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

On Tue, 30 May 2017 08:00:55 +0200, plevine457@gmail.com wrote:

From: Peter Levine plevine457@gmail.com

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

Could you give more information? As mentioned, the change looks more than a one-line summary.

This also breaks the build with older libraries, and it should be avoided as much as possible.

I don't mind to have another file if the change is too incompatible, but please try to keep the build with the old libav versions as of now.

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-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@gmail.com
• • based on rate converter that uses libsamplerate
• • Copyright (c) 2006 by Takashi Iwai tiwai@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