GLOBALHEADER )
videoCodecCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
// Open the codec
if ( ( err = avcodec_open2( videoCodecCtx, videoCodec, 0 )) < 0 )
{
m_errorMsg = QString("Could not open video codec: error %1") .arg( err );
goto cleanup;
}
// Create the video stream, index
videoStream = avformat_new_stream( outputFormatCtx, videoCodecCtx->codec );
if ( !videoStream )
{
m_errorMsg = "Could not allocate video stream";
goto cleanup;
}
// Specify the coder for the stream
videoStream->codec = videoCodecCtx;
// Set the video stream timebase if not set
if ( videoStream->time_base.den == 0 )
videoStream->time_base = videoCodecCtx->time_base;
// Do we also have audio stream?
if ( m_aplayer )
{
// Are we copying the stream data?
if ( !m_convertaudio )
{
// Add the audio stream, index 1
audioStream = avformat_new_stream( outputFormatCtx, 0 );
if ( !audioStream )
{
m_errorMsg = "Could not allocate audio stream";
goto cleanup;
}
AVStream * origAudioStream = m_aplayer->pFormatCtx->streams[m_aplayer->audioStream];
audioStream->time_base = origAudioStream->time_base;
audioStream->disposition = origAudioStream->disposition;
AVCodecContext * newCtx = audioStream->codec;
// We're copying the stream
memcpy( newCtx, m_aplayer->aCodecCtx, sizeof(AVCodecContext) );
if ( newCtx->block_align == 1 && newCtx->codec_id == AV_CODEC_ID_MP3 )
newCtx->block_align= 0;
if ( newCtx->codec_id == AV_CODEC_ID_AC3 )
newCtx->block_align= 0;
}
else
{
// Find the audio codec
audioCodec = avcodec_find_encoder_by_name( qPrintable( m_profile->audioCodec ) );
if ( !audioCodec )
{
m_errorMsg = QString("Could not use the audio codec %1") .arg( m_profile->audioCodec );
goto cleanup;
}
// Hack to use the fixed AC3 codec if available
if ( audioCodec->id == AV_CODEC_ID_AC3 && avcodec_find_encoder_by_name( "ac3_fixed" ) )
audioCodec = avcodec_find_encoder_by_name( "ac3_fixed" );
// Allocate the audio context
audioCodecCtx = avcodec_alloc_context3( audioCodec );
if ( !audioCodecCtx )
{
m_errorMsg = QString( "Context for audio codec %1 cannot be allocated") .arg( m_profile->audioCodec );
goto cleanup;
}
audioCodecCtx->codec_id = audioCodec->id;
audioCodecCtx->codec_type = AVMEDIA_TYPE_AUDIO;
audioCodecCtx->bit_rate = m_audiobitrate;
audioCodecCtx->channels = m_profile->channels;
audioCodecCtx->sample_rate = m_profile->sampleRate;
audioCodecCtx->channel_layout = av_get_channel_layout( m_profile->channels == 1 ? "mono" : "stereo" );
audioCodecCtx->time_base.num = 1;
audioCodecCtx->time_base.den = m_profile->sampleRate;
if ( outputFormatCtx->oformat->flags & AVFMT_GLOBALHEADER )
audioCodecCtx->flags |= CODEC_FLAG_GLOBAL_HEADER;
// Since different audio codecs support different sample formats, look up which one is supported by this specific codec
if ( isAudioSampleFormatSupported( audioCodec->sample_fmts, AV_SAMPLE_FMT_FLTP ) )
{
qDebug("Audio format %s: using AV_SAMPLE_FMT_FLTP", qPrintable( m_profile->audioCodec ) );
audioCodecCtx->sample_fmt = AV_SAMPLE_FMT_FLTP;
}
else if ( isAudioSampleFormatSupported( audioCodec->sample_fmts, AV_SAMPLE_FMT_S16 ) )
{
qDebug("Audio format %s: using AV_SAMPLE_FMT_S16", qPrintable( m_profile->audioCodec ) );
audioCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16;
}
else if ( isAudioSampleFormatSupported( audioCodec->sample_fmts, AV_SAMPLE_FMT_S16P ) )
{
qDebug("Audio format %s: using AV_SAMPLE_FMT_S16P", qPrintable( m_profile->audioCodec ) );
audioCodecCtx->sample_fmt = AV_SAMPLE_FMT_S16P;
}
else
{
QString supported;
