if HAVE_A52
SUBDIRS += a52
endif
-if HAVE_AVCRATE
-SUBDIRS += rate-lavc
+if HAVE_LAVRATE
+SUBDIRS += rate-lav
endif
if HAVE_USBSTREAM
SUBDIRS += usb_stream
asound_module_pcm_a52dir = @ALSA_PLUGIN_DIR@
asound_module_addondir = @ALSA_ADDON_DIR@
-AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ @AVCODEC_CFLAGS@ \
- -DAVCODEC_HEADER="@AVCODEC_HEADER@"
+AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ @LIBAV_CFLAGS@
AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED)
libasound_module_pcm_a52_la_SOURCES = pcm_a52.c
-libasound_module_pcm_a52_la_LIBADD = @ALSA_LIBS@ @AVCODEC_LIBS@
+libasound_module_pcm_a52_la_LIBADD = @ALSA_LIBS@ @LIBAV_LIBS@ @LIBAV_CODEC_LIBS@
#include <alsa/asoundlib.h>
#include <alsa/pcm_external.h>
#include <alsa/pcm_plugin.h>
-#include AVCODEC_HEADER
+#include <libavcodec/avcodec.h>
#include <libavutil/avutil.h>
/* some compatibility wrappers */
fi
fi
-AC_ARG_ENABLE([avcodec],
- AS_HELP_STRING([--disable-avcodec], [Don't build plugins depending on avcodec (a52)]))
+AC_ARG_ENABLE([libav],
+ AS_HELP_STRING([--disable-avlib], [Do not build plugins depending on libav/ffmpeg (a52,lavrate...)]))
-if test "x$enable_avcodec" != "xno"; then
- PKG_CHECK_MODULES(AVCODEC, [libavcodec libavutil], [HAVE_AVCODEC=yes], [HAVE_AVCODEC=no])
+if test "x$enable_libav" != "xno"; then
+ PKG_CHECK_MODULES(LIBAV, [libavcodec libavutil libavresample], [HAVE_LIBAV=yes], [HAVE_LIBAV=no])
fi
-if test "x$HAVE_AVCODEC" = "xno"; then
- AC_ARG_WITH([avcodec-includedir],
- AS_HELP_STRING([--with-avcodec-includedir=dir],
- [AVcodec include directory]),
- [AVCODEC_CFLAGS="-I$withval"], [AVCODEC_CFLAGS=""])
- AC_ARG_WITH([avcodec-libdir],
- AS_HELP_STRING([--with-avcodec-libdir=dir],
- [AVcodec library directory]),
- [AVCODEC_LIBS="-L$withval"], [AVCODEC_LIBS=""])
+if test "x$HAVE_LIBAV" = "xno"; then
+ AC_ARG_WITH([libav-includedir],
+ AS_HELP_STRING([--with-libav-includedir=dir],
+ [Libav/ffmpeg include directory]),
+ [LIBAV_CFLAGS="-I$(withval)"], [LIBAV_CFLAGS=""])
+ AC_ARG_WITH([libav-libdir],
+ AS_HELP_STRING([--with-libav-libdir=dir],
+ [Libav/ffmpeg library directory]),
+ [LIBAV_LIBS="-L$withval"], [LIBAV_LIBS=""])
CFLAGS_saved="$CFLAGS"
LDFLAGS_saved="$LDFLAGS"
- CFLAGS="$CFLAGS $AVCODEC_CFLAGS"
- LDFLAGS="$LDFLAGS $AVCODEC_LIBS"
- AVCODEC_LIBS="$AVCODEC_LIBS -lavcodec"
- AC_CHECK_LIB([avcodec], [avcodec_open], [HAVE_AVCODEC=yes], [HAVE_AVCODEC=no])
+ CFLAGS="$CFLAGS $LIBAV_CFLAGS"
+ LDFLAGS="$LDFLAGS $LIBAV_LIBS"
+ AC_CHECK_LIB([avcodec], [avcodec_open], [HAVE_LIBAV=yes], [HAVE_LIBAV=no])
CFLAGS="$CFLAGS_saved"
LDFLAGS="$LDFLAGS_saved"
+ LIBAV_CODEC_LIBS="-lavcodec"
+ LIBAV_RESAMPLE_LIBS="-lavresample -lavutil"
fi
-if test $HAVE_AVCODEC = yes; then
- AVCODEC_HEADER=""
- CFLAGS_saved="$CFLAGS"
- CFLAGS="$CFLAGS $AVCODEC_CFLAGS"
- AC_CHECK_HEADER([ffmpeg/avcodec.h], [AVCODEC_HEADER='<ffmpeg/avcodec.h>'])
- if test -z "$AVCODEC_HEADER"; then
- AC_CHECK_HEADER([libavcodec/avcodec.h], [AVCODEC_HEADER='<libavcodec/avcodec.h>'])
