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Merge remote-tracking branch 'qatar/master'

Browse Source 
* qatar/master:
  mss3: use standard zigzag table
  mss3: split DSP functions that are used in MTS2(MSS4) into separate file
  motion-test: do not use getopt()
  tcp: add initial timeout limit for incoming connections
  configure: Change the rdtsc check to a linker check
  avconv: propagate fatal errors from lavfi.
  lavfi: add error handling to filter_samples().
  fate-run: make avconv() properly deal with multiple inputs.
  asplit: don't leak the input buffer.
  af_resample: fix request_frame() behavior.
  af_asyncts: fix request_frame() behavior.
  libx264: support aspect ratio switching
  matroskadec: honor error_recognition when encountering unknown elements.
  lavr: resampling: add support for s32p, fltp, and dblp internal sample formats
  lavr: resampling: add filter type and Kaiser window beta to AVOptions
  lavr: Use AV_SAMPLE_FMT_NONE to auto-select the internal sample format
  lavr: mix: validate internal sample format in ff_audio_mix_init()

Conflicts:
	ffmpeg.c
	ffplay.c
	libavcodec/libx264.c
	libavfilter/audio.c
	libavfilter/split.c
	libavformat/tcp.c
	tests/fate-run.sh

Merged-by: Michael Niedermayer <michaelni@gmx.at>
This commit is contained in:
Michael Niedermayer
2012-07-09 22:10:38 +02:00
parent bf5386385d 5467742232
commit f8911b987d
45 changed files with 648 additions and 339 deletions
Download Patch File Download Diff File Expand all files Collapse all files

2
libavcodec/Makefile
View File

@@ -327,7 +327,7 @@ OBJS-$(CONFIG_MSMPEG4V3_ENCODER) += msmpeg4.o msmpeg4enc.o msmpeg4data.o \
h263dec.o h263.o ituh263dec.o \
mpeg4videodec.o
OBJS-$(CONFIG_MSRLE_DECODER) += msrle.o msrledec.o
OBJS-$(CONFIG_MSA1_DECODER) += mss3.o
OBJS-$(CONFIG_MSA1_DECODER) += mss3.o mss34dsp.o
OBJS-$(CONFIG_MSS1_DECODER) += mss1.o
OBJS-$(CONFIG_MSVIDEO1_DECODER) += msvideo1.o
OBJS-$(CONFIG_MSVIDEO1_ENCODER) += msvideo1enc.o elbg.o

4
libavcodec/libx264.c
View File

@@ -175,8 +175,8 @@ static int X264_frame(AVCodecContext *ctx, AVPacket *pkt, const AVFrame *frame,
x4->params.b_tff = frame->top_field_first;
x264_encoder_reconfig(x4->enc, &x4->params);
}
if (x4->params.vui.i_sar_height != ctx->sample_aspect_ratio.den
|| x4->params.vui.i_sar_width != ctx->sample_aspect_ratio.num) {
if (x4->params.vui.i_sar_height != ctx->sample_aspect_ratio.den ||
x4->params.vui.i_sar_width != ctx->sample_aspect_ratio.num) {
x4->params.vui.i_sar_height = ctx->sample_aspect_ratio.den;
x4->params.vui.i_sar_width = ctx->sample_aspect_ratio.num;
x264_encoder_reconfig(x4->enc, &x4->params);

8
libavcodec/motion-test.c
View File

@@ -119,16 +119,10 @@ int main(int argc, char **argv)
int flags[2] = { AV_CPU_FLAG_MMX, AV_CPU_FLAG_MMX2 };
int flags_size = HAVE_MMX2 ? 2 : 1;
for(;;) {
c = getopt(argc, argv, "h");
if (c == -1)
break;
switch(c) {
case 'h':
if (argc > 1) {
help();
return 1;
}
}
printf("ffmpeg motion test\n");

