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rawpatchinlineside by side (parent: 11e9755)
rawpatchinlineside by side (parent: 11e9755)
| author | Stefano Pigozzi <[email protected]> | |
| Sat, 9 Mar 2013 08:30:26 +0000 (9 09:30 +0100) | ||
| committer | Uoti Urpala <[email protected]> | |
| Wed, 9 Oct 2013 20:43:25 +0000 (9 23:43 +0300) |
Remove `af_resample` and `af_lavcresample`. The former is a mess while
the latter uses an API that was long deprecated in libavcodec and is
now removed. Add new filter af_lavrresample which uses libavresample.
`af_lavrresample` has roughly the same features and structure as
`af_lavcresample` had.
Libavresample is now a mandatory dependency. Change configure script
accordingly and remove #ifdefs in other code.
the latter uses an API that was long deprecated in libavcodec and is
now removed. Add new filter af_lavrresample which uses libavresample.
`af_lavrresample` has roughly the same features and structure as
`af_lavcresample` had.
Libavresample is now a mandatory dependency. Change configure script
accordingly and remove #ifdefs in other code.
13 files changed:
diff --git a/DOCS/man/en/af.rst b/DOCS/man/en/af.rst
--- a/DOCS/man/en/af.rst
+++ b/DOCS/man/en/af.rst
Available filters are:
-resample[=srate[:sloppy[:type]]]
- Changes the sample rate of the audio stream. Can be used if you have a
- fixed frequency sound card or if you are stuck with an old sound card that
- is only capable of max 44.1kHz. This filter is automatically enabled if
- necessary. It only supports 16-bit integer and float in native-endian
- format as input.
-
- <srate>
- output sample frequency in Hz. The valid range for this parameter is
- 8000 to 192000. If the input and output sample frequency are the same
- or if this parameter is omitted the filter is automatically unloaded.
- A high sample frequency normally improves the audio quality,
- especially when used in combination with other filters.
- <sloppy>
- Allow (1) or disallow (0) the output frequency to differ slightly from
- the frequency given by <srate> (default: 1). Can be used if the
- startup of the playback is extremely slow.
- <type>
- Select which resampling method to use.
-
- :0: linear interpolation (fast, poor quality especially when
- upsampling)
- :1: polyphase filterbank and integer processing
- :2: polyphase filterbank and floating point processing
- (slow, best quality)
-
- *EXAMPLE*:
-
- ``mplayer --af=resample=44100:0:0``
- would set the output frequency of the resample filter to 44100Hz using
- exact output frequency scaling and linear interpolation.
-
-lavcresample[=srate[:length[:linear[:count[:cutoff]]]]]
+lavrresample[=option1:option2:...]
Changes the sample rate of the audio stream to an integer <srate> in Hz.
- It only supports the 16-bit native-endian format.
- <srate>
- the output sample rate
- <length>
+ This filter is automatically inserted if the audio output device does not
+ support the sample rate of the file played. It only supports the
+ 16-bit integer native-endian format.
+
+ srate=<srate>
+ the output sample rate (defaut: 44100)
+ length=<length>
length of the filter with respect to the lower sampling rate (default:
16)
- <linear>
- if 1 then filters will be linearly interpolated between polyphase
- entries
- <count>
+ phase_shift=<count>
log2 of the number of polyphase entries (..., 10->1024, 11->2048,
12->4096, ...) (default: 10->1024)
- <cutoff>
+ cutoff=<cutoff>
cutoff frequency (0.0-1.0), default set depending upon filter length
+ linear
+ if set then filters will be linearly interpolated between polyphase
+ entries (default: no)
lavcac3enc[=tospdif[:bitrate[:minchn]]]
Encode multi-channel audio to AC-3 at runtime using libavcodec. Supports
--- a/DOCS/man/en/options.rst
+++ b/DOCS/man/en/options.rst
--srate=<Hz>
Select the output sample rate to be used (of course sound cards have
limits on this). If the sample frequency selected is different from that
- of the current media, the resample or lavcresample audio filter will be
- inserted into the audio filter layer to compensate for the difference. The
- type of resampling can be controlled by the ``--af-adv`` option.
+ of the current media, the lavrresample audio filter will be inserted into
+ the audio filter layer to compensate for the difference. The type of
+ resampling can be controlled by the ``--af-adv`` option.
--ss=<time>
Seek to given time position.
diff --git a/Makefile b/Makefile
--- a/Makefile
+++ b/Makefile
libaf/af_hrtf.c \
libaf/af_karaoke.c \
libaf/af_lavcac3enc.c \
- libaf/af_lavcresample.c \
+ libaf/af_lavrresample.c \
libaf/af_pan.c \
- libaf/af_resample.c \
libaf/af_scaletempo.c \
libaf/af_sinesuppress.c \
libaf/af_stats.c \
diff --git a/configure b/configure
--- a/configure
+++ b/configure
--disable-libass disable subtitle rendering with libass [autodetect]
--enable-rpath enable runtime linker path for extra libs [disabled]
--disable-libpostproc disable postprocess filter (vf_pp) [autodetect]
- --disable-libavresample disable libavresample (sample format conversion) [autodetect]
Codecs:
--enable-gif enable GIF support [autodetect]
_ass=auto
_rpath=no
libpostproc=auto
-libavresample=auto
_asmalign_pot=auto
_stream_cache=yes
_priority=no
--disable-rpath) _rpath=no ;;
--enable-libpostproc) libpostproc=yes ;;
--disable-libpostproc) libpostproc=no ;;
- --enable-libavresample) libavresample=yes ;;
- --disable-libavresample) libavresample=no ;;
--enable-enca) _enca=yes ;;
--disable-enca) _enca=no ;;
echores "$_lcms2"
-all_libav_libs="libavutil > 51.21.0:libavcodec >= 54.25.0:libavformat > 53.20.0:libswscale >= 2.0.0"
+all_libav_libs="libavutil > 51.21.0:libavcodec >= 54.25.0:libavformat > 53.20.0:libswscale >= 2.0.0:libavresample >= 1.0.0"
echocheck "Libav ($all_libav_libs)"
if test "$ffmpeg" = auto ; then
IFS=":" # shell should not be used for programming
fi
echores "$libpostproc"
-echocheck "libavresample >= 1.0.0"
-if test "$libavresample" = auto ; then
- libavresample=no
- if pkg_config_add "libavresample >= 1.0.0" ; then
- libavresample=yes
- fi
-fi
-if test "$libavresample" = yes ; then
- def_libavresample='#define CONFIG_LIBAVRESAMPLE 1'
-else
- def_libavresample='#undef CONFIG_LIBAVRESAMPLE'
-fi
-echores "$libavresample"
-
echocheck "libdv-0.9.5+"
