On Mon, 29 May 2017 20:39:55 +0200,
plevine457@xxxxxxxxx wrote:
>
> From: Peter Levine <plevine457@xxxxxxxxx>
>
> Signed-off-by: Peter Levine <plevine457@xxxxxxxxx>
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
>
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