Fix building alsa-plugins against libav-10

From: Peter Levine <[email protected]>

Signed-off-by Peter Levine <[email protected]>

Makefile.am +6 -0

index 69cfe0d..9195b56 100644
@@ -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

configure.ac +17 -1

index f42601c..0af5ec9 100644
@@ -127,6 +127,10 @@ 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
   CFLAGS="$CFLAGS_saved"
 fi
 

@@ -135,6 +139,18 @@ 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)
+
 AC_ARG_ENABLE([speexdsp],
       AS_HELP_STRING([--disable-speexdsp], [Disable building of speexdsp plugin]))
 

@@ -217,7 +233,7 @@ AC_OUTPUT([
 	mix/Makefile
 	rate/Makefile
 	a52/Makefile
-	rate-lavc/Makefile
+	rate-lavr/Makefile
 	maemo/Makefile
 	doc/Makefile
 	usb_stream/Makefile

rate-lavr/Makefile.am new file +22 -0

new file mode 100644
index 0000000..a1dca35
@@ -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

rate-lavr/rate_lavr.c renamed from rate-lavc/rate_lavcrate.c +48 -112

similarity index 58%
copy from rate-lavc/rate_lavcrate.c
copy to rate-lavr/rate_lavr.c
index 14a2198..fe3bf4b 100644
@@ -1,9 +1,6 @@
 /*
- * Rate converter plugin using libavcodec's resampler
- * Copyright (c) 2007 by Nicholas Kain <[email protected]>
- *
- * based on rate converter that uses libsamplerate
- * Copyright (c) 2006 by Takashi Iwai <[email protected]>
+ * 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

@@ -19,21 +16,23 @@
 #include <stdio.h>
 #include <alsa/asoundlib.h>
 #include <alsa/pcm_rate.h>
-#include AVCODEC_HEADER
-#include "gcd.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 {
-	struct AVResampleContext *context;
+	AVAudioResampleContext *avr;
+
 	int in_rate;
 	int out_rate;
-	int stored;
-	int point;
-	int16_t **out;
-	int16_t **in;
 	unsigned int channels;
 };
 

@@ -50,26 +49,7 @@ static snd_pcm_uframes_t output_frames(void *obj, snd_pcm_uframes_t frames)
 static void pcm_src_free(void *obj)
 {
 	struct rate_src *rate = obj;
-	int i;
-
-	if (rate->out) {
-		for (i=0; i<rate->channels; i++) {
-			free(rate->out[i]);
-		}
-		free(rate->out);
-	}
-	if (rate->in) {
-		for (i=0; i<rate->channels; i++) {
-			free(rate->in[i]);
-		}
-		free(rate->in);
-	}
-	rate->out = rate->in = NULL;
-
-	if (rate->context) {
-		av_resample_close(rate->context);
-		rate->context = NULL;
-	}
+	avresample_free(&rate->avr);
 }
 
 static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info)

@@ -77,12 +57,14 @@ static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info)
 	struct rate_src *rate = obj;
 	int i, ir, or;
 
-	if (! rate->context || rate->channels != info->channels) {
+	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 = gcd(or, ir);
+		i = av_gcd(or, ir);
 		if (or > ir) {
 			phase_shift = or/i;
 		} else {

@@ -93,25 +75,27 @@ static int pcm_src_init(void *obj, snd_pcm_rate_info_t *info)
 			if (cutoff < 0.80)
 				cutoff = 0.80;
 		}
-		rate->context = av_resample_init(info->out.rate, info->in.rate,
-			filter_size, phase_shift,
-			(info->out.rate >= info->in.rate ? 0 : 1), cutoff);
-		if (!rate->context)
-			return -EINVAL;
-	}
 
-	rate->out = malloc(rate->channels * sizeof(int16_t *));
-	rate->in = malloc(rate->channels * sizeof(int16_t *));
-	for (i=0; i<rate->channels; i++) {
-		rate->out[i] = calloc(info->out.period_size * 2, 
-			sizeof(int16_t));
-		rate->in[i] = calloc(info->in.period_size * 2,
-			sizeof(int16_t));
-	}
-	rate->point = info->in.period_size / 2;
-	if (!rate->out || !rate->in) {
-		pcm_src_free(rate);
-		return -ENOMEM;
+		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;