if test "$_libdv" = auto ; then
_libdv=no
LCMS2 = $_lcms2
LIBNUT = $_libnut
LIBPOSTPROC = $libpostproc
-LIBAVRESAMPLE = $libavresample
LIBSMBCLIENT = $_smb
LIBQUVI = $_libquvi
LIBTHEORA = $_theora
$def_zlib
$def_libpostproc
-$def_libavresample
$def_libnut
diff --git a/etc/example.conf b/etc/example.conf
--- a/etc/example.conf
+++ b/etc/example.conf
# Specify the mixer device.
#mixer = /dev/mixer
-# Resample the sound to 44100Hz with the lavcresample audio filter.
-#af=lavcresample=44100
+# Resample the sound to 44100Hz with the lavrresample audio filter.
+#af=lavrresample=44100
# Output audio to S/PDIF
#ao=alsa:device=spdif
diff --git a/libaf/af.c b/libaf/af.c
--- a/libaf/af.c
+++ b/libaf/af.c
extern af_info_t af_info_delay;
extern af_info_t af_info_channels;
extern af_info_t af_info_format;
-extern af_info_t af_info_resample;
extern af_info_t af_info_volume;
extern af_info_t af_info_equalizer;
extern af_info_t af_info_gate;
extern af_info_t af_info_volnorm;
extern af_info_t af_info_extrastereo;
extern af_info_t af_info_lavcac3enc;
-extern af_info_t af_info_lavcresample;
+extern af_info_t af_info_lavrresample;
extern af_info_t af_info_sweep;
extern af_info_t af_info_hrtf;
extern af_info_t af_info_ladspa;
&af_info_delay,
&af_info_channels,
&af_info_format,
- &af_info_resample,
&af_info_volume,
&af_info_equalizer,
&af_info_gate,
&af_info_volnorm,
&af_info_extrastereo,
&af_info_lavcac3enc,
- &af_info_lavcresample,
+ &af_info_lavrresample,
&af_info_sweep,
&af_info_hrtf,
#ifdef CONFIG_LADSPA
af = af_control_any_rev(s, AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
&(s->output.rate));
if (!af) {
- char *resampler = "resample";
- if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW)
- resampler = "lavcresample";
+ char *resampler = "lavrresample";
if((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_SLOW){
if(!strcmp(s->first->info->name,"format"))
af = af_append(s,s->first,resampler);
if(!af || (AF_OK != af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET,
&(s->output.rate))))
return -1;
- // Use lin int if the user wants fast
- if ((AF_INIT_TYPE_MASK & s->cfg.force) == AF_INIT_FAST) {
- char args[32];
- sprintf(args, "%d", s->output.rate);
- if (strcmp(resampler, "lavcresample") == 0)
- strcat(args, ":1");
- else
- strcat(args, ":0:0");
- af->control(af, AF_CONTROL_COMMAND_LINE, args);
- }
}
if(AF_OK != af_reinit(s,af))
return -1;
diff --git a/libaf/af.h b/libaf/af.h
--- a/libaf/af.h
+++ b/libaf/af.h
#endif
// Audio data chunk
-typedef struct af_data_s
-{
+typedef struct af_data {
void* audio; // data buffer
int len; // buffer length
int rate; // sample rate
/* Audio filter information not specific for current instance, but for
a specific filter */
-typedef struct af_info_s
-{
+typedef struct af_info {
const char *info;
const char *name;
const char *author;
diff --git a/libaf/af_lavcac3enc.c b/libaf/af_lavcac3enc.c
--- a/libaf/af_lavcac3enc.c
+++ b/libaf/af_lavcac3enc.c
#include <libavutil/mem.h>
#include <libavutil/opt.h>
#include <libavutil/samplefmt.h>
+#include <libavresample/avresample.h>
#include "config.h"
#include "af.h"
#include "reorder_ch.h"
-#ifdef CONFIG_LIBAVRESAMPLE
-#include <libavresample/avresample.h>
-#endif
#define AC3_MAX_CHANNELS 6
#define AC3_MAX_CODED_FRAME_SIZE 3840
// Data for specific instances of this filter
typedef struct af_ac3enc_s {
-#ifdef CONFIG_LIBAVRESAMPLE
AVAudioResampleContext *avr;
uint8_t *resample_buf[AC3_MAX_CHANNELS];
int linesize;
-#endif
AVFrame *frame;
struct AVCodec *lavc_acodec;
struct AVCodecContext *lavc_actx;
avcodec_close(s->lavc_actx);
-#ifdef CONFIG_LIBAVRESAMPLE
if (s->avr) {
uint64_t ch_layout =
av_get_default_channel_layout(af->data->nch);
return AF_ERROR;
}
}
-#endif
// Put sample parameters
s->lavc_actx->channels = af->data->nch;
#else
av_freep(&s->frame);
#endif
-#ifdef CONFIG_LIBAVRESAMPLE
avresample_free(&s->avr);
av_freep(&s->resample_buf[0]);
-#endif
free(s->pending_data);
free(s);
}
s->frame->data[0] = src;
s->frame->linesize[0] = total_samples * bps;
-#ifdef CONFIG_LIBAVRESAMPLE
if (s->avr) {
ret = avresample_convert(s->avr, s->resample_buf, s->linesize,
AC3_FRAME_SIZE, &src, total_samples * bps,
memcpy(s->frame->data, s->resample_buf, sizeof(s->resample_buf));
s->frame->linesize[0] = s->linesize;
}
-#endif
av_init_packet(&pkt);
pkt.data = dst;
"support expected sample formats!\n");
return AF_ERROR;
}
- enum AVSampleFormat fmt_packed = fmts[i];
-#ifdef CONFIG_LIBAVRESAMPLE
- fmt_packed = av_get_packed_sample_fmt(fmt_packed);
-#endif
+ enum AVSampleFormat fmt_packed = av_get_packed_sample_fmt(fmts[i]);
if (fmt_packed == AV_SAMPLE_FMT_S16) {
s->in_sampleformat = AF_FORMAT_S16_NE;
s->lavc_actx->sample_fmt = fmts[i];
}
}
if (av_sample_fmt_is_planar(s->lavc_actx->sample_fmt)) {
-#ifdef CONFIG_LIBAVRESAMPLE
s->avr = avresample_alloc_context();
if (!s->avr)
abort();
-#else
- mp_msg(MSGT_AFILTER, MSGL_ERR, "Libavresample is required for this "
- "filter to work with this libavcodec version.\n");
- return AF_ERROR;
-#endif
}
char buf[100];
mp_msg(MSGT_AFILTER, MSGL_V, "[af_lavcac3enc]: in sample format: %s\n",
diff --git a/libaf/af_lavcresample.c b/libaf/af_lavcresample.c
--- a/libaf/af_lavcresample.c
+++ /dev/null
@@ -1,213 +0,0 @@
-/*
- * Copyright (c) 2004 Michael Niedermayer <[email protected]>
- *
