Re: [alsa-devel] [PATCH - Fix building alsa-plugins against libav-10 1/1] Fix building alsa-plugins against libav-10

On Mon, 29 May 2017 20:39:55 +0200, plevine457@gmail.com wrote:

From: Peter Levine plevine457@gmail.com

Signed-off-by: Peter Levine plevine457@gmail.com

Hmm, the patch does what's far more than here written.

Can we avoid the file rename? The rename makes it a complete refresh, so essentially it look as you wrote everything from scratch.

thanks,

Takashi

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