reinit the tree, so we can have metadata

4 files changed, 379 insertions, 0 deletions

diff --git a/media-plugins/alsa-plugins/files/51-pulseaudio-probe.conf b/media-plugins/alsa-plugins/files/51-pulseaudio-probe.conf
new file mode 100644
index 000000000000..c2272c85b072
--- /dev/null
+++ b/media-plugins/alsa-plugins/files/51-pulseaudio-probe.conf
@@ -0,0 +1,19 @@
+# PulseAudio alsa plugin configuration file to set the pulseaudio plugin as
+# default output for applications using alsa when pulseaudio is running.
+
+hook_func.pulse_load_if_running {
+	lib "/usr/lib/alsa-lib/libasound_module_conf_pulse.so"
+	func "conf_pulse_hook_load_if_running"
+}
+
+@hooks [
+	{
+		func pulse_load_if_running
+		files [
+			"/usr/share/alsa/pulse-default.conf"
+			"/etc/asound.conf"
+			"~/.asoundrc"
+		]
+		errors false
+	}
+]
diff --git a/media-plugins/alsa-plugins/files/alsa-plugins-1.0.23-automagic.patch b/media-plugins/alsa-plugins/files/alsa-plugins-1.0.23-automagic.patch
new file mode 100644
index 000000000000..8e62f20a143d
--- /dev/null
+++ b/media-plugins/alsa-plugins/files/alsa-plugins-1.0.23-automagic.patch
@@ -0,0 +1,12 @@
+diff -uNr alsa-plugins-1.0.23.ORIg//Makefile.am alsa-plugins-1.0.23/Makefile.am
+--- alsa-plugins-1.0.23.ORIg//Makefile.am	2010-04-16 23:38:58.546243512 +0100
++++ alsa-plugins-1.0.23/Makefile.am	2010-04-16 23:39:20.049278487 +0100
+@@ -17,7 +17,7 @@
+ if HAVE_PPH
+ SUBDIRS += pph
+ endif
+-if HAVE_SPEEXDSP
++if USE_LIBSPEEX
+ SUBDIRS += speex
+ endif
+ 
diff --git a/media-plugins/alsa-plugins/files/alsa-plugins-1.0.28-libav10.patch b/media-plugins/alsa-plugins/files/alsa-plugins-1.0.28-libav10.patch
new file mode 100644
index 000000000000..9718369782b3
--- /dev/null
+++ b/media-plugins/alsa-plugins/files/alsa-plugins-1.0.28-libav10.patch
@@ -0,0 +1,338 @@
+https://bugs.gentoo.org/539680
+
+From: Luca Barbato <lu_zero@gentoo.org>
+Description: lavr: Add a libavresample based rate plugin
+Date: Mon, 14 Apr 2014 10:01:07 +0200
+
+Provide lavcrate compatibility.
+
+Index: alsa-plugins-1.0.28/configure.ac
+===================================================================
+--- alsa-plugins-1.0.28.orig/configure.ac
++++ alsa-plugins-1.0.28/configure.ac
+@@ -66,7 +66,7 @@ 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])
+@@ -101,6 +101,10 @@ 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$HAVE_AVCODEC = xyes)
+@@ -108,6 +112,18 @@ AC_SUBST(AVCODEC_CFLAGS)
+ AC_SUBST(AVCODEC_LIBS)
+ AC_SUBST(AVCODEC_HEADER)
+ 
++AC_ARG_ENABLE([avresample],
++      AS_HELP_STRING([--disable-avresample], [Do not build plugins depending on avcodec (lavrate)]))
++
++if test "x$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")
+ 
+@@ -181,7 +197,7 @@ AC_OUTPUT([
+ 	mix/Makefile
+ 	rate/Makefile
+ 	a52/Makefile
+-	rate-lavc/Makefile
++	rate-lavr/Makefile
+ 	maemo/Makefile
+ 	doc/Makefile
+ 	usb_stream/Makefile
+Index: alsa-plugins-1.0.28/Makefile.am
+===================================================================
+--- alsa-plugins-1.0.28.orig/Makefile.am
++++ alsa-plugins-1.0.28/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
+Index: alsa-plugins-1.0.28/rate-lavr/Makefile.am
+===================================================================
+--- /dev/null
++++ alsa-plugins-1.0.28/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
+Index: alsa-plugins-1.0.28/rate-lavr/rate_lavr.c
+===================================================================
+--- /dev/null
++++ alsa-plugins-1.0.28/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);
++}
++
++
diff --git a/media-plugins/alsa-plugins/files/pulse-default.conf b/media-plugins/alsa-plugins/files/pulse-default.conf
new file mode 100644
index 000000000000..8f7cbf29d60c
--- /dev/null
+++ b/media-plugins/alsa-plugins/files/pulse-default.conf
@@ -0,0 +1,10 @@
+# This file is referred to from files in /usr/share/alsa/alsa.conf.d/ in order
+# to set up the pulse device as the default if required.
+
+pcm.!default {
+	type pulse
+}
+
+ctl.!default {
+	type pulse
+}