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <inttypes.h>
-
-#include "config.h"
-#include "af.h"
-#include "libavcodec/avcodec.h"
-#include "libavutil/rational.h"
-
-// Data for specific instances of this filter
-typedef struct af_resample_s{
- struct AVResampleContext *avrctx;
- int16_t *in[AF_NCH];
- int in_alloc;
- int index;
-
- int filter_length;
- int linear;
- int phase_shift;
- double cutoff;
-
- int ctx_out_rate;
- int ctx_in_rate;
- int ctx_filter_size;
- int ctx_phase_shift;
- int ctx_linear;
- double ctx_cutoff;
-}af_resample_t;
-
-
-// Initialization and runtime control
-static int control(struct af_instance_s* af, int cmd, void* arg)
-{
- af_resample_t* s = (af_resample_t*)af->setup;
- af_data_t *data= (af_data_t*)arg;
- int out_rate, test_output_res; // helpers for checking input format
-
- switch(cmd){
- case AF_CONTROL_REINIT:
- if((af->data->rate == data->rate) || (af->data->rate == 0))
- return AF_DETACH;
-
- af->data->nch = data->nch;
- if (af->data->nch > AF_NCH) af->data->nch = AF_NCH;
- af->data->format = AF_FORMAT_S16_NE;
- af->data->bps = 2;
- af->mul = (double)af->data->rate / data->rate;
- af->delay = af->data->nch * s->filter_length / min(af->mul, 1); // *bps*.5
-
- if (s->ctx_out_rate != af->data->rate || s->ctx_in_rate != data->rate || s->ctx_filter_size != s->filter_length ||
- s->ctx_phase_shift != s->phase_shift || s->ctx_linear != s->linear || s->ctx_cutoff != s->cutoff) {
- if(s->avrctx) av_resample_close(s->avrctx);
- s->avrctx= av_resample_init(af->data->rate, /*in_rate*/data->rate, s->filter_length, s->phase_shift, s->linear, s->cutoff);
- s->ctx_out_rate = af->data->rate;
- s->ctx_in_rate = data->rate;
- s->ctx_filter_size = s->filter_length;
- s->ctx_phase_shift = s->phase_shift;
- s->ctx_linear = s->linear;
- s->ctx_cutoff = s->cutoff;
- }
-
- // hack to make af_test_output ignore the samplerate change
- out_rate = af->data->rate;
- af->data->rate = data->rate;
- test_output_res = af_test_output(af, (af_data_t*)arg);
- af->data->rate = out_rate;
- return test_output_res;
- case AF_CONTROL_COMMAND_LINE:{
- s->cutoff= 0.0;
- sscanf((char*)arg,"%d:%d:%d:%d:%lf", &af->data->rate, &s->filter_length, &s->linear, &s->phase_shift, &s->cutoff);
- if(s->cutoff <= 0.0) s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80);
- return AF_OK;
- }
- case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
- af->data->rate = *(int*)arg;
- return AF_OK;
- }
- return AF_UNKNOWN;
-}
-
-// Deallocate memory
-static void uninit(struct af_instance_s* af)
-{
- if(af->data)
- free(af->data->audio);
- free(af->data);
- if(af->setup){
- int i;
- af_resample_t *s = af->setup;
- if(s->avrctx) av_resample_close(s->avrctx);
- for (i=0; i < AF_NCH; i++)
- free(s->in[i]);
- free(s);
- }
-}
-
-// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
-{
- af_resample_t *s = af->setup;
- int i, j, consumed, ret = 0;
- int16_t *in = (int16_t*)data->audio;
- int16_t *out;
- int chans = data->nch;
- int in_len = data->len/(2*chans);
- int out_len = in_len * af->mul + 10;
- int16_t tmp[AF_NCH][out_len];
-
- if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
- return NULL;
-
- out= (int16_t*)af->data->audio;
-
- out_len= min(out_len, af->data->len/(2*chans));
-
- if(s->in_alloc < in_len + s->index){
- s->in_alloc= in_len + s->index;
- for(i=0; i<chans; i++){
- s->in[i]= realloc(s->in[i], s->in_alloc*sizeof(int16_t));
- }
- }
-
- if(chans==1){
- memcpy(&s->in[0][s->index], in, in_len * sizeof(int16_t));
- }else if(chans==2){
- for(j=0; j<in_len; j++){
- s->in[0][j + s->index]= *(in++);
- s->in[1][j + s->index]= *(in++);
- }
- }else{
- for(j=0; j<in_len; j++){
- for(i=0; i<chans; i++){
- s->in[i][j + s->index]= *(in++);
- }
- }
- }
- in_len += s->index;
-
- for(i=0; i<chans; i++){
- ret= av_resample(s->avrctx, tmp[i], s->in[i], &consumed, in_len, out_len, i+1 == chans);
- }
- out_len= ret;
-
- s->index= in_len - consumed;
- for(i=0; i<chans; i++){
- memmove(s->in[i], s->in[i] + consumed, s->index*sizeof(int16_t));
- }
-
- if(chans==1){
- memcpy(out, tmp[0], out_len*sizeof(int16_t));
- }else if(chans==2){
- for(j=0; j<out_len; j++){
- *(out++)= tmp[0][j];
- *(out++)= tmp[1][j];
- }
- }else{
- for(j=0; j<out_len; j++){
- for(i=0; i<chans; i++){
- *(out++)= tmp[i][j];
- }
- }
- }
-
- data->audio = af->data->audio;
- data->len = out_len*chans*2;
- data->rate = af->data->rate;
- return data;
-}
-
-static int af_open(af_instance_t* af){
- af_resample_t *s = calloc(1,sizeof(af_resample_t));
- af->control=control;
- af->uninit=uninit;
- af->play=play;
- af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
- s->filter_length= 16;
- s->cutoff= max(1.0 - 6.5/(s->filter_length+8), 0.80);
- s->phase_shift= 10;
-// s->setup = RSMP_INT | FREQ_SLOPPY;
- af->setup=s;
- return AF_OK;
-}
-
-af_info_t af_info_lavcresample = {
- "Sample frequency conversion using libavcodec",
- "lavcresample",
- "Michael Niedermayer",
- "",
- AF_FLAGS_REENTRANT,
- af_open
-};
diff --git a/libaf/af_lavrresample.c b/libaf/af_lavrresample.c
--- /dev/null
+++ b/libaf/af_lavrresample.c
@@ -0,0 +1,254 @@
+/*
+ * Copyright (c) 2004 Michael Niedermayer <[email protected]>
+ * Copyright (c) 2013 Stefano Pigozzi <[email protected]>
+ * Copyright (c) 2013 Uoti Urpala <[email protected]>
+ *
+ * This file is part of mplayer2.