108
libavcodec/mss3.c
View File

@@ -26,6 +26,8 @@
#include "avcodec.h"
#include "bytestream.h"
#include "dsputil.h"
#include "mss34dsp.h"
#define HEADER_SIZE 27
@@ -119,39 +121,6 @@ typedef struct MSS3Context {
int hblock[16 * 16];
} MSS3Context;
static const uint8_t mss3_luma_quant[64] = {
16, 11, 10, 16, 24, 40, 51, 61,
12, 12, 14, 19, 26, 58, 60, 55,
14, 13, 16, 24, 40, 57, 69, 56,
14, 17, 22, 29, 51, 87, 80, 62,
18, 22, 37, 56, 68, 109, 103, 77,
24, 35, 55, 64, 81, 104, 113, 92,
49, 64, 78, 87, 103, 121, 120, 101,
72, 92, 95, 98, 112, 100, 103, 99
};
static const uint8_t mss3_chroma_quant[64] = {
17, 18, 24, 47, 99, 99, 99, 99,
18, 21, 26, 66, 99, 99, 99, 99,
24, 26, 56, 99, 99, 99, 99, 99,
47, 66, 99, 99, 99, 99, 99, 99,
99, 99, 99, 99, 99, 99, 99, 99,
99, 99, 99, 99, 99, 99, 99, 99,
99, 99, 99, 99, 99, 99, 99, 99,
99, 99, 99, 99, 99, 99, 99, 99
};
static const uint8_t zigzag_scan[64] = {
0, 1, 8, 16, 9, 2, 3, 10,
17, 24, 32, 25, 18, 11, 4, 5,
12, 19, 26, 33, 40, 48, 41, 34,
27, 20, 13, 6, 7, 14, 21, 28,
35, 42, 49, 56, 57, 50, 43, 36,
29, 22, 15, 23, 30, 37, 44, 51,
58, 59, 52, 45, 38, 31, 39, 46,
53, 60, 61, 54, 47, 55, 62, 63
};
static void model2_reset(Model2 *m)
{
@@ -578,7 +547,7 @@ static int decode_dct(RangeCoder *c, DCTBlockCoder *bc, int *block,
if (!sign)
val = -val;
zz_pos = zigzag_scan[pos];
zz_pos = ff_zigzag_direct[pos];
block[zz_pos] = val * bc->qmat[zz_pos];
pos++;
}
@@ -586,58 +555,6 @@ static int decode_dct(RangeCoder *c, DCTBlockCoder *bc, int *block,
return pos == 64 ? 0 : -1;
}
#define DCT_TEMPLATE(blk, step, SOP, shift) \
const int t0 = -39409 * blk[7 * step] - 58980 * blk[1 * step]; \
const int t1 = 39410 * blk[1 * step] - 58980 * blk[7 * step]; \
const int t2 = -33410 * blk[5 * step] - 167963 * blk[3 * step]; \
const int t3 = 33410 * blk[3 * step] - 167963 * blk[5 * step]; \
const int t4 = blk[3 * step] + blk[7 * step]; \
const int t5 = blk[1 * step] + blk[5 * step]; \
const int t6 = 77062 * t4 + 51491 * t5; \
const int t7 = 77062 * t5 - 51491 * t4; \
const int t8 = 35470 * blk[2 * step] - 85623 * blk[6 * step]; \
const int t9 = 35470 * blk[6 * step] + 85623 * blk[2 * step]; \
const int tA = SOP(blk[0 * step] - blk[4 * step]); \
const int tB = SOP(blk[0 * step] + blk[4 * step]); \
\
blk[0 * step] = ( t1 + t6 + t9 + tB) >> shift; \
blk[1 * step] = ( t3 + t7 + t8 + tA) >> shift; \
blk[2 * step] = ( t2 + t6 - t8 + tA) >> shift; \
blk[3 * step] = ( t0 + t7 - t9 + tB) >> shift; \
blk[4 * step] = (-(t0 + t7) - t9 + tB) >> shift; \
blk[5 * step] = (-(t2 + t6) - t8 + tA) >> shift; \
blk[6 * step] = (-(t3 + t7) + t8 + tA) >> shift; \
blk[7 * step] = (-(t1 + t6) + t9 + tB) >> shift; \
#define SOP_ROW(a) ((a) << 16) + 0x2000
#define SOP_COL(a) ((a + 32) << 16)
static void dct_put(uint8_t *dst, int stride, int *block)
{
int i, j;
int *ptr;
ptr = block;
for (i = 0; i < 8; i++) {
DCT_TEMPLATE(ptr, 1, SOP_ROW, 13);
ptr += 8;
}
ptr = block;
for (i = 0; i < 8; i++) {
DCT_TEMPLATE(ptr, 8, SOP_COL, 22);
ptr++;
}
ptr = block;
for (j = 0; j < 8; j++) {
for (i = 0; i < 8; i++)
dst[i] = av_clip_uint8(ptr[i] + 128);
dst += stride;
ptr += 8;
}
}
static void decode_dct_block(RangeCoder *c, DCTBlockCoder *bc,
uint8_t *dst, int stride, int block_size,
int *block, int mb_x, int mb_y)
@@ -655,7 +572,7 @@ static void decode_dct_block(RangeCoder *c, DCTBlockCoder *bc,
c->got_error = 1;
return;
}
dct_put(dst + i * 8, stride, block);
ff_mss34_dct_put(dst + i * 8, stride, block);
}
dst += 8 * stride;
}
@@ -702,14 +619,6 @@ static void decode_haar_block(RangeCoder *c, HaarBlockCoder *hc,
}
}
static void gen_quant_mat(uint16_t *qmat, const uint8_t *ref, float scale)
{
int i;
for (i = 0; i < 64; i++)
qmat[i] = (uint16_t)(ref[i] * scale + 50.0) / 100;
}
static void reset_coders(MSS3Context *ctx, int quality)
{
int i, j;
@@ -726,15 +635,8 @@ static void reset_coders(MSS3Context *ctx, int quality)
for (j = 0; j < 125; j++)
model_reset(&ctx->image_coder[i].vq_model[j]);
if (ctx->dct_coder[i].quality != quality) {
float scale;
ctx->dct_coder[i].quality = quality;
if (quality > 50)
scale = 200.0f - 2 * quality;
else
scale = 5000.0f / quality;
gen_quant_mat(ctx->dct_coder[i].qmat,
i ? mss3_chroma_quant : mss3_luma_quant,
scale);
ff_mss34_gen_quant_mat(ctx->dct_coder[i].qmat, quality, !i);
}
memset(ctx->dct_coder[i].prev_dc, 0,
sizeof(*ctx->dct_coder[i].prev_dc) *