for ( const enum AVSampleFormat * fmt = audioCodec->sample_fmts; *fmt != AV_SAMPLE_FMT_NONE; fmt++ )
supported+= QString(" %1") .arg( *fmt );
m_errorMsg = QString("Could not find the sample format supported by the audio codec %1; supported: %2") . arg( m_profile->audioCodec ) .arg(supported);
goto cleanup;
}
// Open the audio codec
err = avcodec_open2( audioCodecCtx, audioCodec, 0 );
if ( err < 0 )
{
m_errorMsg = QString("Could not open the audio codec: %1") . arg( err );
goto cleanup;
}
// Allocate the audio stream
audioStream = avformat_new_stream( outputFormatCtx, audioCodec );
if ( !audioStream )
{
m_errorMsg = "Could not allocate audio stream";
goto cleanup;
}
// Remember the codec for the stream
audioStream->codec = audioCodecCtx;
// Setup the audio resampler
audioResampleCtx = avresample_alloc_context();
if ( !audioResampleCtx )
{
m_errorMsg = QString("Could not open the audio resampler");
goto cleanup;
}
// Some formats (i.e. WAV) do not produce the proper channel layout
if ( m_aplayer->aCodecCtx->channel_layout == 0 )
av_opt_set_int( audioResampleCtx, "in_channel_layout", av_get_channel_layout( m_profile->channels == 1 ? "mono" : "stereo" ), 0 );
else
av_opt_set_int( audioResampleCtx, "in_channel_layout", m_aplayer->aCodecCtx->channel_layout, 0 );
av_opt_set_int( audioResampleCtx, "in_sample_fmt", m_aplayer->aCodecCtx->sample_fmt, 0);
av_opt_set_int( audioResampleCtx, "in_sample_rate", m_aplayer->aCodecCtx->sample_rate, 0);
av_opt_set_int( audioResampleCtx, "in_channels", m_aplayer->aCodecCtx->channels,0);
av_opt_set_int( audioResampleCtx, "out_channel_layout", audioCodecCtx->channel_layout, 0);
av_opt_set_int( audioResampleCtx, "out_sample_fmt", audioCodecCtx->sample_fmt, 0);
av_opt_set_int( audioResampleCtx, "out_sample_rate", audioCodecCtx->sample_rate, 0);
av_opt_set_int( audioResampleCtx, "out_channels", audioCodecCtx->channels, 0);
err = avresample_open( audioResampleCtx );
if ( err < 0 )
{
m_errorMsg = QString("Could not open the audio resampler: %1") . arg( err );
goto cleanup;
}
audioFrame = av_frame_alloc();
if ( !audioFrame )
{
m_errorMsg = "Could not allocate audio frame";
goto cleanup;
}
audioFrame->nb_samples = audioStream->codec->frame_size;
audioFrame->format = audioStream->codec->sample_fmt;
audioFrame->channel_layout = audioStream->codec->channel_layout;
// Tthe codec gives us the frame size, in samples,we calculate the size of the samples buffer in bytes
audioSampleBuffer_size = av_samples_get_buffer_size( NULL, audioCodecCtx->channels, audioCodecCtx->frame_size, audioCodecCtx->sample_fmt, 0 );
audioSampleBuffer = (uint8_t*) av_malloc( audioSampleBuffer_size );
if ( !audioSampleBuffer )
{
m_errorMsg = "Could not allocate audio buffer";
goto cleanup;
}
// Setup the data pointers in the AVFrame
if ( avcodec_fill_audio_frame( audioFrame, audioStream->codec->channels, audioStream->codec->sample_fmt, (const uint8_t*) audioSampleBuffer, audioSampleBuffer_size, 0 ) < 0 )
{
m_errorMsg = "Could not set up audio frame";
goto cleanup;
}
if ( audioStream->codec->block_align == 1 && audioStream->codec->codec_id == AV_CODEC_ID_MP3 )
audioStream->codec->block_align= 0;
if ( audioStream->codec->codec_id == AV_CODEC_ID_AC3 )
audioStream->codec->block_align= 0;
}
// Rewind the audio player
m_aplayer->reset();
}
// Allocate the buffer for the picture
size = avpicture_get_size( videoCodecCtx->pix_fmt, videoCodecCtx->width, videoCodecCtx->height );
videoImageBuffer = new uint8_t[size];
if ( !videoImageBuffer )
{
m_errorMsg = "Could not open allocate picture buffer";
goto cleanup;