- fi
- if test -z "$AVCODEC_HEADER"; then
- HAVE_AVCODEC=no
- fi
- CFLAGS="$CFLAGS_saved"
-fi
-
-AM_CONDITIONAL(HAVE_AVCODEC, test x$HAVE_AVCODEC = xyes)
-AC_SUBST(AVCODEC_CFLAGS)
-AC_SUBST(AVCODEC_LIBS)
-AC_SUBST(AVCODEC_HEADER)
+AM_CONDITIONAL(HAVE_LIBAV, test x$HAVE_LIBAV = xyes)
+AC_SUBST(LIBAV_CFLAGS)
+AC_SUBST(LIBAV_LIBS)
+AC_SUBST(LIBAV_CODEC_LIBS)
+AC_SUBST(LIBAV_RESAMPLE_LIBS)
AC_ARG_ENABLE([a52],
AS_HELP_STRING([--disable-a52], [Disable building of A52 encoder plugin]))
-if test "x$enable_a52" != "xno" -a "$HAVE_AVCODEC" = "yes"; then
+if test "x$enable_a52" != "xno" -a "$HAVE_LIBAV" = "yes"; then
HAVE_A52=yes
fi
AM_CONDITIONAL(HAVE_A52, test x$HAVE_A52 = xyes)
-AC_ARG_ENABLE([avcrate],
- AS_HELP_STRING([--disable-avcrate], [Disable building of AVC rate plugin]))
+AC_ARG_ENABLE([lavrate],
+ AS_HELP_STRING([--disable-lavrate], [Disable building of libav/ffmpeg rate plugin]))
-if test "x$enable_avcrate" != "xno" -a "$HAVE_AVCODEC" = "yes"; then
- HAVE_AVCRATE=yes
+if test "x$enable_lavrate" != "xno" -a "$HAVE_LIBAV" = "yes"; then
+ HAVE_LAVRATE=yes
fi
-AM_CONDITIONAL(HAVE_AVCRATE, test x$HAVE_AVCRATE = xyes)
+AM_CONDITIONAL(HAVE_LAVRATE, test x$HAVE_LAVRATE = xyes)
AC_ARG_ENABLE([speexdsp],
AS_HELP_STRING([--disable-speexdsp], [Disable building of speexdsp plugin]))
mix/Makefile
rate/Makefile
a52/Makefile
- rate-lavc/Makefile
+ rate-lav/Makefile
maemo/Makefile
doc/Makefile
usb_stream/Makefile
fi
echo "Maemo plugin: $HAVE_MAEMO_PLUGIN"
echo " Using Osso resource manager: $use_maemo_rm"
-if test "$HAVE_AVCODEC" = "yes"; then
- echo "AVCodec config:"
- echo " AVCODEC_CFLAGS: $AVCODEC_CFLAGS"
- echo " AVCODEC_LIBS: $AVCODEC_LIBS"
- echo " AVCODEC_HEADER: $AVCODEC_HEADER"
+if test "$HAVE_LIBAV" = "yes"; then
+ echo "Libav/ffmpeg config:"
+ echo " LIBAV_CFLAGS: $LIBAV_CFLAGS"
+ echo " LIBAV_LIBS: $LIBAV_LIBS / $LIBAV_CODEC_LIBS / $LIBAV_RESAMPLE_LIBS"
fi
-echo "A52 plugin: $HAVE_A52"
-echo "AVC rate plugin: $HAVE_AVCRATE"
+echo "Libav A52 plugin: $HAVE_A52"
+echo "Libav rate plugin: $HAVE_LAVRATE"
echo "Speex rate plugin: $PPH"
echo "Speex preprocess plugin: $HAVE_SPEEXDSP"
if test "$HAVE_SPEEX" = "yes"; then
if HAVE_A52
EXTRA_DIST += a52.txt
endif
-if HAVE_AVCRATE
-EXTRA_DIST += lavcrate.txt
+if HAVE_LAVRATE
+EXTRA_DIST += lavrate.txt
endif
if HAVE_PPH
EXTRA_DIST += speexrate.txt
-Rate Converter Plugin Using libavcodec
-======================================
+Rate Converter Plugin Using libavresample
+=========================================0
-The plugin in rate-lavc subdirectory is an external rate converter using
-libavcodec's resampler. You can use this rate converter plugin by defining a
+The plugin in rate-lavr subdirectory is an external rate converter using
+libavresample library. You can use this rate converter plugin by defining a
rate PCM with "converter" parameter, such as:
pcm.my_rate {
type rate
slave.pcm "hw"
- converter "lavcrate"
+ converter "lavrate"
}
The plug plugin has also a similar field, "rate_converter".