114
libavcodec/mss34dsp.c Normal file
View File

@@ -0,0 +1,114 @@
/*
* Common stuff for some Microsoft Screen codecs
* Copyright (C) 2012 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
#include "libavutil/common.h"
#include "mss34dsp.h"
static const uint8_t luma_quant[64] = {
16, 11, 10, 16, 24, 40, 51, 61,
12, 12, 14, 19, 26, 58, 60, 55,
14, 13, 16, 24, 40, 57, 69, 56,
14, 17, 22, 29, 51, 87, 80, 62,
18, 22, 37, 56, 68, 109, 103, 77,
24, 35, 55, 64, 81, 104, 113, 92,
49, 64, 78, 87, 103, 121, 120, 101,
72, 92, 95, 98, 112, 100, 103, 99
};
static const uint8_t chroma_quant[64] = {
17, 18, 24, 47, 99, 99, 99, 99,
18, 21, 26, 66, 99, 99, 99, 99,
24, 26, 56, 99, 99, 99, 99, 99,
47, 66, 99, 99, 99, 99, 99, 99,
99, 99, 99, 99, 99, 99, 99, 99,
99, 99, 99, 99, 99, 99, 99, 99,
99, 99, 99, 99, 99, 99, 99, 99,
99, 99, 99, 99, 99, 99, 99, 99
};
void ff_mss34_gen_quant_mat(uint16_t *qmat, int quality, int luma)
{
int i;
const uint8_t *qsrc = luma ? luma_quant : chroma_quant;
if (quality >= 50) {
int scale = 200 - 2 * quality;
for (i = 0; i < 64; i++)
qmat[i] = (qsrc[i] * scale + 50) / 100;
} else {
for (i = 0; i < 64; i++)
qmat[i] = (5000 * qsrc[i] / quality + 50) / 100;
}
}
#define DCT_TEMPLATE(blk, step, SOP, shift) \
const int t0 = -39409 * blk[7 * step] - 58980 * blk[1 * step]; \
const int t1 = 39410 * blk[1 * step] - 58980 * blk[7 * step]; \
const int t2 = -33410 * blk[5 * step] - 167963 * blk[3 * step]; \
const int t3 = 33410 * blk[3 * step] - 167963 * blk[5 * step]; \
const int t4 = blk[3 * step] + blk[7 * step]; \
const int t5 = blk[1 * step] + blk[5 * step]; \
const int t6 = 77062 * t4 + 51491 * t5; \
const int t7 = 77062 * t5 - 51491 * t4; \
const int t8 = 35470 * blk[2 * step] - 85623 * blk[6 * step]; \
const int t9 = 35470 * blk[6 * step] + 85623 * blk[2 * step]; \
const int tA = SOP(blk[0 * step] - blk[4 * step]); \
const int tB = SOP(blk[0 * step] + blk[4 * step]); \
\
blk[0 * step] = ( t1 + t6 + t9 + tB) >> shift; \
blk[1 * step] = ( t3 + t7 + t8 + tA) >> shift; \
blk[2 * step] = ( t2 + t6 - t8 + tA) >> shift; \
blk[3 * step] = ( t0 + t7 - t9 + tB) >> shift; \
blk[4 * step] = (-(t0 + t7) - t9 + tB) >> shift; \
blk[5 * step] = (-(t2 + t6) - t8 + tA) >> shift; \
blk[6 * step] = (-(t3 + t7) + t8 + tA) >> shift; \
blk[7 * step] = (-(t1 + t6) + t9 + tB) >> shift; \
#define SOP_ROW(a) ((a) << 16) + 0x2000
#define SOP_COL(a) ((a + 32) << 16)
void ff_mss34_dct_put(uint8_t *dst, int stride, int *block)
{
int i, j;
int *ptr;
ptr = block;
for (i = 0; i < 8; i++) {
DCT_TEMPLATE(ptr, 1, SOP_ROW, 13);
ptr += 8;
}
ptr = block;
for (i = 0; i < 8; i++) {
DCT_TEMPLATE(ptr, 8, SOP_COL, 22);
ptr++;
}
ptr = block;
for (j = 0; j < 8; j++) {
for (i = 0; i < 8; i++)
dst[i] = av_clip_uint8(ptr[i] + 128);
dst += stride;
ptr += 8;
}
}

