diff --git a/audacity-1.3.13-ffmpeg.patch b/audacity-1.3.13-ffmpeg.patch deleted file mode 100644 index 30fa9e9..0000000 --- a/audacity-1.3.13-ffmpeg.patch +++ /dev/null @@ -1,164 +0,0 @@ ---- 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/audacity-minsrc-1.3.14-beta.tar.bz2 b/audacity-minsrc-1.3.14-beta.tar.bz2 deleted file mode 100644 index da79dcd..0000000 --- a/audacity-minsrc-1.3.14-beta.tar.bz2 +++ /dev/null @@ -1,3 +0,0 @@ -version https://git-lfs.github.com/spec/v1 -oid sha256:35990403cf078b862e7c839c73e5ef5f6b5160de1e3ccbd846e6153883194ab2 -size 6595572 diff --git a/audacity-minsrc-2.0.0.tar.bz2 b/audacity-minsrc-2.0.0.tar.bz2 new file mode 100644 index 0000000..21b26ad --- /dev/null +++ b/audacity-minsrc-2.0.0.tar.bz2 @@ -0,0 +1,3 @@ +version https://git-lfs.github.com/spec/v1 +oid sha256:d849288ca5f90764d497953e7d2744e269bfbb6f71840762c66419e4c68aeb6a +size 6625078 diff --git a/audacity.changes b/audacity.changes index c28162e..0349dbb 100644 --- a/audacity.changes +++ b/audacity.changes @@ -1,3 +1,46 @@ +------------------------------------------------------------------- +Wed Mar 14 15:06:03 UTC 2012 - reddwarf@opensuse.org + +- Update to version 2.0.0 + + * Bug fixes for: + + - Interface: + - Fixed playback speed and synchronization problems when dragging + clips or tracks between tracks having different sample rates. + + - Imports and Exports: + - Fixed crashes when changing the sample format of read-directly WAV + or AIFF files using the Track Drop-Down Menu. + - Fixed a crash importing MP3 files that had duplicate metadata tags + (this is a bug in current libsndfile which has been patched in + Audacity; MP3 files mislabeled as WAV which have duplicate tags + will still crash Audacity on Linux if Audacity has been compiled + against an affected version of system libsndfile). + - Fixed an issue where excessively high or corrupted sample values in + the audio could corrupt exports from the start of the problem for + the rest of the file, and could corrupt the rest of the project. + - Fixed Audacity could not be compiled against FFmpeg + 0.7.x and 0.8.x. + + - Effects and Analysis: + - Fixed crash on launch when using "Ambisonic Decoders (PC)" VST + plug-ins and other plug-ins that enable additional floating point + exceptions. + - Fixed Plot Spectrum background could be transparent on some machines. + - Bug fixes for Click Track, High Pass, Low Pass and Vocal Remover. + - Chirp, Tone and Silence generators now remember their settings. + + - Other miscellaneous bug fixes. + + * Changes and Improvements: + + - New Interface preference to show the track name in the display (this + is off by default). + - Longer default Playback preference for effects preview and preview + before cut. + - Restored use of Page Up and Page Down to scroll horizontally. + ------------------------------------------------------------------- Tue Dec 20 00:00:00 CET 2011 - detlef@links2linux.de diff --git a/audacity.spec b/audacity.spec index 619258b..6c93003 100644 --- a/audacity.spec +++ b/audacity.spec @@ -1,7 +1,7 @@ # # spec file for package audacity # -# Copyright (c) 2011 SUSE LINUX Products GmbH, Nuernberg, Germany. +# Copyright (c) 2012 SUSE LINUX Products GmbH, Nuernberg, Germany. # # All modifications and additions to the file contributed by third parties # remain the property of their copyright owners, unless otherwise agreed @@ -22,8 +22,6 @@ Name: audacity BuildRequires: alsa-devel -# we use internal PortAudio(so it is not included here in BuildRequires), because audacity team grab sources -# from Portaudio's svn more frequently than we (e.g. see support for non-mmap devices like pulseaudio) BuildRequires: flac-devel BuildRequires: gcc-c++ BuildRequires: jack-devel @@ -32,18 +30,19 @@ BuildRequires: libid3tag-devel BuildRequires: libsamplerate-devel BuildRequires: libsndfile-devel BuildRequires: libvorbis-devel +# This would require to patch our portaudio package with "PortMixer"... an extra API that never got integrated in PortAudio. +#BuildRequires: portaudio-devel BuildRequires: soundtouch-devel BuildRequires: taglib-devel BuildRequires: update-desktop-files %if %suse_version > 1130 BuildRequires: wxWidgets-wxcontainer-devel +%define _use_internal_dependency_generator 0 +%define __find_requires %wx_requires %else BuildRequires: wxGTK-devel %endif -#vamp-plugin-sdk-devel is available since openSUSE 11.2 -%if %suse_version > 1110 BuildRequires: vamp-plugin-sdk-devel -%endif %if %{with ffmpeg} BuildRequires: libffmpeg-devel %endif @@ -53,25 +52,19 @@ BuildRequires: libmad-devel %if %{with twolame} BuildRequires: libtwolame-devel %endif -Version: 1.3.14 -Release: 1 -License: GPL-2.0+ +Version: 2.0.0 +Release: 0 Summary: A Free, Cross-Platform Digital Audio Editor -Url: http://audacity.sourceforge.net/ +License: GPL-2.0+ Group: Productivity/Multimedia/Sound/Editors and Convertors -Source0: http://%{name}.googlecode.com/files/%{name}-minsrc-%{version}-beta.tar.bz2 +Url: http://audacity.sourceforge.net/ +Source0: http://%{name}.googlecode.com/files/%{name}-minsrc-%{version}.tar.bz2 Source1: %{name}.png Source2: %{name}-license-nyquist # PATCH-FIX-OPENSUSE %{name}-no_buildstamp.patch reddwarf@opensuse.org -- this patch removes the buildstamp Patch0: %{name}-no_buildstamp.patch Patch16: %{name}-implicit.patch -# PATCH-FIX-UPSTREAM %{name}-1.3.13-ffmpeg.patch reddwarf@opensuse.org -- make it compile against latest ffmpeg -Patch17: %{name}-1.3.13-ffmpeg.patch BuildRoot: %{_tmppath}/%{name}-%{version}-build -%if %suse_version > 1130 -%define _use_internal_dependency_generator 0 -%define __find_requires %wx_requires -%endif %description Audacity is a program that manipulates digital audio wave forms. In @@ -81,13 +74,10 @@ and Ogg Vorbis. With Audacity, you can edit wave data larger than the physical memory size of your computer. %prep -%setup -q -n %{name}-src-%{version}-beta +%setup -q -n %{name}-src-%{version} %patch0 %patch16 -%if %{with ffmpeg} -%patch17 -p1 -%endif -%{__cp} %{S:2} LICENSE_NYQUIST.txt +%{__cp} %{S:2} LICENSE_NYQUIST.txt %build %configure \