@@ -129,48 +113,10 @@ static int pcm_src_adjust_pitch(void *obj, snd_pcm_rate_info_t *info)
 static void pcm_src_reset(void *obj)
 {
 	struct rate_src *rate = obj;
-	rate->stored = 0;
-}
 
-static void deinterleave(const int16_t *src, int16_t **dst, unsigned int frames,
-	unsigned int chans, int overflow)
-{
-	int i, j;
-
-	if (chans == 1) {
-		memcpy(dst + overflow, src, frames*sizeof(int16_t));
-	} else if (chans == 2) {
-		for (j=overflow; j<(frames + overflow); j++) {
-			dst[0][j] = *(src++);
-			dst[1][j] = *(src++);
-		}
-	} else {
-		for (j=overflow; j<(frames + overflow); j++) {
-			for (i=0; i<chans; i++) {
-				dst[i][j] = *(src++);
-			}
-		}
-	}
-}
-
-static void reinterleave(int16_t **src, int16_t *dst, unsigned int frames,
-	unsigned int chans)
-{
-	int i, j;
-
-	if (chans == 1) {
-		memcpy(dst, src, frames*sizeof(int16_t));
-	} else if (chans == 2) {
-		for (j=0; j<frames; j++) {
-			*(dst++) = src[0][j];
-			*(dst++) = src[1][j];
-		}
-	} else {
-		for (j=0; j<frames; j++) {
-			for (i=0; i<chans; i++) {
-				*(dst++) = src[i][j];
-			}
-		}
+	if (rate->avr) {
+		avresample_close(rate->avr);
+		avresample_open(rate->avr);
 	}
 }
 

@@ -179,22 +125,13 @@ static void pcm_src_convert_s16(void *obj, int16_t *dst, unsigned int
 {
 	struct rate_src *rate = obj;
 	int consumed = 0, chans=rate->channels, ret=0, i;
-	int total_in = rate->stored + src_frames, new_stored;
-
-	deinterleave(src, rate->in, src_frames, chans, rate->point);
-	for (i=0; i<chans; ++i) {	
-		ret = av_resample(rate->context, rate->out[i],
-				rate->in[i]+rate->point-rate->stored, &consumed,
-				total_in, dst_frames, i == (chans - 1));
-		new_stored = total_in-consumed;
-		memmove(rate->in[i]+rate->point-new_stored,
-				rate->in[i]+rate->point-rate->stored+consumed,
-				new_stored*sizeof(int16_t));
-	}
-	av_resample_compensate(rate->context,
-			total_in-src_frames>filter_size?0:1, src_frames);
-	reinterleave(rate->out, dst, ret, chans);
-	rate->stored = total_in-consumed;
+	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)

@@ -212,7 +149,7 @@ static int get_supported_rates(void *obj, unsigned int *rate_min,
 
 static void dump(void *obj, snd_output_t *out)
 {
-	snd_output_printf(out, "Converter: liblavc\n");
+	snd_output_printf(out, "Converter: libavr\n");
 }
 #endif
 

@@ -220,7 +157,6 @@ static snd_pcm_rate_ops_t pcm_src_ops = {
 	.close = pcm_src_close,
 	.init = pcm_src_init,
 	.free = pcm_src_free,
-	.reset = pcm_src_reset,
 	.adjust_pitch = pcm_src_adjust_pitch,
 	.convert_s16 = pcm_src_convert_s16,
 	.input_frames = input_frames,

@@ -248,7 +184,7 @@ int pcm_src_open(unsigned int version, void **objp, snd_pcm_rate_ops_t *ops)
 		return -ENOMEM;
 
 	*objp = rate;
-	rate->context = NULL;
+	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));

Signature · 1 line

2.13.0