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Merge remote-tracking branch 'qatar/master' 

* qatar/master:
  lavfi: autoinsert resample filter when necessary.
  lavfi: add lavr-based audio resampling filter.
  x86: vc1: drop MMX loop filter implementation, which uses MMX2 instructions.

Conflicts:
	configure
	doc/filters.texi
	libavcodec/x86/vc1dsp_mmx.c
	libavfilter/Makefile
	libavfilter/allfilters.c
	libavfilter/avfiltergraph.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer 2012-05-13 00:13:49 +02:00
parent ad6f006081 012f04a277
commit 1caf614bec
7 changed files with 233 additions and 13 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

1
configure vendored
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@ -1690,6 +1690,7 @@ movie_filter_deps="avcodec avformat"
mp_filter_deps="gpl avcodec swscale postproc"
mptestsrc_filter_deps="gpl"
negate_filter_deps="lut_filter"
resample_filter_deps="avresample"
ocv_filter_deps="libopencv"
pan_filter_deps="swresample"
removelogo_filter_deps="avcodec avformat swscale"

4
doc/filters.texi
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@ -502,6 +502,10 @@ volume=-12dB
@end example
@end itemize
@section resample
Convert the audio sample format, sample rate and channel layout. This filter is
not meant to be used directly.
@c man end AUDIO FILTERS
@chapter Audio Sources

2
libavcodec/x86/vc1dsp_mmx.c
View File

@ -701,7 +701,6 @@ static void vc1_h_loop_filter16_ ## EXT(uint8_t *src, int stride, int pq) \
}
#if HAVE_YASM
LOOP_FILTER(mmx)
LOOP_FILTER(mmx2)
LOOP_FILTER(sse2)
LOOP_FILTER(ssse3)
@ -803,7 +802,6 @@ void ff_vc1dsp_init_mmx(VC1DSPContext *dsp)
#if HAVE_YASM
if (mm_flags & AV_CPU_FLAG_MMX) {
ASSIGN_LF(mmx);
}
return;
if (mm_flags & AV_CPU_FLAG_MMX2) {

11
libavcodec/x86/vc1dsp_yasm.asm
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@ -227,13 +227,6 @@ section .text
imul r2, 0x01010101
%endmacro
; I do not know why the sign extension is needed...
%macro PSIGNW_SRA_MMX 2
psraw %2, 15
PSIGNW_MMX %1, %2
%endmacro
%macro VC1_LF_MMX 1
INIT_MMX
cglobal vc1_v_loop_filter_internal_%1
@ -274,10 +267,6 @@ cglobal vc1_h_loop_filter8_%1, 3,5,0
RET
%endmacro
%define PABSW PABSW_MMX
%define PSIGNW PSIGNW_SRA_MMX
VC1_LF_MMX mmx
%define PABSW PABSW_MMX2
VC1_LF_MMX mmx2

2
libavfilter/Makefile
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@ -2,6 +2,7 @@ include $(SUBDIR)../config.mak
NAME = avfilter
FFLIBS = avutil swscale
FFLIBS-$(CONFIG_RESAMPLE_FILTER) += avresample
FFLIBS-$(CONFIG_ACONVERT_FILTER) += swresample
FFLIBS-$(CONFIG_AMOVIE_FILTER) += avformat avcodec
@ -48,6 +49,7 @@ OBJS-$(CONFIG_ASPLIT_FILTER) += af_asplit.o
OBJS-$(CONFIG_ASTREAMSYNC_FILTER) += af_astreamsync.o
OBJS-$(CONFIG_EARWAX_FILTER) += af_earwax.o
OBJS-$(CONFIG_PAN_FILTER) += af_pan.o
OBJS-$(CONFIG_RESAMPLE_FILTER) += af_resample.o
OBJS-$(CONFIG_SILENCEDETECT_FILTER) += af_silencedetect.o
OBJS-$(CONFIG_VOLUME_FILTER) += af_volume.o