+ *
+ * mplayer2 is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * mplayer2 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 General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License along
+ * with mplayer2. If not, see <http://www.gnu.org/licenses/>.
+ */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <inttypes.h>
+#include <stdbool.h>
+
+#include <libavutil/common.h>
+#include <libavutil/opt.h>
+#include <libavutil/samplefmt.h>
+#include <libavutil/mathematics.h>
+#include <libavresample/avresample.h>
+
+#include "talloc.h"
+#include "config.h"
+
+#include "mp_msg.h"
+#include "subopt-helper.h"
+#include "af.h"
+
+struct af_resample_opts {
+ int filter_size;
+ int phase_shift;
+ int linear;
+ double cutoff;
+
+ int out_rate;
+ int in_rate;
+};
+
+struct af_resample {
+ struct AVAudioResampleContext *avrctx;
+ struct af_resample_opts ctx; // opts in the context
+ struct af_resample_opts opts; // opts requested by the user
+};
+
+static double af_resample_default_cutoff(int filter_size)
+{
+ return FFMAX(1.0 - 6.5 / (filter_size + 8), 0.80);
+}
+
+static bool needs_lavrctx_reconfigure(struct af_resample *s,
+ struct af_data *in,
+ struct af_data *out)
+{
+ return s->ctx.out_rate != out->rate ||
+ s->ctx.in_rate != in->rate ||
+ s->ctx.filter_size != s->opts.filter_size ||
+ s->ctx.phase_shift != s->opts.phase_shift ||
+ s->ctx.linear != s->opts.linear ||
+ s->ctx.cutoff != s->opts.cutoff;
+
+}
+
+#define ctx_opt_set_int(a,b) av_opt_set_int(s->avrctx, (a), (b), 0)
+#define ctx_opt_set_dbl(a,b) av_opt_set_double(s->avrctx, (a), (b), 0)
+
+static int control(struct af_instance_s *af, int cmd, void *arg)
+{
+ struct af_resample *s = af->setup;
+ struct af_data *in = arg;
+ struct af_data *out = af->data;
+
+ switch (cmd) {
+ case AF_CONTROL_REINIT: {
+ if ((out->rate == in->rate) || (out->rate == 0))
+ return AF_DETACH;
+
+ out->nch = FFMIN(in->nch, AF_NCH);
+ out->format = AF_FORMAT_S16_NE;
+ out->bps = 2;
+ af->mul = (double) out->rate / in->rate;
+ af->delay = out->nch * s->opts.filter_size / FFMIN(af->mul, 1);
+
+ if (needs_lavrctx_reconfigure(s, in, out)) {
+ if (s->avrctx)
+ avresample_close(s->avrctx);
+
+ s->ctx.out_rate = out->rate;
+ s->ctx.in_rate = in->rate;
+ s->ctx.filter_size = s->opts.filter_size;
+ s->ctx.phase_shift = s->opts.phase_shift;
+ s->ctx.linear = s->opts.linear;
+ s->ctx.cutoff = s->opts.cutoff;
+
+ int ch_layout = av_get_default_channel_layout(out->nch);
+
+ ctx_opt_set_int("in_channel_layout", ch_layout);
+ ctx_opt_set_int("out_channel_layout", ch_layout);
+
+ ctx_opt_set_int("in_sample_rate", s->ctx.in_rate);
+ ctx_opt_set_int("out_sample_rate", s->ctx.out_rate);
+
+ ctx_opt_set_int("in_sample_fmt", AV_SAMPLE_FMT_S16);
+ ctx_opt_set_int("out_sample_fmt", AV_SAMPLE_FMT_S16);
+
+ ctx_opt_set_int("filter_size", s->ctx.filter_size);
+ ctx_opt_set_int("phase_shift", s->ctx.phase_shift);
+ ctx_opt_set_int("linear_interp", s->ctx.linear);
+
+ ctx_opt_set_dbl("cutoff", s->ctx.cutoff);
+
+ if (avresample_open(s->avrctx) < 0) {
+ mp_msg(MSGT_AFILTER, MSGL_ERR, "[lavrresample] Cannot open "
+ "Libavresample Context. \n");
+ return AF_ERROR;
+ }
+ }
+
+ int out_rate, test_output_res;
+ // hack to make af_test_output ignore the samplerate change
+ out_rate = out->rate;
+ out->rate = in->rate;
+ test_output_res = af_test_output(af, in);
+ out->rate = out_rate;
+ return test_output_res;
+ }
+ case AF_CONTROL_COMMAND_LINE: {
+ s->opts.cutoff = 0.0;
+
+ const opt_t subopts[] = {
+ {"srate", OPT_ARG_INT, &out->rate, NULL},
+ {"filter_size", OPT_ARG_INT, &s->opts.filter_size, NULL},
+ {"phase_shift", OPT_ARG_INT, &s->opts.phase_shift, NULL},
+ {"linear", OPT_ARG_BOOL, &s->opts.linear, NULL},
+ {"cutoff", OPT_ARG_FLOAT, &s->opts.cutoff, NULL},
+ };
+
+ if (subopt_parse(arg, subopts) != 0) {
+ mp_msg(MSGT_AFILTER, MSGL_ERR, "[lavrresample] Invalid option "
+ "specified.\n");
+ return AF_ERROR;
+ }
+
+ if (s->opts.cutoff <= 0.0)
+ s->opts.cutoff = af_resample_default_cutoff(s->opts.filter_size);
+ return AF_OK;
+ }
+ case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
+ out->rate = *(int *)arg;
+ return AF_OK;
+ }
+ return AF_UNKNOWN;
+}
+
+#undef ctx_opt_set_int
+#undef ctx_opt_set_dbl
+
+static void uninit(struct af_instance_s *af)
+{
+ if (af->setup) {
+ struct af_resample *s = af->setup;
+ if (s->avrctx)
+ avresample_close(s->avrctx);
+ talloc_free(af->setup);
+ }
+}
+
+static struct af_data *play(struct af_instance_s *af, struct af_data *data)
+{
+ struct af_resample *s = af->setup;
+ struct af_data *in = data;
+ struct af_data *out = af->data;
