On Tue, 30 May 2017 08:00:55 +0200,
plevine457@xxxxxxxxx wrote:
>
> From: Peter Levine <plevine457@xxxxxxxxx>
>
> Signed-off-by: Peter Levine <plevine457@xxxxxxxxx>
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@xxxxxxxxx>
> - *
> - * based on rate converter that uses libsamplerate
> - * Copyright (c) 2006 by Takashi Iwai <tiwai@xxxxxxx>
> + * 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
>
_______________________________________________
Alsa-devel mailing list
Alsa-devel@xxxxxxxxxxxxxxxx
http://mailman.alsa-project.org/mailman/listinfo/alsa-devel