225
libavfilter/af_resample.c Normal file
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@ -0,0 +1,225 @@
/*
*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* sample format and channel layout conversion audio filter
*/
#include "libavutil/avassert.h"
#include "libavutil/avstring.h"
#include "libavutil/mathematics.h"
#include "libavutil/opt.h"
#include "libavresample/avresample.h"
#include "audio.h"
#include "avfilter.h"
#include "internal.h"
typedef struct ResampleContext {
AVAudioResampleContext *avr;
int64_t next_pts;
} ResampleContext;
static av_cold void uninit(AVFilterContext *ctx)
{
ResampleContext *s = ctx->priv;
if (s->avr) {
avresample_close(s->avr);
avresample_free(&s->avr);
}
}
static int query_formats(AVFilterContext *ctx)
{
AVFilterLink *inlink = ctx->inputs[0];
AVFilterLink *outlink = ctx->outputs[0];
AVFilterFormats *in_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
AVFilterFormats *out_formats = avfilter_all_formats(AVMEDIA_TYPE_AUDIO);
avfilter_formats_ref(in_formats, &inlink->out_formats);
avfilter_formats_ref(out_formats, &outlink->in_formats);
return 0;
}
static int config_output(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
AVFilterLink *inlink = ctx->inputs[0];
ResampleContext *s = ctx->priv;
char buf1[64], buf2[64];
int ret;
if (s->avr) {
avresample_close(s->avr);
avresample_free(&s->avr);
}
if (inlink->channel_layout == outlink->channel_layout &&
inlink->sample_rate == outlink->sample_rate &&
inlink->format == outlink->format)
return 0;
if (!(s->avr = avresample_alloc_context()))
return AVERROR(ENOMEM);
av_opt_set_int(s->avr, "in_channel_layout", inlink ->channel_layout, 0);
av_opt_set_int(s->avr, "out_channel_layout", outlink->channel_layout, 0);
av_opt_set_int(s->avr, "in_sample_fmt", inlink ->format, 0);
av_opt_set_int(s->avr, "out_sample_fmt", outlink->format, 0);
av_opt_set_int(s->avr, "in_sample_rate", inlink ->sample_rate, 0);
av_opt_set_int(s->avr, "out_sample_rate", outlink->sample_rate, 0);
/* if both the input and output formats are s16 or u8, use s16 as
the internal sample format */
if (av_get_bytes_per_sample(inlink->format) <= 2 &&
av_get_bytes_per_sample(outlink->format) <= 2)
av_opt_set_int(s->avr, "internal_sample_fmt", AV_SAMPLE_FMT_S16P, 0);
if ((ret = avresample_open(s->avr)) < 0)
return ret;
outlink->time_base = (AVRational){ 1, outlink->sample_rate };
s->next_pts = AV_NOPTS_VALUE;
av_get_channel_layout_string(buf1, sizeof(buf1),
-1, inlink ->channel_layout);
av_get_channel_layout_string(buf2, sizeof(buf2),
-1, outlink->channel_layout);
av_log(ctx, AV_LOG_VERBOSE,
"fmt:%s srate: %d cl:%s -> fmt:%s srate: %d cl:%s\n",
av_get_sample_fmt_name(inlink ->format), inlink ->sample_rate, buf1,
av_get_sample_fmt_name(outlink->format), outlink->sample_rate, buf2);
return 0;
}
static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ResampleContext *s = ctx->priv;
int ret = avfilter_request_frame(ctx->inputs[0]);
/* flush the lavr delay buffer */
if (ret == AVERROR_EOF && s->avr) {
AVFilterBufferRef *buf;
int nb_samples = av_rescale_rnd(avresample_get_delay(s->avr),
outlink->sample_rate,
ctx->inputs[0]->sample_rate,
AV_ROUND_UP);
if (!nb_samples)
return ret;
buf = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
if (!buf)
return AVERROR(ENOMEM);
ret = avresample_convert(s->avr, (void**)buf->extended_data,
buf->linesize[0], nb_samples,
NULL, 0, 0);
if (ret <= 0) {
avfilter_unref_buffer(buf);
return (ret == 0) ? AVERROR_EOF : ret;
}
buf->pts = s->next_pts;
ff_filter_samples(outlink, buf);
return 0;
}
return ret;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ResampleContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
if (s->avr) {
AVFilterBufferRef *buf_out;
int delay, nb_samples, ret;
/* maximum possible samples lavr can output */
delay = avresample_get_delay(s->avr);
nb_samples = av_rescale_rnd(buf->audio->nb_samples + delay,
outlink->sample_rate, inlink->sample_rate,
AV_ROUND_UP);
buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
ret = avresample_convert(s->avr, (void**)buf_out->extended_data,
buf_out->linesize[0], nb_samples,
(void**)buf->extended_data, buf->linesize[0],
buf->audio->nb_samples);
av_assert0(!avresample_available(s->avr));
if (s->next_pts == AV_NOPTS_VALUE) {
if (buf->pts == AV_NOPTS_VALUE) {
av_log(ctx, AV_LOG_WARNING, "First timestamp is missing, "
"assuming 0.\n");
s->next_pts = 0;
} else
s->next_pts = av_rescale_q(buf->pts, inlink->time_base,
outlink->time_base);
}
if (ret > 0) {
buf_out->audio->nb_samples = ret;
if (buf->pts != AV_NOPTS_VALUE) {
buf_out->pts = av_rescale_q(buf->pts, inlink->time_base,
outlink->time_base) -
av_rescale(delay, outlink->sample_rate,
inlink->sample_rate);
} else
buf_out->pts = s->next_pts;
s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
ff_filter_samples(outlink, buf_out);
}
avfilter_unref_buffer(buf);
} else
ff_filter_samples(outlink, buf);
}
AVFilter avfilter_af_resample = {
.name = "resample",
.description = NULL_IF_CONFIG_SMALL("Audio resampling and conversion."),
.priv_size = sizeof(ResampleContext),
.uninit = uninit,
.query_formats = query_formats,
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ },
{ .name = NULL}},
.outputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.config_props = config_output,
.request_frame = request_frame },
{ .name = NULL}},
};

1
libavfilter/allfilters.c
View File

@ -46,6 +46,7 @@ void avfilter_register_all(void)
REGISTER_FILTER (PAN, pan, af);
REGISTER_FILTER (SILENCEDETECT, silencedetect, af);
REGISTER_FILTER (VOLUME, volume, af);
REGISTER_FILTER (RESAMPLE, resample, af);
REGISTER_FILTER (ABUFFER, abuffer, asrc);
REGISTER_FILTER (AEVALSRC, aevalsrc, asrc);