[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-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); +} + +
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
participants (2)
-
plevine457＠gmail.com -
Takashi Iwai