45
libavcodec/mss34dsp.h Normal file
View File

@@ -0,0 +1,45 @@
/*
* Common stuff for some Microsoft Screen codecs
* Copyright (C) 2012 Konstantin Shishkov
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_MSS34DSP_H
#define AVCODEC_MSS34DSP_H
#include <stdint.h>
/**
* Generate quantisation matrix for given quality.
*
* @param qmat destination matrix
* @param quality quality setting (1-100)
* @param luma generate quantisation matrix for luma or chroma
*/
void ff_mss34_gen_quant_mat(uint16_t *qmat, int quality, int luma);
/**
* Transform and output DCT block.
*
* @param dst output plane
* @param stride output plane stride
* @param block block to transform and output
*/
void ff_mss34_dct_put(uint8_t *dst, int stride, int *block);
#endif /* AVCODEC_MSS34DSP_H */

6
libavfilter/af_aconvert.c
View File

@@ -135,12 +135,13 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AConvertContext *aconvert = inlink->dst->priv;
const int n = insamplesref->audio->nb_samples;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n);
int ret;
swr_convert(aconvert->swr, outsamplesref->data, n,
(void *)insamplesref->data, n);
@@ -148,8 +149,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
outsamplesref->audio->channel_layout = outlink->channel_layout;
ff_filter_samples(outlink, outsamplesref);
ret = ff_filter_samples(outlink, outsamplesref);
avfilter_unref_buffer(insamplesref);
return ret;
}
AVFilter avfilter_af_aconvert = {

6
libavfilter/af_amerge.c
View File

@@ -212,7 +212,7 @@ static inline void copy_samples(int nb_inputs, struct amerge_input in[],
}
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AMergeContext *am = ctx->priv;
@@ -232,7 +232,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
for (i = 1; i < am->nb_inputs; i++)
nb_samples = FFMIN(nb_samples, am->in[i].nb_samples);
if (!nb_samples)
return;
return 0;
outbuf = ff_get_audio_buffer(ctx->outputs[0], AV_PERM_WRITE, nb_samples);
outs = outbuf->data[0];
@@ -285,7 +285,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
}
}
ff_filter_samples(ctx->outputs[0], outbuf);
return ff_filter_samples(ctx->outputs[0], outbuf);
}
static av_cold int init(AVFilterContext *ctx, const char *args)

