Dear maintainer,
Now that libav10 is in unstable, I've updated the NMU for alsa-plugins
(versioned as 1.0.27-2.1) and uploaded it to unstable. I've also
canceled the previous one last-minute. The reason was that I forgot to
add libavresample-dev as build depends to debian/control.
Please find the updated patch attached to this email.
Regards.
diff -Nru alsa-plugins-1.0.27/debian/changelog alsa-plugins-1.0.27/debian/changelog
--- alsa-plugins-1.0.27/debian/changelog 2013-05-09 06:40:50.000000000 -0400
+++ alsa-plugins-1.0.27/debian/changelog 2014-05-12 07:33:20.000000000 -0400
@@ -1,3 +1,10 @@
+alsa-plugins (1.0.27-2.1) unstable; urgency=medium
+
+ * Non-maintainer upload.
+ * Add libav10.patch to allow compilation against Libav10 (Closes: #739209)
+
+ -- Reinhard Tartler <siret...@tauware.de> Mon, 12 May 2014 07:33:19 -0400
+
alsa-plugins (1.0.27-2) unstable; urgency=low
* Upload to unstable.
diff -Nru alsa-plugins-1.0.27/debian/control alsa-plugins-1.0.27/debian/control
--- alsa-plugins-1.0.27/debian/control 2013-03-31 20:24:44.000000000 -0400
+++ alsa-plugins-1.0.27/debian/control 2014-05-12 07:35:20.000000000 -0400
@@ -9,6 +9,7 @@
dh-autoreconf,
libasound2-dev (>= 1.0.25),
libavcodec-dev,
+ libavresample-dev,
libavutil-dev,
libdbus-1-dev (>= 1.4.12-3~),
libjack-dev (>= 1:0.121.0+svn4538-2~),
diff -Nru alsa-plugins-1.0.27/debian/patches/libav10.patch alsa-plugins-1.0.27/debian/patches/libav10.patch
--- alsa-plugins-1.0.27/debian/patches/libav10.patch 1969-12-31 19:00:00.000000000 -0500
+++ alsa-plugins-1.0.27/debian/patches/libav10.patch 2014-05-12 07:30:30.000000000 -0400
@@ -0,0 +1,339 @@
+From: Luca Barbato <lu_z...@gentoo.org>
+Description: lavr: Add a libavresample based rate plugin
+Date: Mon, 14 Apr 2014 10:01:07 +0200
+
+Provide lavcrate compatibility.
+
+--- a/configure.in
++++ b/configure.in
+@@ -64,7 +64,7 @@ if test "$use_maemo_rm" = "yes"; then
+ fi
+
+ AC_ARG_ENABLE([avcodec],
+- AS_HELP_STRING([--disable-avcodec], [Don't build plugins depending on avcodec (a52)]))
++ AS_HELP_STRING([--disable-avcodec], [Do not build plugins depending on avcodec (a52)]))
+
+ if test "x$enable_avcodec" != "xno"; then
+ PKG_CHECK_MODULES(AVCODEC, [libavcodec libavutil], [HAVE_AVCODEC=yes], [HAVE_AVCODEC=no])
+@@ -99,6 +99,10 @@ if test $HAVE_AVCODEC = yes; then
+ if test -z "$AVCODEC_HEADER"; then
+ HAVE_AVCODEC=no
+ fi
++ SAVE_LIBS=$LIBS
++ LIBS="$LIBS $AVCODEC_LIBS"
++ AC_CHECK_FUNCS([av_resample_init])
++ LIBS=$SAVE_LIBS
+ fi
+
+ AM_CONDITIONAL(HAVE_AVCODEC, test x$HAVE_AVCODEC = xyes)
+@@ -106,6 +110,18 @@ AC_SUBST(AVCODEC_CFLAGS)
+ AC_SUBST(AVCODEC_LIBS)
+ AC_SUBST(AVCODEC_HEADER)
+
++AC_ARG_ENABLE([avresample],
++ AS_HELP_STRING([--disable-avresample], [Do not build plugins depending on avcodec (lavrate)]))
++
++if test "x$enable_avresample" != "xno"; then
++ PKG_CHECK_MODULES(AVRESAMPLE, [libavresample libavutil], [HAVE_AVRESAMPLE=yes], [HAVE_AVRESAMPLE=no])
++fi
++
++AM_CONDITIONAL(HAVE_AVRESAMPLE, test x$HAVE_AVCODEC = xyes)
++AC_SUBST(AVRESAMPLE_CFLAGS)
++AC_SUBST(AVRESAMPLE_LIBS)
++AC_SUBST(AVRESAMPLE_HEADER)
++
+ PKG_CHECK_MODULES(speexdsp, [speexdsp >= 1.2], [HAVE_SPEEXDSP="yes"], [HAVE_SPEEXDSP=""])
+ AM_CONDITIONAL(HAVE_SPEEXDSP, test "$HAVE_SPEEXDSP" = "yes")
+
+@@ -179,7 +195,7 @@ AC_OUTPUT([
+ mix/Makefile
+ rate/Makefile
+ a52/Makefile
+- rate-lavc/Makefile
++ rate-lavr/Makefile
+ maemo/Makefile
+ doc/Makefile
+ usb_stream/Makefile
+--- a/Makefile.am
++++ b/Makefile.am
+@@ -9,8 +9,14 @@ if HAVE_SAMPLERATE
+ SUBDIRS += rate
+ endif
+ if HAVE_AVCODEC
++SUBDIRS += a52
++if !HAVE_AVRESAMPLE
+ SUBDIRS += a52 rate-lavc
+ endif
++endif
++if HAVE_AVRESAMPLE
++SUBDIRS += rate-lavr
++endif
+ if HAVE_MAEMO_PLUGIN
+ SUBDIRS += maemo
+ endif
+--- /dev/null
++++ b/rate-lavr/Makefile.am
+@@ -0,0 +1,22 @@
++asound_module_rate_lavr_LTLIBRARIES = libasound_module_rate_lavr.la
++
++asound_module_rate_lavrdir = @ALSA_PLUGIN_DIR@
++
++AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ @AVRESAMPLE_CFLAGS@
++AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED)
++
++libasound_module_rate_lavr_la_SOURCES = rate_lavr.c
++libasound_module_rate_lavr_la_LIBADD = @ALSA_LIBS@ @AVRESAMPLE_LIBS@
++
++
++install-exec-hook:
++ rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so
++ $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate.so
++ $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_higher.so
++ $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_high.so
++ $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_fast.so
++ $(LN_S) libasound_module_rate_lavr.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_faster.so
++
++uninstall-hook:
++ rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate*.so
++ rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavr*.so
+--- /dev/null
++++ b/rate-lavr/rate_lavr.c
+@@ -0,0 +1,227 @@
++/*
++ * Rate converter plugin using libavresample
++ * Copyright (c) 2014 by Anton Khirnov
++ *
++ * This library is free software; you can redistribute it and/or
++ * modify it under the terms of the GNU Lesser General Public
++ * License as published by the Free Software Foundation; either
++ * version 2.1 of the License, or (at your option) any later version.
++ *
++ * This library 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
++ * Lesser General Public License for more details.
++ */
++
++#include <stdio.h>
++#include <alsa/asoundlib.h>
++#include <alsa/pcm_rate.h>
++
++#include <libavresample/avresample.h>
++#include <libavutil/channel_layout.h>
++#include <libavutil/opt.h>
++#include <libavutil/mathematics.h>
++#include <libavutil/samplefmt.h>
++
++
++static int filter_size = 16;
++static int phase_shift = 10; /* auto-adjusts */
++static double cutoff = 0; /* auto-adjusts */
++
++struct rate_src {
++ AVAudioResampleContext *avr;
++
++ int in_rate;
++ int out_rate;
++ unsigned int channels;
++};
++
++static snd_pcm_uframes_t input_frames(void *obj, snd_pcm_uframes_t frames)
++{
++ return frames;
++}
++
++static snd_pcm_uframes_t output_frames(void *obj, snd_pcm_uframes_t frames)
++{
++ return frames;
++}
++
++static void pcm_src_free(void *obj)
++{
++ struct rate_src *rate = obj;
++ avresample_free(&rate->avr);
++}
++
++static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info)
++{
++ struct rate_src *rate = obj;
++ int i, ir, or;
++
++ if (!rate->avr || rate->channels != info->channels) {
++ int ret;
++
++ pcm_src_free(rate);
++ rate->channels = info->channels;
++ ir = rate->in_rate = info->in.rate;
++ or = rate->out_rate = info->out.rate;
++ i = av_gcd(or, ir);
++ if (or > ir) {
++ phase_shift = or/i;
++ } else {
++ phase_shift = ir/i;
++ }
++ if (cutoff <= 0.0) {
++ cutoff = 1.0 - 1.0/filter_size;
++ if (cutoff < 0.80)
++ cutoff = 0.80;
++ }
++
++ rate->avr = avresample_alloc_context();
++ if (!rate->avr)
++ return -ENOMEM;
++
++ av_opt_set_int(rate->avr, "in_sample_rate", info->in.rate, 0);
++ av_opt_set_int(rate->avr, "out_sample_rate", info->out.rate, 0);
++ av_opt_set_int(rate->avr, "in_sample_format", AV_SAMPLE_FMT_S16, 0);
++ av_opt_set_int(rate->avr, "out_sample_format", AV_SAMPLE_FMT_S16, 0);
++ av_opt_set_int(rate->avr, "in_channel_layout", av_get_default_channel_layout(rate->channels), 0);
++ av_opt_set_int(rate->avr, "out_channel_layout", av_get_default_channel_layout(rate->channels), 0);
++
++ av_opt_set_int(rate->avr, "filter_size", filter_size, 0);
++ av_opt_set_int(rate->avr, "phase_shift", phase_shift, 0);
++ av_opt_set_double(rate->avr, "cutoff", cutoff, 0);
++
++ ret = avresample_open(rate->avr);
++ if (ret < 0) {
++ avresample_free(&rate->avr);
++ return -EINVAL;
++ }
++ }
++
++ return 0;
++}
++
++static int pcm_src_adjust_pitch(void *obj, snd_pcm_rate_info_t *info)
++{
++ struct rate_src *rate = obj;
++
++ if (info->out.rate != rate->out_rate || info->in.rate != rate->in_rate)