}
// Allocate the YUV frame
videoFrame = av_frame_alloc();
if ( !videoFrame )
{
m_errorMsg = "Could not open allocate picture frame buffer";
goto cleanup;
}
// Reset the PTS
videoFrame->pts = 0;
// Setup the planes
avpicture_fill( (AVPicture *)videoFrame, videoImageBuffer,videoCodecCtx->pix_fmt, videoCodecCtx->width, videoCodecCtx->height );
// Create the file and write the header
strncpy( outputFormatCtx->filename, FFMPEG_FILENAME( fileName ), sizeof(outputFormatCtx->filename) );
if ( avio_open( &outputFormatCtx->pb, FFMPEG_FILENAME( fileName ), AVIO_FLAG_WRITE) < 0 )
{
m_errorMsg = "Could not create the video file";
goto cleanup;
}
avformat_write_header( outputFormatCtx, 0 );
/*
// Dump output streams
for ( unsigned int i = 0; i < outputFormatCtx->nb_streams; i++)
{
qDebug( "Output stream %d: %s %.02f FPS, ", i, outputFormatCtx->streams[i]->codec->codec->name,
((double) outputFormatCtx->streams[i]->codec->time_base.den / outputFormatCtx->streams[i]->codec->time_base.num) );
if ( outputFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_VIDEO)
qDebug("width %d height %d bitrate %d\n", outputFormatCtx->streams[i]->codec->width,
outputFormatCtx->streams[i]->codec->height, outputFormatCtx->streams[i]->codec->bit_rate );
if ( outputFormatCtx->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO )
qDebug( "channels %d sample_rate %d bitrate %d\n", outputFormatCtx->streams[i]->codec->channels,
outputFormatCtx->streams[i]->codec->sample_rate,
outputFormatCtx->streams[i]->codec->bit_rate );
}
*/
videoFrameNumber = 0;
audioPTStracker = 0;
outputTotalSize = 0;
outputFileOpened = true;
return true;
cleanup:
close();
return false;
}
int FFMpegVideoEncoderPriv::encodeImage( const QImage &img, qint64 )
{
int err;
AVPacket pkt;
int got_packet;
// If we have audio, first add all audio packets for this time
if ( m_aplayer )
{
while ( true )
{
double audio_pts = (double) av_stream_get_end_pts(audioStream) * av_q2d( audioStream->time_base );
double video_pts = (double) av_stream_get_end_pts(videoStream) * av_q2d( videoStream->time_base );
//qDebug( "PTS check: A: %g V: %g", audio_pts, video_pts );
if ( video_pts < audio_pts )
break;
AVPacket pkt;
// Read a frame
if ( av_read_frame( m_aplayer->pFormatCtx, &pkt ) < 0 )
return false; // Frame read failed (e.g. end of stream)
if ( pkt.stream_index != m_aplayer->audioStream )
{
av_free_packet( &pkt );
continue;
}
// Initialize the frame
AVFrame *srcaudio = av_frame_alloc();
// Decode the original audio into the srcaudio frame
int got_audio;
err = avcodec_decode_audio4( m_aplayer->aCodecCtx, srcaudio, &got_audio, &pkt );
if ( err < 0 )
{
qWarning( "Error decoding audio frame: %d", err );
return -1;
}
// We don't need the AV packet anymore
av_free_packet( &pkt );
// Next packet if we didn't get audio
if ( !got_audio )
continue;
// Resample the input into the audioSampleBuffer until we proceed the whole decoded data
if ( (err = avresample_convert( audioResampleCtx,
NULL,
0,
0,
srcaudio->data,
0,
srcaudio->nb_samples )) < 0 )
{
qWarning( "Error resampling decoded audio: %d", err );
return -1;
}
while( avresample_available( audioResampleCtx ) >= audioFrame->nb_samples )
{
// Read a frame audio data from the resample fifo
if ( avresample_read( audioResampleCtx, audioFrame->data, audioFrame->nb_samples ) != audioFrame->nb_samples )
{
qWarning( "Error reading resampled audio: %d", err );
return -1;
}
// Prepare the packet
av_init_packet( &pkt );
// packet data will be allocated by the encoder - av_init_packet() does NOT do that!
