diff options
Diffstat (limited to 'media-sound/karlyriceditor/files/karlyriceditor-1.11-libav.patch')
| -rw-r--r-- | media-sound/karlyriceditor/files/karlyriceditor-1.11-libav.patch | 320 |
1 files changed, 320 insertions, 0 deletions
diff --git a/media-sound/karlyriceditor/files/karlyriceditor-1.11-libav.patch b/media-sound/karlyriceditor/files/karlyriceditor-1.11-libav.patch new file mode 100644 index 000000000000..6322c0708540 --- /dev/null +++ b/media-sound/karlyriceditor/files/karlyriceditor-1.11-libav.patch @@ -0,0 +1,320 @@ +From: =?utf-8?q?Martin_Stegh=C3=B6fer?= <martin@steghoefer.eu> +Date: Tue, 13 Nov 2012 20:19:11 +0100 +Subject: Add missing includes for libavutil +MIME-Version: 1.0 +Content-Type: text/plain; charset="utf-8" +Content-Transfer-Encoding: 8bit + +Author: Martin Steghöfer <martin@steghoefer.eu> +Bug: http://sourceforge.net/tracker/?func=detail&aid=3512390&group_id=290648&atid=1229531 +Forwarded: not-needed + +Added includes necessary for the use of the function "av_rescale_q". +Forwarding to upstream not needed because an equivalent patch has already been +posted to upstream's bug tracker (see URL in the "Bug" field). +--- + src/audioplayerprivate.cpp | 3 +++ + src/ffmpegvideoencoder.cpp | 3 +++ + 2 files changed, 6 insertions(+) + +diff --git a/src/audioplayerprivate.cpp b/src/audioplayerprivate.cpp +index ffff90e..1b6b32d 100644 +--- a/src/audioplayerprivate.cpp ++++ b/src/audioplayerprivate.cpp +@@ -21,6 +21,9 @@ + + #include "audioplayer.h" + #include "audioplayerprivate.h" ++extern "C" { ++#include "libavutil/mathematics.h" ++} + #include <SDL/SDL.h> + + // SDL defines its own main() function in SDL_main. And so does Qt, so if we continue without +diff --git a/src/ffmpegvideoencoder.cpp b/src/ffmpegvideoencoder.cpp +index 5734d2e..49182b5 100644 +--- a/src/ffmpegvideoencoder.cpp ++++ b/src/ffmpegvideoencoder.cpp +@@ -28,6 +28,9 @@ + #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 +From: =?utf-8?q?Martin_Stegh=C3=B6fer?= <martin@steghoefer.eu> +Date: Sat, 12 Apr 2014 15:19:48 +0200 +Subject: Fix compilation: FFmpeg/Libav +MIME-Version: 1.0 +Content-Type: text/plain; charset="utf-8" +Content-Transfer-Encoding: 8bit + +Author: Martin Steghöfer <martin@steghoefer.eu> +Forwarded: not-needed + +The upstream developers compile Karlyriceditor against FFmpeg, but in Debian +there is only Libav available, whose API is slowly drifting away from +FFmpeg's. This patch adapts the code to several of those differences: +* The second parameters of "avformat_new_stream" is of non-const pointer + type, but the code tries to pass a const pointer to it. +* There is no AV_ROUND_PASS_MINMAX flag for the rounding parameter of + av_rescale_q_rnd in Libav. +* The member r_frame_rate is no longer present in AVStream. The recommended + replacement is avg_frame_rate. +* The enums CODEC_ID_MP3 and CODEC_ID_AC3 have been prefixed with AV_. +* The function avcodec_alloc_frame was removed in favor of av_frame_alloc. + The corresponding deallocation function is av_frame_free (not the generic + av_free). +--- + src/audioplayerprivate.cpp | 4 ++-- + src/ffmpegvideodecoder.cpp | 8 ++++---- + src/ffmpegvideoencoder.cpp | 32 ++++++++++++++++++++------------ + 3 files changed, 26 insertions(+), 18 deletions(-) + +diff --git a/src/audioplayerprivate.cpp b/src/audioplayerprivate.cpp +index 1b6b32d..a6536cd 100644 +--- a/src/audioplayerprivate.cpp ++++ b/src/audioplayerprivate.cpp +@@ -118,7 +118,7 @@ void AudioPlayerPrivate::close() + } + + if ( m_frame ) +- av_free( m_frame ); ++ av_frame_free( &m_frame ); + + m_frame = 0; + pFormatCtx = 0; +@@ -227,7 +227,7 @@ bool AudioPlayerPrivate::open( const QString& filename ) + } + + // Allocate the buffer +- m_frame = avcodec_alloc_frame(); ++ m_frame = av_frame_alloc(); + + if ( !m_frame ) + { +diff --git a/src/ffmpegvideodecoder.cpp b/src/ffmpegvideodecoder.cpp +index 2ec5969..7820f72 100644 +--- a/src/ffmpegvideodecoder.cpp ++++ b/src/ffmpegvideodecoder.cpp +@@ -113,8 +113,8 @@ bool FFMpegVideoDecoder::openFile( const QString& filename, unsigned int seekto + if ( d->videoStream == -1 ) + return false; // Didn't find a video stream + +- d->m_fps_den = d->pFormatCtx->streams[d->videoStream]->r_frame_rate.den; +- d->m_fps_num = d->pFormatCtx->streams[d->videoStream]->r_frame_rate.num; ++ d->m_fps_den = d->pFormatCtx->streams[d->videoStream]->avg_frame_rate.den; ++ d->m_fps_num = d->pFormatCtx->streams[d->videoStream]->avg_frame_rate.num; + + if ( d->m_fps_den == 60000 ) + d->m_fps_den = 30000; +@@ -139,10 +139,10 @@ bool FFMpegVideoDecoder::openFile( const QString& filename, unsigned int seekto + } + + // Allocate video frame +- d->pFrame = avcodec_alloc_frame(); ++ d->pFrame = av_frame_alloc(); + + // Allocate an AVFrame structure +- d->pFrameRGB = avcodec_alloc_frame(); ++ d->pFrameRGB = av_frame_alloc(); + + if ( !d->pFrame || !d->pFrameRGB ) + { +diff --git a/src/ffmpegvideoencoder.cpp b/src/ffmpegvideoencoder.cpp +index 49182b5..bdf1730 100644 +--- a/src/ffmpegvideoencoder.cpp ++++ b/src/ffmpegvideoencoder.cpp +@@ -174,10 +174,10 @@ bool FFMpegVideoEncoderPriv::close() + delete[] audioSampleBuffer; + + if ( videoFrame ) +- av_free(videoFrame); ++ av_frame_free( &videoFrame ); + + if ( audioFrame ) +- av_free( audioFrame ); ++ av_frame_free( &audioFrame ); + + outputFormatCtx = 0; + outputFormat = 0; +@@ -383,7 +383,11 @@ av_log_set_level(AV_LOG_VERBOSE); + } + + // Create the video stream, index +- videoStream = avformat_new_stream( outputFormatCtx, videoCodecCtx->codec ); ++ // Use a block to keep the helper variable "codec" local to avoid conflict with gotos ++ { ++ AVCodec codec = *videoCodecCtx->codec; ++ videoStream = avformat_new_stream( outputFormatCtx, &codec ); ++ } + + if ( !videoStream ) + { +@@ -425,10 +429,10 @@ av_log_set_level(AV_LOG_VERBOSE); + // We're copying the stream + memcpy( newCtx, m_aplayer->aCodecCtx, sizeof(AVCodecContext) ); + +- if ( newCtx->block_align == 1 && newCtx->codec_id == CODEC_ID_MP3 ) ++ if ( newCtx->block_align == 1 && newCtx->codec_id == AV_CODEC_ID_MP3 ) + newCtx->block_align= 0; + +- if ( newCtx->codec_id == CODEC_ID_AC3 ) ++ if ( newCtx->codec_id == AV_CODEC_ID_AC3 ) + newCtx->block_align= 0; + } + else +@@ -443,7 +447,7 @@ av_log_set_level(AV_LOG_VERBOSE); + } + + // Hack to use the fixed AC3 codec if available +- if ( audioCodec->id == CODEC_ID_AC3 && avcodec_find_encoder_by_name( "ac3_fixed" ) ) ++ 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 +@@ -544,7 +548,7 @@ av_log_set_level(AV_LOG_VERBOSE); + goto cleanup; + } + +- audioFrame = avcodec_alloc_frame(); ++ audioFrame = av_frame_alloc(); + + if ( !audioFrame ) + { +@@ -573,10 +577,10 @@ av_log_set_level(AV_LOG_VERBOSE); + goto cleanup; + } + +- if ( audioStream->codec->block_align == 1 && audioStream->codec->codec_id == CODEC_ID_MP3 ) ++ if ( audioStream->codec->block_align == 1 && audioStream->codec->codec_id == AV_CODEC_ID_MP3 ) + audioStream->codec->block_align= 0; + +- if ( audioStream->codec->codec_id == CODEC_ID_AC3 ) ++ if ( audioStream->codec->codec_id == AV_CODEC_ID_AC3 ) + audioStream->codec->block_align= 0; + } + +@@ -595,7 +599,7 @@ av_log_set_level(AV_LOG_VERBOSE); + } + + // Allocate the YUV frame +- videoFrame = avcodec_alloc_frame(); ++ videoFrame = av_frame_alloc(); + + if ( !videoFrame ) + { +@@ -753,8 +757,12 @@ int FFMpegVideoEncoderPriv::encodeImage( const QImage &img, qint64 ) + 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) ); ++ if ( pkt.pts != AV_NOPTS_VALUE ) { ++ pkt.pts = av_rescale_q_rnd( pkt.pts, audioCodecCtx->time_base, audioStream->time_base, AV_ROUND_NEAR_INF ); ++ } ++ if ( pkt.dts != AV_NOPTS_VALUE ) { ++ pkt.dts = av_rescale_q_rnd( pkt.dts, audioCodecCtx->time_base, audioStream->time_base, AV_ROUND_NEAR_INF ); ++ } + pkt.duration = av_rescale_q( pkt.duration, audioCodecCtx->time_base, audioStream->time_base); + + // And write the file +From: =?utf-8?q?Martin_Stegh=C3=B6fer?