[alsa-devel] [PATCH - Fix building alsa-plugins against libav-10 1/1] Fix building alsa-plugins against libav-10
From: Peter Levine plevine457@gmail.com
Signed-off-by: Peter Levine plevine457@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@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));
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);@@ -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
participants (2)
-
plevine457＠gmail.com -
Takashi Iwai