pkt.data = 0;
pkt.size = 0;
// this only works for theora+vorbis
// if ( videoCodec->id == AV_CODEC_ID_THEORA && audioCodec->id == AV_CODEC_ID_VORBIS )
{
audioFrame->pts = av_rescale_q( audioPTStracker, (AVRational){1, audioCodecCtx->sample_rate}, audioCodecCtx->time_base);
audioPTStracker += audioFrame->nb_samples;
}
// and encode the audio into the audiopkt
if ( (err = avcodec_encode_audio2( audioCodecCtx, &pkt, audioFrame, &got_packet )) < 0 )
{
qWarning("Audio encoder failed with error %d", err );
return -1;
}
if ( got_packet )
{
// Set up the packet index
pkt.stream_index = audioStream->index;
// Newer ffmpeg versions do it anyway, but just in case
pkt.flags |= AV_PKT_FLAG_KEY;
// Rescale output packet timestamp values from codec to stream timebase
pkt.pts = av_rescale_q_rnd( pkt.pts, audioCodecCtx->time_base, audioStream->time_base, (AVRounding) (AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX) );
pkt.dts = av_rescale_q_rnd( pkt.dts, audioCodecCtx->time_base, audioStream->time_base, (AVRounding) (AV_ROUND_NEAR_INF|AV_ROUND_PASS_MINMAX) );
pkt.duration = av_rescale_q( pkt.duration, audioCodecCtx->time_base, audioStream->time_base);
// And write the file
if ( (err = av_interleaved_write_frame( outputFormatCtx, &pkt )) < 0 )
{
qWarning("Failed to write the audio packet: error %d", err );
return -1;
}
outputTotalSize += pkt.size;
av_free_packet( &pkt );
}
}
av_frame_free( &srcaudio );
}
}
// SWS conversion
convertImage_sws(img);
// Setup frame data
videoFrame->interlaced_frame = (m_videoformat->flags & VIFO_INTERLACED) ? 1 : 0;
videoFrame->pts = videoFrameNumber++;
//qDebug("Video time: %g", ((double) videoFrameNumber * videoCodecCtx->time_base.num) / videoCodecCtx->time_base.den );
av_init_packet( &pkt );
// packet data will be allocated by the encoder - av_init_packet() does NOT do that!
pkt.data = 0;
pkt.size = 0;
err = avcodec_encode_video2( videoCodecCtx, &pkt, videoFrame, &got_packet );
if ( err < 0 )
{
qWarning( "Error encoding video: %d", err );
return -1;
}
if ( got_packet )
{
// Convert the PTS from the packet base to stream base
if ( pkt.pts != AV_NOPTS_VALUE )
pkt.pts = av_rescale_q( pkt.pts, videoCodecCtx->time_base, videoStream->time_base );
// Convert the DTS from the packet base to stream base
if ( pkt.dts != AV_NOPTS_VALUE )
pkt.dts = av_rescale_q( pkt.dts, videoCodecCtx->time_base, videoStream->time_base );
if ( pkt.duration > 0 )
pkt.duration = av_rescale_q( pkt.duration, videoCodecCtx->time_base, videoStream->time_base );
if ( videoCodecCtx->coded_frame->key_frame )
pkt.flags |= AV_PKT_FLAG_KEY;
pkt.stream_index = videoStream->index;
err = av_interleaved_write_frame( outputFormatCtx, &pkt );
if ( err < 0 )
return -1;
outputTotalSize += pkt.size;
av_free_packet( &pkt );
}
return outputTotalSize;
}
/**
\brief Convert the QImage to the internal YUV format
SWS conversion
Caution: the QImage is allocated by QT without guarantee about the alignment and bytes per lines.
It *should* be okay as we make sure the image is a multiple of many bytes (8 or 16)...
... however it is not guaranteed that sws_scale won't at some point require more bytes per line.
We keep the custom conversion for that case.
**/
bool FFMpegVideoEncoderPriv::convertImage_sws(const QImage &img)
{
// Check if the image matches the size
if ( img.width() != (int) m_videoformat->width || img.height() != (int) m_videoformat->height )
{
printf("Wrong image size!\n");
return false;
}
if ( img.format()!=QImage::Format_RGB32 && img.format() != QImage::Format_ARGB32 )
{
printf("Wrong image format\n");
return false;
}
videoConvertCtx = sws_getCachedContext( videoConvertCtx,
m_videoformat->width,
m_videoformat->height,
AV_PIX_FMT_BGRA,
m_videoformat->width,
m_videoformat->height,
AV_PIX_FMT_YUV420P,
SWS_BICUBIC,
NULL,
NULL,
NULL );
if ( videoConvertCtx == NULL )
{
printf("Cannot initialize the conversion context\n");
return false;
}
uint8_t *srcplanes[3];
srcplanes[0]=(uint8_t*)img.bits();
srcplanes[1]=0;
srcplanes[2]=0;
int srcstride[3];
srcstride[0]=img.bytesPerLine();
srcstride[1]=0;
srcstride[2]=0;
sws_scale( videoConvertCtx, srcplanes, srcstride,0, m_videoformat->height, videoFrame->data, videoFrame->linesize);
return true;
}
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