+
+
+ int in_size = data->len;
+ int in_samples = in_size / (data->bps * data->nch);
+ int out_samples = avresample_available(s->avrctx) +
+ av_rescale_rnd(avresample_get_delay(s->avrctx) + in_samples,
+ s->ctx.out_rate, s->ctx.in_rate, AV_ROUND_UP);
+ int out_size = out->bps * out_samples * out->nch;
+
+ if (talloc_get_size(out->audio) < out_size)
+ out->audio = talloc_realloc_size(out, out->audio, out_size);
+
+ af->delay = out->bps * av_rescale_rnd(avresample_get_delay(s->avrctx),
+ s->ctx.out_rate, s->ctx.in_rate,
+ AV_ROUND_UP);
+
+ out_samples = avresample_convert(s->avrctx,
+ (uint8_t **) &out->audio, out_size, out_samples,
+ (uint8_t **) &in->audio, in_size, in_samples);
+
+ out_size = out->bps * out_samples * out->nch;
+ in->audio = out->audio;
+ in->len = out_size;
+ in->rate = s->ctx.out_rate;
+ return data;
+}
+
+static int af_open(struct af_instance_s *af)
+{
+ struct af_resample *s = talloc_zero(NULL, struct af_resample);
+
+ af->control = control;
+ af->uninit = uninit;
+ af->play = play;
+ af->mul = 1;
+ af->data = talloc_zero(s, struct af_data);
+
+ af->data->rate = 44100;
+
+ int default_filter_size = 16;
+ s->opts = (struct af_resample_opts) {
+ .linear = 0,
+ .filter_size = default_filter_size,
+ .cutoff = af_resample_default_cutoff(default_filter_size),
+ .phase_shift = 10,
+ };
+
+ s->avrctx = avresample_alloc_context();
+ af->setup = s;
+
+ if (s->avrctx) {
+ return AF_OK;
+ } else {
+ mp_msg(MSGT_AFILTER, MSGL_ERR, "[lavrresample] Cannot initialize "
+ "libavresample context. \n");
+ uninit(af);
+ return AF_ERROR;
+ }
+}
+
+struct af_info af_info_lavrresample = {
+ "Sample frequency conversion using libavresample",
+ "lavrresample",
+ "Stefano Pigozzi (based on Michael Niedermayer's lavcresample)",
+ "",
+ AF_FLAGS_REENTRANT,
+ af_open
+};
+#warning don't use subopt / positional arguments
+#warning use lavr default option values
+#warning support float and other data types
diff --git a/libaf/af_resample.c b/libaf/af_resample.c
--- a/libaf/af_resample.c
+++ /dev/null
@@ -1,394 +0,0 @@
-/*
- * This audio filter changes the sample rate.
- *
- * Copyright (C) 2002 Anders Johansson [email protected]
- *
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <inttypes.h>
-
-#include "libavutil/common.h"
-#include "libavutil/mathematics.h"
-#include "af.h"
-#include "dsp.h"
-
-/* Below definition selects the length of each poly phase component.
- Valid definitions are L8 and L16, where the number denotes the
- length of the filter. This definition affects the computational
- complexity (see play()), the performance (see filter.h) and the
- memory usage. The filter length is chosen to 8 if the machine is
- slow and to 16 if the machine is fast and has MMX.
-*/
-
-#if !HAVE_MMX // This machine is slow
-#define L8
-#else
-#define L16
-#endif
-
-#include "af_resample_template.c"
-
-// Filtering types
-#define RSMP_LIN (0<<0) // Linear interpolation
-#define RSMP_INT (1<<0) // 16 bit integer
-#define RSMP_FLOAT (2<<0) // 32 bit floating point
-#define RSMP_MASK (3<<0)
-
-// Defines for sloppy or exact resampling
-#define FREQ_SLOPPY (0<<2)
-#define FREQ_EXACT (1<<2)
-#define FREQ_MASK (1<<2)
-
-// Accuracy for linear interpolation
-#define STEPACCURACY 32
-
-// local data
-typedef struct af_resample_s
-{
- void* w; // Current filter weights
- void** xq; // Circular buffers
- uint32_t xi; // Index for circular buffers
- uint32_t wi; // Index for w
- uint32_t i; // Number of new samples to put in x queue
- uint32_t dn; // Down sampling factor
- uint32_t up; // Up sampling factor
- uint64_t step; // Step size for linear interpolation
- uint64_t pt; // Pointer remainder for linear interpolation
- int setup; // Setup parameters cmdline or through postcreate
-} af_resample_t;
-
-// Fast linear interpolation resample with modest audio quality
-static int linint(af_data_t* c,af_data_t* l, af_resample_t* s)
-{
- uint32_t len = 0; // Number of input samples
- uint32_t nch = l->nch; // Words pre transfer
- uint64_t step = s->step;
- int16_t* in16 = ((int16_t*)c->audio);
- int16_t* out16 = ((int16_t*)l->audio);
- int32_t* in32 = ((int32_t*)c->audio);
- int32_t* out32 = ((int32_t*)l->audio);
- uint64_t end = ((((uint64_t)c->len)/2LL)<<STEPACCURACY);
- uint64_t pt = s->pt;
- uint16_t tmp;
-
- switch (nch){
- case 1:
- while(pt < end){
- out16[len++]=in16[pt>>STEPACCURACY];
- pt+=step;
- }
- s->pt=pt & ((1LL<<STEPACCURACY)-1);
- break;
- case 2:
- end/=2;
- while(pt < end){
- out32[len++]=in32[pt>>STEPACCURACY];
- pt+=step;
- }
- len=(len<<1);
- s->pt=pt & ((1LL<<STEPACCURACY)-1);