++ pcm_src_init(obj, info);
++ return 0;
++}
++
++static void pcm_src_reset(void *obj)
++{
++ struct rate_src *rate = obj;
++
++ if (rate->avr) {
++ avresample_close(rate->avr);
++ avresample_open(rate->avr);
++ }
++}
++
++static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int
++ dst_frames, const int16_t *src, unsigned int src_frames)
++{
++ struct rate_src *rate = obj;
++ int consumed = 0, chans=rate->channels, ret=0, i;
++ int total_in = avresample_get_delay(rate->avr) + src_frames;
++
++ ret = avresample_convert(rate->avr, &dst, dst_frames * chans * 2, dst_frames,
++ &src, src_frames * chans * 2, src_frames);
++
++ avresample_set_compensation(rate->avr,
++ total_in - src_frames > filter_size ? 0 : 1, src_frames);
++}
++
++static void pcm_src_close(void *obj)
++{
++ pcm_src_free(obj);
++}
++
++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
++static int get_supported_rates(void *obj, unsigned int *rate_min,
++ unsigned int *rate_max)
++{
++ *rate_min = *rate_max = 0; /* both unlimited */
++ return 0;
++}
++
++static void dump(void *obj, snd_output_t *out)
++{
++ snd_output_printf(out, "Converter: libavr\n");
++}
++#endif
++
++static snd_pcm_rate_ops_t pcm_src_ops = {
++ .close = pcm_src_close,
++ .init = pcm_src_init,
++ .free = pcm_src_free,
++ .adjust_pitch = pcm_src_adjust_pitch,
++ .convert_s16 = pcm_src_convert_s16,
++ .input_frames = input_frames,
++ .output_frames = output_frames,
++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
++ .version = SND_PCM_RATE_PLUGIN_VERSION,
++ .get_supported_rates = get_supported_rates,
++ .dump = dump,
++#endif
++};
++
++int pcm_src_open(unsigned int version, void **objp, snd_pcm_rate_ops_t *ops)
++
++{
++ struct rate_src *rate;
++
++#if SND_PCM_RATE_PLUGIN_VERSION < 0x010002
++ if (version != SND_PCM_RATE_PLUGIN_VERSION) {
++ fprintf(stderr, "Invalid rate plugin version %x\n", version);
++ return -EINVAL;
++ }
++#endif
++ rate = calloc(1, sizeof(*rate));
++ if (!rate)
++ return -ENOMEM;
++
++ *objp = rate;
++ rate->avr = NULL;
++#if SND_PCM_RATE_PLUGIN_VERSION >= 0x010002
++ if (version == 0x010001)
++ memcpy(ops, &pcm_src_ops, sizeof(snd_pcm_rate_old_ops_t));
++ else
++#endif
++ *ops = pcm_src_ops;
++ return 0;
++}
++
++int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate)(unsigned int version, void **objp,
++ snd_pcm_rate_ops_t *ops)
++{
++ return pcm_src_open(version, objp, ops);
++}
++int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_higher)(unsigned int version,
++ void **objp, snd_pcm_rate_ops_t *ops)
++{
++ filter_size = 64;
++ return pcm_src_open(version, objp, ops);
++}
++int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_high)(unsigned int version,
++ void **objp, snd_pcm_rate_ops_t *ops)
++{
++ filter_size = 32;
++ return pcm_src_open(version, objp, ops);
++}
++int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_fast)(unsigned int version,
++ void **objp, snd_pcm_rate_ops_t *ops)
++{
++ filter_size = 8;
++ return pcm_src_open(version, objp, ops);
++}
++int SND_PCM_RATE_PLUGIN_ENTRY(lavcrate_faster)(unsigned int version,
++ void **objp, snd_pcm_rate_ops_t *ops)
++{
++ filter_size = 4;
++ return pcm_src_open(version, objp, ops);
++}
++
++
+--- a/a52/pcm_a52.c
++++ b/a52/pcm_a52.c
+@@ -864,7 +864,7 @@ SND_PCM_PLUGIN_DEFINE_FUNC(a52)
+ if (rec->codec == NULL)
+ rec->codec = avcodec_find_encoder_by_name("ac3");
+ if (rec->codec == NULL)
+- rec->codec = avcodec_find_encoder(CODEC_ID_AC3);
++ rec->codec = avcodec_find_encoder(AV_CODEC_ID_AC3);
+ if (rec->codec == NULL) {
+ SNDERR("Cannot find codec engine");
+ err = -EINVAL;
diff -Nru alsa-plugins-1.0.27/debian/patches/series alsa-plugins-1.0.27/debian/patches/series
--- alsa-plugins-1.0.27/debian/patches/series 2013-04-26 17:01:28.000000000 -0400
+++ alsa-plugins-1.0.27/debian/patches/series 2014-05-12 07:30:30.000000000 -0400
@@ -1,2 +1,3 @@
arcam-av_uses_pthreads.patch
use_libavutil.patch
+libav10.patch
