[alsa-devel] [PATCH - Fix building alsa-plugins against libav-10 1/1] Fix building alsa-plugins against libav-10
plevine457 at gmail.com
plevine457 at gmail.com
Tue May 30 08:00:55 CEST 2017
From: Peter Levine <plevine457 at gmail.com>
Signed-off-by: Peter Levine <plevine457 at gmail.com>
diff --git a/Makefile.am b/Makefile.am
index 69cfe0d..9195b56 100644
--- 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
diff --git a/configure.ac b/configure.ac
index f42601c..0af5ec9 100644
--- a/configure.ac
+++ b/configure.ac
@@ -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
diff --git a/rate-lavr/Makefile.am b/rate-lavr/Makefile.am
new file mode 100644
index 0000000..a1dca35
--- /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
diff --git a/rate-lavc/rate_lavcrate.c b/rate-lavr/rate_lavr.c
similarity index 58%
copy from rate-lavc/rate_lavcrate.c
copy to rate-lavr/rate_lavr.c
index 14a2198..fe3bf4b 100644
--- a/rate-lavc/rate_lavcrate.c
+++ b/rate-lavr/rate_lavr.c
@@ -1,9 +1,6 @@
/*
- * Rate converter plugin using libavcodec's resampler
- * Copyright (c) 2007 by Nicholas Kain <njkain at gmail.com>
- *
- * based on rate converter that uses libsamplerate
- * Copyright (c) 2006 by Takashi Iwai <tiwai at suse.de>
+ * 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));
--
2.13.0
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