- break;
- default:
- end /=nch;
- while(pt < end){
- tmp=nch;
- do {
- tmp--;
- out16[len+tmp]=in16[tmp+(pt>>STEPACCURACY)*nch];
- } while (tmp);
- len+=nch;
- pt+=step;
- }
- s->pt=pt & ((1LL<<STEPACCURACY)-1);
- }
- return len;
-}
-
-/* Determine resampling type and format */
-static int set_types(struct af_instance_s* af, af_data_t* data)
-{
- af_resample_t* s = af->setup;
- int rv = AF_OK;
- float rd = 0;
-
- // Make sure this filter isn't redundant
- if((af->data->rate == data->rate) || (af->data->rate == 0))
- return AF_DETACH;
- /* If sloppy and small resampling difference (2%) */
- rd = abs((float)af->data->rate - (float)data->rate)/(float)data->rate;
- if((((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (rd < 0.02) &&
- (data->format != (AF_FORMAT_FLOAT_NE))) ||
- ((s->setup & RSMP_MASK) == RSMP_LIN)){
- s->setup = (s->setup & ~RSMP_MASK) | RSMP_LIN;
- af->data->format = AF_FORMAT_S16_NE;
- af->data->bps = 2;
- mp_msg(MSGT_AFILTER, MSGL_V, "[resample] Using linear interpolation. \n");
- }
- else{
- /* If the input format is float or if float is explicitly selected
- use float, otherwise use int */
- if((data->format == (AF_FORMAT_FLOAT_NE)) ||
- ((s->setup & RSMP_MASK) == RSMP_FLOAT)){
- s->setup = (s->setup & ~RSMP_MASK) | RSMP_FLOAT;
- af->data->format = AF_FORMAT_FLOAT_NE;
- af->data->bps = 4;
- }
- else{
- s->setup = (s->setup & ~RSMP_MASK) | RSMP_INT;
- af->data->format = AF_FORMAT_S16_NE;
- af->data->bps = 2;
- }
- mp_msg(MSGT_AFILTER, MSGL_V, "[resample] Using %s processing and %s frequecy"
- " conversion.\n",
- ((s->setup & RSMP_MASK) == RSMP_FLOAT)?"floating point":"integer",
- ((s->setup & FREQ_MASK) == FREQ_SLOPPY)?"inexact":"exact");
- }
-
- if(af->data->format != data->format || af->data->bps != data->bps)
- rv = AF_FALSE;
- data->format = af->data->format;
- data->bps = af->data->bps;
- af->data->nch = data->nch;
- return rv;
-}
-
-// Initialization and runtime control
-static int control(struct af_instance_s* af, int cmd, void* arg)
-{
- switch(cmd){
- case AF_CONTROL_REINIT:{
- af_resample_t* s = af->setup;
- af_data_t* n = arg; // New configuration
- int i,d = 0;
- int rv = AF_OK;
-
- // Free space for circular buffers
- if(s->xq){
- free(s->xq[0]);
- free(s->xq);
- s->xq = NULL;
- }
-
- if(AF_DETACH == (rv = set_types(af,n)))
- return AF_DETACH;
-
- // If linear interpolation
- if((s->setup & RSMP_MASK) == RSMP_LIN){
- s->pt=0LL;
- s->step=((uint64_t)n->rate<<STEPACCURACY)/(uint64_t)af->data->rate+1LL;
- mp_msg(MSGT_AFILTER, MSGL_DBG2, "[resample] Linear interpolation step: 0x%016"PRIX64".\n",
- s->step);
- af->mul = (double)af->data->rate / n->rate;
- return rv;
- }
-
- // Calculate up and down sampling factors
- d=av_gcd(af->data->rate,n->rate);
-
- // If sloppy resampling is enabled limit the upsampling factor
- if(((s->setup & FREQ_MASK) == FREQ_SLOPPY) && (af->data->rate/d > 5000)){
- int up=af->data->rate/2;
- int dn=n->rate/2;
- int m=2;
- while(af->data->rate/(d*m) > 5000){
- d=av_gcd(up,dn);
- up/=2; dn/=2; m*=2;
- }
- d*=m;
- }
-
- // Create space for circular buffers
- s->xq = malloc(n->nch*sizeof(void*));
- s->xq[0] = calloc(n->nch, 2*L*af->data->bps);
- for(i=1;i<n->nch;i++)
- s->xq[i] = (uint8_t *)s->xq[i-1] + 2*L*af->data->bps;
- s->xi = 0;
-
- // Check if the design needs to be redone
- if(s->up != af->data->rate/d || s->dn != n->rate/d){
- float* w;
- float* wt;
- float fc;
- int j;
- s->up = af->data->rate/d;
- s->dn = n->rate/d;
- s->wi = 0;
- s->i = 0;
-
- // Calculate cutoff frequency for filter
- fc = 1/(float)(max(s->up,s->dn));
- // Allocate space for polyphase filter bank and prototype filter
- w = malloc(sizeof(float) * s->up *L);
- free(s->w);
- s->w = malloc(L*s->up*af->data->bps);
-
- // Design prototype filter type using Kaiser window with beta = 10
- if(NULL == w || NULL == s->w ||
- -1 == af_filter_design_fir(s->up*L, w, &fc, LP|KAISER , 10.0)){
- mp_msg(MSGT_AFILTER, MSGL_ERR, "[resample] Unable to design prototype filter.\n");
- return AF_ERROR;
- }
- // Copy data from prototype to polyphase filter
- wt=w;
- for(j=0;j<L;j++){//Columns
- for(i=0;i<s->up;i++){//Rows
- if((s->setup & RSMP_MASK) == RSMP_INT){
- float t=(float)s->up*32767.0*(*wt);
- ((int16_t*)s->w)[i*L+j] = (int16_t)((t>=0.0)?(t+0.5):(t-0.5));
- }
- else
- ((float*)s->w)[i*L+j] = (float)s->up*(*wt);
- wt++;
- }
- }
- free(w);
- mp_msg(MSGT_AFILTER, MSGL_V, "[resample] New filter designed up: %i "
- "down: %i\n", s->up, s->dn);
- }
-
- // Set multiplier and delay
- af->delay = 0; // not set correctly, but shouldn't be too large anyway
