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forked from pool/audacity

Accepting request 109636 from multimedia:apps

- 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 (forwarded request 109608 from RedDwarf)

OBS-URL: https://build.opensuse.org/request/show/109636
OBS-URL: https://build.opensuse.org/package/show/openSUSE:Factory/audacity?expand=0&rev=51
This commit is contained in:
Stephan Kulow 2012-03-20 10:25:40 +00:00 committed by Git OBS Bridge
commit 0aca4d62b6
5 changed files with 58 additions and 189 deletions

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@ -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

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@ -1,3 +0,0 @@
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@ -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

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@ -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 \