20
libavfilter/af_amix.c
View File

@@ -305,9 +305,7 @@ static int output_frame(AVFilterLink *outlink, int nb_samples)
if (s->next_pts != AV_NOPTS_VALUE)
s->next_pts += nb_samples;
ff_filter_samples(outlink, out_buf);
return 0;
return ff_filter_samples(outlink, out_buf);
}
/**
@@ -448,31 +446,37 @@ static int request_frame(AVFilterLink *outlink)
return output_frame(outlink, available_samples);
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
MixContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int i;
int i, ret = 0;
for (i = 0; i < ctx->nb_inputs; i++)
if (ctx->inputs[i] == inlink)
break;
if (i >= ctx->nb_inputs) {
av_log(ctx, AV_LOG_ERROR, "unknown input link\n");
return;
ret = AVERROR(EINVAL);
goto fail;
}
if (i == 0) {
int64_t pts = av_rescale_q(buf->pts, inlink->time_base,
outlink->time_base);
frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
ret = frame_list_add_frame(s->frame_list, buf->audio->nb_samples, pts);
if (ret < 0)
goto fail;
}
av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
ret = av_audio_fifo_write(s->fifos[i], (void **)buf->extended_data,
buf->audio->nb_samples);
fail:
avfilter_unref_buffer(buf);
return ret;
}
static int init(AVFilterContext *ctx, const char *args)

8
libavfilter/af_aresample.c
View File

@@ -168,13 +168,14 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref)
{
AResampleContext *aresample = inlink->dst->priv;
const int n_in = insamplesref->audio->nb_samples;
int n_out = n_in * aresample->ratio * 2 ;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamplesref = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n_out);
int ret;
avfilter_copy_buffer_ref_props(outsamplesref, insamplesref);
@@ -193,15 +194,16 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamplesref
if (n_out <= 0) {
avfilter_unref_buffer(outsamplesref);
avfilter_unref_buffer(insamplesref);
return;
return 0;
}
outsamplesref->audio->sample_rate = outlink->sample_rate;
outsamplesref->audio->nb_samples = n_out;
ff_filter_samples(outlink, outsamplesref);
ret = ff_filter_samples(outlink, outsamplesref);
aresample->req_fullfilled= 1;
avfilter_unref_buffer(insamplesref);
return ret;
}
static int request_frame(AVFilterLink *outlink)

5
libavfilter/af_asetnsamples.c
View File

@@ -131,7 +131,7 @@ static int push_samples(AVFilterLink *outlink)
return nb_out_samples;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
ASNSContext *asns = ctx->priv;
@@ -145,7 +145,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
if (ret < 0) {
av_log(ctx, AV_LOG_ERROR,
"Stretching audio fifo failed, discarded %d samples\n", nb_samples);
return;
return -1;
}
}
av_audio_fifo_write(asns->fifo, (void **)insamples->extended_data, nb_samples);
@@ -155,6 +155,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
if (av_audio_fifo_size(asns->fifo) >= asns->nb_out_samples)
push_samples(outlink);
return 0;
}
static int request_frame(AVFilterLink *outlink)

4
libavfilter/af_ashowinfo.c
View File

@@ -40,7 +40,7 @@ static av_cold int init(AVFilterContext *ctx, const char *args)
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterContext *ctx = inlink->dst;
ShowInfoContext *showinfo = ctx->priv;
@@ -83,7 +83,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
av_log(ctx, AV_LOG_INFO, "]\n");
showinfo->frame++;
ff_filter_samples(inlink->dst->outputs[0], samplesref);
return ff_filter_samples(inlink->dst->outputs[0], samplesref);
}
AVFilter avfilter_af_ashowinfo = {