- af->mul = (double)s->up / s->dn;
- return rv;
- }
- case AF_CONTROL_COMMAND_LINE:{
- af_resample_t* s = af->setup;
- int rate=0;
- int type=RSMP_INT;
- int sloppy=1;
- sscanf((char*)arg,"%i:%i:%i", &rate, &sloppy, &type);
- s->setup = (sloppy?FREQ_SLOPPY:FREQ_EXACT) |
- (clamp(type,RSMP_LIN,RSMP_FLOAT));
- return af->control(af,AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET, &rate);
- }
- case AF_CONTROL_POST_CREATE:
- if((((af_cfg_t*)arg)->force & AF_INIT_FORMAT_MASK) == AF_INIT_FLOAT)
- ((af_resample_t*)af->setup)->setup = RSMP_FLOAT;
- return AF_OK;
- case AF_CONTROL_RESAMPLE_RATE | AF_CONTROL_SET:
- // Reinit must be called after this function has been called
-
- // Sanity check
- if(((int*)arg)[0] < 8000 || ((int*)arg)[0] > 192000){
- mp_msg(MSGT_AFILTER, MSGL_ERR, "[resample] The output sample frequency "
- "must be between 8kHz and 192kHz. Current value is %i \n",
- ((int*)arg)[0]);
- return AF_ERROR;
- }
-
- af->data->rate=((int*)arg)[0];
- mp_msg(MSGT_AFILTER, MSGL_V, "[resample] Changing sample rate "
- "to %iHz\n",af->data->rate);
- return AF_OK;
- }
- return AF_UNKNOWN;
-}
-
-// Deallocate memory
-static void uninit(struct af_instance_s* af)
-{
- af_resample_t *s = af->setup;
- if (s) {
- if (s->xq) free(s->xq[0]);
- free(s->xq);
- free(s->w);
- free(s);
- }
- if(af->data)
- free(af->data->audio);
- free(af->data);
-}
-
-// Filter data through filter
-static af_data_t* play(struct af_instance_s* af, af_data_t* data)
-{
- int len = 0; // Length of output data
- af_data_t* c = data; // Current working data
- af_data_t* l = af->data; // Local data
- af_resample_t* s = af->setup;
-
- if(AF_OK != RESIZE_LOCAL_BUFFER(af,data))
- return NULL;
-
- // Run resampling
- switch(s->setup & RSMP_MASK){
- case(RSMP_INT):
-# define FORMAT_I 1
- if(s->up>s->dn){
-# define UP
-# include "af_resample_template.c"
-# undef UP
- }
- else{
-# define DN
-# include "af_resample_template.c"
-# undef DN
- }
- break;
- case(RSMP_FLOAT):
-# undef FORMAT_I
-# define FORMAT_F 1
- if(s->up>s->dn){
-# define UP
-# include "af_resample_template.c"
-# undef UP
- }
- else{
-# define DN
-# include "af_resample_template.c"
-# undef DN
- }
- break;
- case(RSMP_LIN):
- len = linint(c, l, s);
- break;
- }
-
- // Set output data
- c->audio = l->audio;
- c->len = len*l->bps;
- c->rate = l->rate;
-
- return c;
-}
-
-// Allocate memory and set function pointers
-static int af_open(af_instance_t* af){
- af->control=control;
- af->uninit=uninit;
- af->play=play;
- af->mul=1;
- af->data=calloc(1,sizeof(af_data_t));
- af->setup=calloc(1,sizeof(af_resample_t));
- if(af->data == NULL || af->setup == NULL)
- return AF_ERROR;
- ((af_resample_t*)af->setup)->setup = RSMP_INT | FREQ_SLOPPY;
- return AF_OK;
-}
-
-// Description of this plugin
-af_info_t af_info_resample = {
- "Sample frequency conversion",
- "resample",
- "Anders",
- "",
- AF_FLAGS_REENTRANT,
- af_open
-};
diff --git a/libaf/af_resample_template.c b/libaf/af_resample_template.c
+++ /dev/null
@@ -1,171 +0,0 @@
-/*
- * Copyright (C) 2002 Anders Johansson [email protected]
- *
- * This file is part of MPlayer.
- *
- * MPlayer is free software; you can redistribute it and/or modify
- * it under the terms of the GNU General Public License as published by
- * the Free Software Foundation; either version 2 of the License, or
- * (at your option) any later version.
- *
- * MPlayer 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 General Public License for more details.
- *
- * You should have received a copy of the GNU General Public License along
- * with MPlayer; if not, write to the Free Software Foundation, Inc.,
- * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
- */
-
-/* This file contains the resampling engine, the sample format is
- controlled by the FORMAT parameter, the filter length by the L
- parameter and the resampling type by UP and DN. This file should
- only be included by af_resample.c
-*/
-
-#undef L
-#undef SHIFT
-#undef FORMAT
-#undef FIR
-#undef ADDQUE
-
-/* The length Lxx definition selects the length of each poly phase
- component. Valid definitions are L8 and L16 where the number
- defines the nuber of taps. This definition affects the
- computational complexity, the performance and the memory usage.
-*/
-
-/* The FORMAT_x parameter selects the sample format type currently
- float and int16 are supported. Thes two formats are selected by
- defining eiter FORMAT_F or FORMAT_I. The advantage of using float
- is that the amplitude and therefore the SNR isn't affected by the
- filtering, the disadvantage is that it is a lot slower.