-pcm.lavcrate {
+pcm.lavrate {
@args [ SLAVE RATE CONVERTER ]
@args.SLAVE {
type string
}
@args.CONVERTER {
type string
- default "lavcrate"
+ default "lavrate"
}
type rate
converter $CONVERTER
@func refer
name defaults.namehint.basic
}
- description "Rate Converter Plugin Using AVC Library"
+ description "Rate Converter Plugin Using Libav/FFmpeg Library"
}
}
--- /dev/null
+EXTRA_DIST = 10-rate-lav.conf
+
+asound_module_rate_lavrate_LTLIBRARIES = libasound_module_rate_lavrate.la
+asound_module_addon_DATA = 10-rate-lav.conf
+
+asound_module_rate_lavratedir = @ALSA_PLUGIN_DIR@
+asound_module_addondir = @ALSA_ADDON_DIR@
+
+AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ @LIBAV_CFLAGS@
+AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED)
+
+libasound_module_rate_lavrate_la_SOURCES = rate_lavrate.c
+libasound_module_rate_lavrate_la_LIBADD = @ALSA_LIBS@ @LIBAV_LIBS@ @LIBAV_RESAMPLE_LIBS@
+
+noinst_HEADERS = gcd.h
+
+install-exec-hook:
+ rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavrate_*.so
+ $(LN_S) libasound_module_rate_lavrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavrate_higher.so
+ $(LN_S) libasound_module_rate_lavrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavrate_high.so
+ $(LN_S) libasound_module_rate_lavrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavrate_fast.so
+ $(LN_S) libasound_module_rate_lavrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavrate_faster.so
+
+uninstall-hook:
+ rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavrate_*.so
--- /dev/null
+/*
+ * 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 unsigned int filter_size = 16;
+static unsigned int phase_shift = 10; /* auto-adjusts */
+static double cutoff = 0; /* auto-adjusts */
+
+struct rate_src {
+ AVAudioResampleContext *avr;
+
+ unsigned int in_rate;
+ unsigned int out_rate;
+ unsigned int channels;
+};
+
+static snd_pcm_uframes_t input_frames(void *obj ATTRIBUTE_UNUSED,
+ snd_pcm_uframes_t frames)
+{
+ return frames;
+}
+
+static snd_pcm_uframes_t output_frames(void *obj ATTRIBUTE_UNUSED,
+ 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) {
+#if 0
+ avresample_close(rate->avr);
+ avresample_open(rate->avr);
+#endif
+ }
+}
+
+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 chans = rate->channels;
+ unsigned int total_in = avresample_get_delay(rate->avr) + src_frames;
+
+ avresample_convert(rate->avr, (uint8_t **)&dst, dst_frames * chans * 2, dst_frames,
+ (uint8_t **)&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 ATTRIBUTE_UNUSED,
+ unsigned int *rate_min,
+ unsigned int *rate_max)
+{
+ *rate_min = *rate_max = 0; /* both unlimited */
+ return 0;
+}
+
+static void dump(void *obj ATTRIBUTE_UNUSED, 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,
+ .reset = pcm_src_reset,
+ .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(lavrate)(unsigned int version, void **objp,
+ snd_pcm_rate_ops_t *ops)
+{
+ return pcm_src_open(version, objp, ops);
+}
+int SND_PCM_RATE_PLUGIN_ENTRY(lavrate_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(lavrate_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(lavrate_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(lavrate_faster)(unsigned int version,
+ void **objp, snd_pcm_rate_ops_t *ops)
+{
+ filter_size = 4;
+ return pcm_src_open(version, objp, ops);
+}
+
+
+++ /dev/null
-EXTRA_DIST = 10-rate-lavc.conf
-
-asound_module_rate_lavcrate_LTLIBRARIES = libasound_module_rate_lavcrate.la
-asound_module_addon_DATA = 10-rate-lavc.conf
-
-asound_module_rate_lavcratedir = @ALSA_PLUGIN_DIR@
-asound_module_addondir = @ALSA_ADDON_DIR@
-
-AM_CFLAGS = -Wall -g @ALSA_CFLAGS@ @AVCODEC_CFLAGS@ \
- -DAVCODEC_HEADER="@AVCODEC_HEADER@"
-AM_LDFLAGS = -module -avoid-version -export-dynamic -no-undefined $(LDFLAGS_NOUNDEFINED)
-
-libasound_module_rate_lavcrate_la_SOURCES = rate_lavcrate.c
-libasound_module_rate_lavcrate_la_LIBADD = @ALSA_LIBS@ @AVCODEC_LIBS@
-
-noinst_HEADERS = gcd.h
-
-install-exec-hook:
- rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_*.so
- $(LN_S) libasound_module_rate_lavcrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_higher.so
- $(LN_S) libasound_module_rate_lavcrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_high.so
- $(LN_S) libasound_module_rate_lavcrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_fast.so
- $(LN_S) libasound_module_rate_lavcrate.so $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_faster.so
-
-uninstall-hook:
- rm -f $(DESTDIR)@ALSA_PLUGIN_DIR@/libasound_module_rate_lavcrate_*.so
+++ /dev/null
-/*
- * 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>
- *
- * 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 AVCODEC_HEADER
-#include "gcd.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;
- int in_rate;
- int out_rate;
- int stored;
- int point;
- int16_t **out;
- int16_t **in;
- 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;
- 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;
- }
-}
-
-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) {
- 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);
- 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->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;
- }
-
- 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;
- 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];
- }
- }
- }
-}
-
-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 = 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;
-}
-
-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: liblavc\n");
-}
-#endif
-
-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,
- .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->context = 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);
-}
-
-
git://git.alsa-project.org / alsa-plugins.git / commitdiff