[patch] alsa-plugins-1.0.28-libav10.patch

alsa-plugins-1.0.28-libav10.patch (text/plain), 9.84 KB, created by Russell Knighton on 2015-03-13 20:07:30 UTC

From: Luca Barbato <[e-mail]>
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);
+}
+
+