= <martin@steghoefer.eu> +Date: Sat, 10 May 2014 01:04:56 +0200 +Subject: Fix segfault: AVFrame initialization (Libav/FFmpeg) +MIME-Version: 1.0 +Content-Type: text/plain; charset="utf-8" +Content-Transfer-Encoding: 8bit + +Author: Martin Steghöfer <martin@steghoefer.eu> +Forwarded: not-needed + +The upstream developers compile Karlyriceditor against FFmpeg, but in Debian +there is only Libav available, whose API is slowly drifting away from +FFmpeg's. This patch adapts the code to one of those differences. +In Libav the function avcodec_get_frame_defaults cannot be called on a +completely uninitialized object. It expects at least some pointers inside +the struct to be initialized to NULL. Otherwise (depending on the random +pointer value) it may perform a free() on that random pointer. In Libav the +preferred way to initialize an AVFrame object is by calling +avcodec_alloc_frame(), which allocates the memory, initializes it to zero +and then calls avcodec_get_frame_defaults itself. This involves changing +"srcaudio" from a stack object to a heap object and freeing it after use. +--- + src/ffmpegvideoencoder.cpp | 11 ++++++----- + 1 file changed, 6 insertions(+), 5 deletions(-) + +diff --git a/src/ffmpegvideoencoder.cpp b/src/ffmpegvideoencoder.cpp +index bdf1730..ccfdc7c 100644 +--- a/src/ffmpegvideoencoder.cpp ++++ b/src/ffmpegvideoencoder.cpp +@@ -685,12 +685,11 @@ int FFMpegVideoEncoderPriv::encodeImage( const QImage &img, qint64 ) + } + + // Initialize the frame +- AVFrame srcaudio; +- avcodec_get_frame_defaults( &srcaudio ); ++ 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 ); ++ err = avcodec_decode_audio4( m_aplayer->aCodecCtx, srcaudio, &got_audio, &pkt ); + + if ( err < 0 ) + { +@@ -710,9 +709,9 @@ int FFMpegVideoEncoderPriv::encodeImage( const QImage &img, qint64 ) + NULL, + 0, + 0, +- srcaudio.data, ++ srcaudio->data, + 0, +- srcaudio.nb_samples )) < 0 ) ++ srcaudio->nb_samples )) < 0 ) + { + qWarning( "Error resampling decoded audio: %d", err ); + return -1; +@@ -777,6 +776,8 @@ int FFMpegVideoEncoderPriv::encodeImage( const QImage &img, qint64 ) + av_free_packet( &pkt ); + } + } ++ ++ av_frame_free( &srcaudio ); + } + } + +From: =?utf-8?q?Martin_Stegh=C3=B6fer?= <martin@steghoefer.eu> +Date: Sat, 10 May 2014 13:19:25 +0200 +Subject: Fix FP exception: Sample aspect ratio (Libav/FFmpeg) +MIME-Version: 1.0 +Content-Type: text/plain; charset="utf-8" +Content-Transfer-Encoding: 8bit + +Author: Martin Steghöfer <martin@steghoefer.eu> +Forwarded: not-needed + +The upstream developers compile Karlyriceditor against FFmpeg, but in Debian +there is only Libav available, whose API is slowly drifting away from +FFmpeg's. This patch adapts the code to one of those differences. +In Libav sample aspect ratio of a AVStream object is not automatically +initialized to the one of its codec. So this has to be done manually. +--- + src/ffmpegvideoencoder.cpp | 1 + + 1 file changed, 1 insertion(+) + +diff --git a/src/ffmpegvideoencoder.cpp b/src/ffmpegvideoencoder.cpp +index ccfdc7c..3122f2a 100644 +--- a/src/ffmpegvideoencoder.cpp ++++ b/src/ffmpegvideoencoder.cpp +@@ -399,6 +399,7 @@ av_log_set_level(AV_LOG_VERBOSE); + videoStream->codec = videoCodecCtx; + + // Set the video stream timebase if not set ++ videoStream->sample_aspect_ratio = videoCodecCtx->sample_aspect_ratio; + if ( videoStream->time_base.den == 0 ) + videoStream->time_base = videoCodecCtx->time_base; + |