[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
Mon May 29 20:39:55 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-lavr/rate_lavr.c b/rate-lavr/rate_lavr.c
new file mode 100644
index 0000000..fe3bf4b
--- /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);
+}
+
+
--
2.13.0
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