-*/
-
-#if defined(FORMAT_I)
-#define SHIFT >>16
-#define FORMAT int16_t
-#else
-#define SHIFT
-#define FORMAT float
-#endif
-
-// Short filter
-#if defined(L8)
-
-#define L 8 // Filter length
-// Unrolled loop to speed up execution
-#define FIR(x,w,y) \
- (y[0]) = ( w[0]*x[0]+w[1]*x[1]+w[2]*x[2]+w[3]*x[3] \
- + w[4]*x[4]+w[5]*x[5]+w[6]*x[6]+w[7]*x[7] ) SHIFT
-
-
-
-#else /* L8/L16 */
-
-#define L 16
-// Unrolled loop to speed up execution
-#define FIR(x,w,y) \
- y[0] = ( w[0] *x[0] +w[1] *x[1] +w[2] *x[2] +w[3] *x[3] \
- + w[4] *x[4] +w[5] *x[5] +w[6] *x[6] +w[7] *x[7] \
- + w[8] *x[8] +w[9] *x[9] +w[10]*x[10]+w[11]*x[11] \
- + w[12]*x[12]+w[13]*x[13]+w[14]*x[14]+w[15]*x[15] ) SHIFT
-
-#endif /* L8/L16 */
-
-// Macro to add data to circular que
-#define ADDQUE(xi,xq,in)\
- xq[xi]=xq[(xi)+L]=*(in);\
- xi=((xi)-1)&(L-1);
-
-#if defined(UP)
-
- uint32_t ci = l->nch; // Index for channels
- uint32_t nch = l->nch; // Number of channels
- uint32_t inc = s->up/s->dn;
- uint32_t level = s->up%s->dn;
- uint32_t up = s->up;
- uint32_t dn = s->dn;
- uint32_t ns = c->len/l->bps;
- register FORMAT* w = s->w;
-
- register uint32_t wi = 0;
- register uint32_t xi = 0;
-
- // Index current channel
- while(ci--){
- // Temporary pointers
- register FORMAT* x = s->xq[ci];
- register FORMAT* in = ((FORMAT*)c->audio)+ci;
- register FORMAT* out = ((FORMAT*)l->audio)+ci;
- FORMAT* end = in+ns; // Block loop end
- wi = s->wi; xi = s->xi;
-
- while(in < end){
- register uint32_t i = inc;
- if(wi<level) i++;
-
- ADDQUE(xi,x,in);
- in+=nch;
- while(i--){
- // Run the FIR filter
- FIR((&x[xi]),(&w[wi*L]),out);
- len++; out+=nch;
- // Update wi to point at the correct polyphase component
- wi=(wi+dn)%up;
- }
- }
-
- }
- // Save values that needs to be kept for next time
- s->wi = wi;
- s->xi = xi;
-#endif /* UP */
-
-#if defined(DN) /* DN */
- uint32_t ci = l->nch; // Index for channels
- uint32_t nch = l->nch; // Number of channels
- uint32_t inc = s->dn/s->up;
- uint32_t level = s->dn%s->up;
- uint32_t up = s->up;
- uint32_t dn = s->dn;
- uint32_t ns = c->len/l->bps;
- FORMAT* w = s->w;
-
- register int32_t i = 0;
- register uint32_t wi = 0;
- register uint32_t xi = 0;
-
- // Index current channel
- while(ci--){
- // Temporary pointers
- register FORMAT* x = s->xq[ci];
- register FORMAT* in = ((FORMAT*)c->audio)+ci;
- register FORMAT* out = ((FORMAT*)l->audio)+ci;
- register FORMAT* end = in+ns; // Block loop end
- i = s->i; wi = s->wi; xi = s->xi;
-
- while(in < end){
-
- ADDQUE(xi,x,in);
- in+=nch;
- if((--i)<=0){
- // Run the FIR filter
- FIR((&x[xi]),(&w[wi*L]),out);
- len++; out+=nch;
-
- // Update wi to point at the correct polyphase component
- wi=(wi+dn)%up;
-
- // Insert i number of new samples in queue
- i = inc;
- if(wi<level) i++;
- }
- }
- }
- // Save values that needs to be kept for next time
- s->wi = wi;
- s->xi = xi;
- s->i = i;
-#endif /* DN */
--- a/libmpcodecs/ad_ffmpeg.c
+++ b/libmpcodecs/ad_ffmpeg.c
#include <libavutil/audioconvert.h>
#include <libavutil/opt.h>
#include <libavutil/samplefmt.h>
+#include <libavresample/avresample.h>
#include "talloc.h"
#include "mpbswap.h"
-#ifdef CONFIG_LIBAVRESAMPLE
-#include <libavresample/avresample.h>
-#endif
-
static const ad_info_t info =
{
"libavcodec audio decoders",
int unitsize;
int previous_data_left; // input demuxer packet data
-#ifdef CONFIG_LIBAVRESAMPLE
AVAudioResampleContext *avr;
enum AVSampleFormat resample_fmt;
enum AVSampleFormat out_fmt;
int resample_channels;
uint8_t *resample_buf;
uint64_t resample_buf_size;
-#endif
};
static int preinit(sh_audio_t *sh)
int sample_format = sample_fmt_lavc2native(codec->sample_fmt);
if (sample_format == AF_FORMAT_UNKNOWN) {
-#ifndef CONFIG_LIBAVRESAMPLE
- if (av_sample_fmt_is_planar(codec->sample_fmt))
- mp_msg(MSGT_DECAUDIO, MSGL_ERR,
- "The player has been compiled without libavresample "
- "support,\nwhich is needed with this libavcodec decoder "
- "version.\nCompile with libavresample enabled to make "
- "audio decoding work!\n");
- else
- mp_msg(MSGT_DECAUDIO, MSGL_ERR, "Unsupported sample format\n");
- goto error;
-#else
if (priv->avr && (priv->resample_fmt != codec->sample_fmt ||
priv->resample_channels != codec->channels))
avresample_free(&priv->avr);
sample_format = sh_audio->sample_format;
} else if (priv->avr) {
avresample_free(&priv->avr);
-#endif
}
bool broken_srate = false;
}
return 0;
error:
-#ifdef CONFIG_LIBAVRESAMPLE
avresample_free(&priv->avr);
-#endif
return -1;
}
av_freep(&lavc_context->extradata);
av_freep(&lavc_context);
}
-#ifdef CONFIG_LIBAVRESAMPLE
avresample_free(&ctx->avr);
-#endif
#if LIBAVCODEC_VERSION_INT >= (54 << 16 | 28 << 8)
avcodec_free_frame(&ctx->avframe);
#else
if (format_result < 0)
return format_result;
-#ifdef CONFIG_LIBAVRESAMPLE
if (priv->avr) {
int ret;
uint64_t needed_size = av_samples_get_buffer_size(
priv->output = priv->resample_buf;
priv->output_left = priv->unitsize * ret;
- } else
-#endif
- {
+ } else {
uint64_t unitsize = av_get_bytes_per_sample(avctx->sample_fmt) *
(uint64_t)avctx->channels;
if (unitsize > 100000)