diff options
author | 2015-08-08 13:49:04 -0700 | |
---|---|---|
committer | 2015-08-08 17:38:18 -0700 | |
commit | 56bd759df1d0c750a065b8c845e93d5dfa6b549d (patch) | |
tree | 3f91093cdb475e565ae857f1c5a7fd339e2d781e /media-sound/audacity/files | |
download | gentoo-56bd759df1d0c750a065b8c845e93d5dfa6b549d.tar.gz gentoo-56bd759df1d0c750a065b8c845e93d5dfa6b549d.tar.bz2 gentoo-56bd759df1d0c750a065b8c845e93d5dfa6b549d.zip |
proj/gentoo: Initial commit
This commit represents a new era for Gentoo:
Storing the gentoo-x86 tree in Git, as converted from CVS.
This commit is the start of the NEW history.
Any historical data is intended to be grafted onto this point.
Creation process:
1. Take final CVS checkout snapshot
2. Remove ALL ChangeLog* files
3. Transform all Manifests to thin
4. Remove empty Manifests
5. Convert all stale $Header$/$Id$ CVS keywords to non-expanded Git $Id$
5.1. Do not touch files with -kb/-ko keyword flags.
Signed-off-by: Robin H. Johnson <robbat2@gentoo.org>
X-Thanks: Alec Warner <antarus@gentoo.org> - did the GSoC 2006 migration tests
X-Thanks: Robin H. Johnson <robbat2@gentoo.org> - infra guy, herding this project
X-Thanks: Nguyen Thai Ngoc Duy <pclouds@gentoo.org> - Former Gentoo developer, wrote Git features for the migration
X-Thanks: Brian Harring <ferringb@gentoo.org> - wrote much python to improve cvs2svn
X-Thanks: Rich Freeman <rich0@gentoo.org> - validation scripts
X-Thanks: Patrick Lauer <patrick@gentoo.org> - Gentoo dev, running new 2014 work in migration
X-Thanks: Michał Górny <mgorny@gentoo.org> - scripts, QA, nagging
X-Thanks: All of other Gentoo developers - many ideas and lots of paint on the bikeshed
Diffstat (limited to 'media-sound/audacity/files')
3 files changed, 240 insertions, 0 deletions
diff --git a/media-sound/audacity/files/audacity-1.3.13-automagic.patch b/media-sound/audacity/files/audacity-1.3.13-automagic.patch new file mode 100644 index 000000000000..4e7c2917e5db --- /dev/null +++ b/media-sound/audacity/files/audacity-1.3.13-automagic.patch @@ -0,0 +1,61 @@ +--- audacity-src-1.3.13-beta.orig/configure.in ++++ audacity-src-1.3.13-beta/configure.in +@@ -173,6 +173,9 @@ + lib_preference=$withval, + lib_preference="system local") + ++AC_ARG_WITH([alsa], AS_HELP_STRING([--without-alsa], [Build without alsa library (default: test)])) ++AC_ARG_WITH([jack], AS_HELP_STRING([--without-jack], [Build without jack library (default: test)])) ++ + dnl AC_ARG_WITH(wx-version, + dnl [AS_HELP_STRING([--with-wx-version], + dnl [select wxWidgets version (if both installed) [2.8,]])], +@@ -574,13 +577,21 @@ + ;; + *) + dnl Unix +- AC_CHECK_LIB(asound, snd_pcm_open, have_alsa=yes, have_alsa=no) +- if [[ $have_alsa = "yes" ]] ; then +- LIBS="$LIBS -lasound" +- fi +- PKG_CHECK_MODULES(JACK, jack, have_jack=yes, have_jack=no) +- if [[ $have_jack = "yes" ]] ; then +- LIBS="$LIBS $JACK_LIBS" ++ if test "x$with_alsa" != "xno"; then ++ AC_CHECK_LIB(asound, snd_pcm_open, have_alsa=yes, have_alsa=no) ++ if [[ $have_alsa = "yes" ]] ; then ++ LIBS="$LIBS -lasound" ++ else ++ AC_MSG_WARN([Support for alsa not available]) ++ fi ++ fi ++ if test "x$with_jack" != "xno"; then ++ PKG_CHECK_MODULES(JACK, jack, have_jack=yes, have_jack=no) ++ if [[ $have_jack = "yes" ]] ; then ++ LIBS="$LIBS $JACK_LIBS" ++ else ++ AC_MSG_WARN([Support for jack not available]) ++ fi + fi + AC_CHECK_LIB(hpi, HPI_SubSysCreate, have_asihpi=yes, have_asihpi=no, -lm) + if [[ $have_asihpi = "yes" ]] ; then +--- audacity-src-1.3.13-beta.orig/lib-src/portmixer/configure.ac ++++ audacity-src-1.3.13-beta/lib-src/portmixer/configure.ac +@@ -31,6 +31,8 @@ + [AC_SUBST( cflags, ["$cflags -g"] ) AC_MSG_RESULT(yes)], + [AC_SUBST( cflags, ["$cflags -O2"] ) AC_MSG_RESULT(no)]) + ++AC_ARG_WITH([alsa], AC_HELP_STRING([--without-alsa], [Build without alsa library (default: test)])) ++ + # + # Check for portaudio path + # +@@ -133,7 +135,7 @@ + have_support=yes + fi + +-if [[ $have_alsa = "yes" ]] ; then ++if [[ $have_alsa = "yes" -a "x$with_alsa" != "xno" ]] ; then + AC_MSG_NOTICE([Including support for ALSA]) + AC_DEFINE(PX_USE_LINUX_ALSA) + objects="$objects px_linux_alsa.o" diff --git a/media-sound/audacity/files/audacity-1.3.13-ffmpeg.patch b/media-sound/audacity/files/audacity-1.3.13-ffmpeg.patch new file mode 100644 index 000000000000..675470913c8d --- /dev/null +++ b/media-sound/audacity/files/audacity-1.3.13-ffmpeg.patch @@ -0,0 +1,164 @@ +--- audacity-src-1.3.13-beta.orig/src/export/ExportFFmpeg.cpp ++++ audacity-src-1.3.13-beta/src/export/ExportFFmpeg.cpp +@@ -352,7 +352,7 @@ + avcodec_get_context_defaults(mEncAudioCodecCtx); + + mEncAudioCodecCtx->codec_id = ExportFFmpegOptions::fmts[mSubFormat].codecid; +- mEncAudioCodecCtx->codec_type = CODEC_TYPE_AUDIO; ++ mEncAudioCodecCtx->codec_type = AVMEDIA_TYPE_AUDIO; + mEncAudioCodecCtx->codec_tag = av_codec_get_tag((const AVCodecTag **)mEncFormatCtx->oformat->codec_tag,mEncAudioCodecCtx->codec_id); + mSampleRate = (int)project->GetRate(); + mEncAudioCodecCtx->global_quality = -99999; //quality mode is off by default; +@@ -403,7 +403,6 @@ + mEncAudioCodecCtx->flags2 = 0; + if (gPrefs->Read(wxT("/FileFormats/FFmpegBitReservoir"),true)) mEncAudioCodecCtx->flags2 |= CODEC_FLAG2_BIT_RESERVOIR; + if (gPrefs->Read(wxT("/FileFormats/FFmpegVariableBlockLen"),true)) mEncAudioCodecCtx->flags2 |= 0x0004; //WMA only? +- mEncAudioCodecCtx->use_lpc = gPrefs->Read(wxT("/FileFormats/FFmpegUseLPC"),true); + mEncAudioCodecCtx->compression_level = gPrefs->Read(wxT("/FileFormats/FFmpegCompLevel"),-1); + mEncAudioCodecCtx->frame_size = gPrefs->Read(wxT("/FileFormats/FFmpegFrameSize"),(long)0); + mEncAudioCodecCtx->lpc_coeff_precision = gPrefs->Read(wxT("/FileFormats/FFmpegLPCCoefPrec"),(long)0); +@@ -569,7 +569,7 @@ + pkt.stream_index = mEncAudioStream->index; + pkt.data = mEncAudioEncodedBuf; + pkt.size = nEncodedBytes; +- pkt.flags |= PKT_FLAG_KEY; ++ pkt.flags |= AV_PKT_FLAG_KEY; + + // Set presentation time of frame (currently in the codec's timebase) in the stream timebase. + if(mEncAudioCodecCtx->coded_frame && mEncAudioCodecCtx->coded_frame->pts != int64_t(AV_NOPTS_VALUE)) +@@ -656,7 +656,7 @@ + + pkt.stream_index = mEncAudioStream->index; + pkt.data = mEncAudioEncodedBuf; +- pkt.flags |= PKT_FLAG_KEY; ++ pkt.flags |= AV_PKT_FLAG_KEY; + + // Write the encoded audio frame to the output file. + if ((ret = av_interleaved_write_frame(mEncFormatCtx, &pkt)) != 0) +--- audacity-src-1.3.13-beta.orig/src/export/ExportFFmpegDialogs.cpp ++++ audacity-src-1.3.13-beta/src/export/ExportFFmpegDialogs.cpp +@@ -1288,7 +1288,7 @@ + while ((codec = av_codec_next(codec))) + { + // We're only interested in audio and only in encoders +- if (codec->type == CODEC_TYPE_AUDIO && codec->encode) ++ if (codec->type == AVMEDIA_TYPE_AUDIO && codec->encode) + { + mCodecNames.Add(wxString::FromUTF8(codec->name)); + mCodecLongNames.Add(wxString::Format(wxT("%s - %s"),mCodecNames.Last().c_str(),wxString::FromUTF8(codec->long_name).c_str())); +@@ -1528,7 +1528,7 @@ + // Find the codec, that is claimed to be compatible + AVCodec *codec = avcodec_find_encoder(CompatibilityList[i].codec); + // If it exists, is audio and has encoder +- if (codec != NULL && (codec->type == CODEC_TYPE_AUDIO) && codec->encode) ++ if (codec != NULL && (codec->type == AVMEDIA_TYPE_AUDIO) && codec->encode) + { + // If it was selected - remember it's new index + if ((id >= 0) && codec->id == id) index = mShownCodecNames.GetCount(); +@@ -1543,7 +1543,7 @@ + AVCodec *codec = NULL; + while ((codec = av_codec_next(codec))) + { +- if (codec->type == CODEC_TYPE_AUDIO && codec->encode) ++ if (codec->type == AVMEDIA_TYPE_AUDIO && codec->encode) + { + if (mShownCodecNames.Index(wxString::FromUTF8(codec->name)) < 0) + { +@@ -1563,7 +1563,7 @@ + if (format != NULL) + { + AVCodec *codec = avcodec_find_encoder(format->audio_codec); +- if (codec != NULL && (codec->type == CODEC_TYPE_AUDIO) && codec->encode) ++ if (codec != NULL && (codec->type == AVMEDIA_TYPE_AUDIO) && codec->encode) + { + if ((id >= 0) && codec->id == id) index = mShownCodecNames.GetCount(); + mShownCodecNames.Add(wxString::FromUTF8(codec->name)); +--- audacity-src-1.3.13-beta.orig/src/FFmpeg.cpp ++++ audacity-src-1.3.13-beta/src/FFmpeg.cpp +@@ -316,7 +316,7 @@ + pd.buf_size = 0; + pd.buf = (unsigned char *) av_malloc(PROBE_BUF_MAX + AVPROBE_PADDING_SIZE); + if (pd.buf == NULL) { +- err = AVERROR_NOMEM; ++ err = AVERROR(ENOMEM); + goto fail; + } + +@@ -381,7 +381,7 @@ + + // Didn't find a suitable format, so bail + if (!fmt) { +- err = AVERROR_NOFMT; ++ err = AVERROR(EILSEQ); + goto fail; + } + +@@ -855,7 +855,6 @@ + FFMPEG_INITDYN(codec, avcodec_find_decoder); + FFMPEG_INITDYN(codec, avcodec_get_context_defaults); + FFMPEG_INITDYN(codec, avcodec_open); +- FFMPEG_INITDYN(codec, avcodec_decode_audio2); + FFMPEG_INITDYN(codec, avcodec_decode_audio3); + FFMPEG_INITDYN(codec, avcodec_encode_audio); + FFMPEG_INITDYN(codec, avcodec_close); +--- audacity-src-1.3.13-beta.orig/src/FFmpeg.h ++++ audacity-src-1.3.13-beta/src/FFmpeg.h +@@ -559,7 +559,11 @@ + FFMPEG_FUNCTION_WITH_RETURN( + void*, + av_fast_realloc, ++#if LIBAVUTIL_VERSION_MAJOR < 51 + (void *ptr, unsigned int *size, unsigned int min_size), ++#else ++ (void *ptr, unsigned int *size, size_t min_size), ++#endif + (ptr, size, min_size) + ); + FFMPEG_FUNCTION_WITH_RETURN( +@@ -747,7 +751,11 @@ + FFMPEG_FUNCTION_WITH_RETURN( + void*, + av_malloc, ++#if LIBAVUTIL_VERSION_MAJOR < 51 + (unsigned int size), ++#else ++ (size_t size), ++#endif + (size) + ); + FFMPEG_FUNCTION_NO_RETURN( +--- audacity-src-1.3.13-beta.orig/src/import/ImportFFmpeg.cpp ++++ audacity-src-1.3.13-beta/src/import/ImportFFmpeg.cpp +@@ -416,7 +416,7 @@ + // Fill the stream contexts + for (unsigned int i = 0; i < mFormatContext->nb_streams; i++) + { +- if (mFormatContext->streams[i]->codec->codec_type == CODEC_TYPE_AUDIO) ++ if (mFormatContext->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) + { + //Create a context + streamContext *sc = new streamContext; +--- audacity-src-1.3.13-beta.orig/src/ondemand/ODDecodeFFmpegTask.cpp ++++ audacity-src-1.3.13-beta/src/ondemand/ODDecodeFFmpegTask.cpp +@@ -156,7 +156,7 @@ + //test the audio stream(s) + for (unsigned int i = 0; i < ic->nb_streams; i++) + { +- if (ic->streams[i]->codec->codec_type == CODEC_TYPE_AUDIO) ++ if (ic->streams[i]->codec->codec_type == AVMEDIA_TYPE_AUDIO) + { + audioStreamExists = true; + st = ic->streams[i]; +@@ -573,10 +573,10 @@ + } + } + +- // avcodec_decode_audio2() expects the size of the output buffer as the 3rd parameter but ++ // avcodec_decode_audio3() expects the size of the output buffer as the 3rd parameter but + // also returns the number of bytes it decoded in the same parameter. + sc->m_decodedAudioSamplesValidSiz = sc->m_decodedAudioSamplesSiz; +- nBytesDecoded = avcodec_decode_audio2(sc->m_codecCtx, ++ nBytesDecoded = avcodec_decode_audio3(sc->m_codecCtx, + sc->m_decodedAudioSamples, // out + &sc->m_decodedAudioSamplesValidSiz, // in/out + pDecode, nDecodeSiz); // in diff --git a/media-sound/audacity/files/audacity-1.3.14-typecast.patch b/media-sound/audacity/files/audacity-1.3.14-typecast.patch new file mode 100644 index 000000000000..c01046143b21 --- /dev/null +++ b/media-sound/audacity/files/audacity-1.3.14-typecast.patch @@ -0,0 +1,15 @@ +--- audacity-src-1.3.14-beta/src/TrackPanel.cpp.orig ++++ audacity-src-1.3.14-beta/src/TrackPanel.cpp +@@ -2915,7 +2915,11 @@ + { + // Make sure we always have the first linked track of a stereo track + if (!mouseTrack->GetLinked() && mTracks->GetLink(mouseTrack)) +- mouseTrack = mTracks->GetLink(mouseTrack); ++ mouseTrack = ++#ifndef USE_MIDI ++ (WaveTrack*) ++#endif ++ mTracks->GetLink(mouseTrack); + + // Temporary apply the offset because we want to see if the + // track fits with the desired offset |