9
libavfilter/af_astreamsync.c
View File

@@ -107,11 +107,12 @@ static int config_output(AVFilterLink *outlink)
return 0;
}
static void send_out(AVFilterContext *ctx, int out_id)
static int send_out(AVFilterContext *ctx, int out_id)
{
AStreamSyncContext *as = ctx->priv;
struct buf_queue *queue = &as->queue[out_id];
AVFilterBufferRef *buf = queue->buf[queue->tail];
int ret;
queue->buf[queue->tail] = NULL;
as->var_values[VAR_B1 + out_id]++;
@@ -121,11 +122,12 @@ static void send_out(AVFilterContext *ctx, int out_id)
av_q2d(ctx->outputs[out_id]->time_base) * buf->pts;
as->var_values[VAR_T1 + out_id] += buf->audio->nb_samples /
(double)ctx->inputs[out_id]->sample_rate;
ff_filter_samples(ctx->outputs[out_id], buf);
ret = ff_filter_samples(ctx->outputs[out_id], buf);
queue->nb--;
queue->tail = (queue->tail + 1) % QUEUE_SIZE;
if (as->req[out_id])
as->req[out_id]--;
return ret;
}
static void send_next(AVFilterContext *ctx)
@@ -165,7 +167,7 @@ static int request_frame(AVFilterLink *outlink)
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AStreamSyncContext *as = ctx->priv;
@@ -175,6 +177,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
insamples;
as->eof &= ~(1 << id);
send_next(ctx);
return 0;
}
AVFilter avfilter_af_astreamsync = {

44
libavfilter/af_asyncts.c
View File

@@ -37,6 +37,9 @@ typedef struct ASyncContext {
int resample;
float min_delta_sec;
int max_comp;
/* set by filter_samples() to signal an output frame to request_frame() */
int got_output;
} ASyncContext;
#define OFFSET(x) offsetof(ASyncContext, x)
@@ -112,9 +115,13 @@ static int request_frame(AVFilterLink *link)
{
AVFilterContext *ctx = link->src;
ASyncContext *s = ctx->priv;
int ret = ff_request_frame(ctx->inputs[0]);
int ret = 0;
int nb_samples;
s->got_output = 0;
while (ret >= 0 && !s->got_output)
ret = ff_request_frame(ctx->inputs[0]);
/* flush the fifo */
if (ret == AVERROR_EOF && (nb_samples = avresample_get_delay(s->avr))) {
AVFilterBufferRef *buf = ff_get_audio_buffer(link, AV_PERM_WRITE,
@@ -124,18 +131,18 @@ static int request_frame(AVFilterLink *link)
avresample_convert(s->avr, (void**)buf->extended_data, buf->linesize[0],
nb_samples, NULL, 0, 0);
buf->pts = s->pts;
ff_filter_samples(link, buf);
return 0;
return ff_filter_samples(link, buf);
}
return ret;
}
static void write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
static int write_to_fifo(ASyncContext *s, AVFilterBufferRef *buf)
{
avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
int ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
buf->linesize[0], buf->audio->nb_samples);
avfilter_unref_buffer(buf);
return ret;
}
/* get amount of data currently buffered, in samples */
@@ -144,7 +151,7 @@ static int64_t get_delay(ASyncContext *s)
return avresample_available(s->avr) + avresample_get_delay(s->avr);
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ASyncContext *s = ctx->priv;
@@ -152,7 +159,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
int nb_channels = av_get_channel_layout_nb_channels(buf->audio->channel_layout);
int64_t pts = (buf->pts == AV_NOPTS_VALUE) ? buf->pts :
av_rescale_q(buf->pts, inlink->time_base, outlink->time_base);
int out_size;
int out_size, ret;
int64_t delta;
/* buffer data until we get the first timestamp */
@@ -160,14 +167,12 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
if (pts != AV_NOPTS_VALUE) {
s->pts = pts - get_delay(s);
}
write_to_fifo(s, buf);
return;
return write_to_fifo(s, buf);
}
/* now wait for the next timestamp */
if (pts == AV_NOPTS_VALUE) {
write_to_fifo(s, buf);
return;
return write_to_fifo(s, buf);
}
/* when we have two timestamps, compute how many samples would we have
@@ -190,8 +195,10 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
if (out_size > 0) {
AVFilterBufferRef *buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE,
out_size);
if (!buf_out)
return;
if (!buf_out) {
ret = AVERROR(ENOMEM);
goto fail;
}
avresample_read(s->avr, (void**)buf_out->extended_data, out_size);
buf_out->pts = s->pts;
@@ -200,7 +207,10 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
av_samples_set_silence(buf_out->extended_data, out_size - delta,
delta, nb_channels, buf->format);
}
ff_filter_samples(outlink, buf_out);
ret = ff_filter_samples(outlink, buf_out);
if (ret < 0)
goto fail;
s->got_output = 1;
} else {
av_log(ctx, AV_LOG_WARNING, "Non-monotonous timestamps, dropping "
"whole buffer.\n");
@@ -210,9 +220,13 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
avresample_read(s->avr, NULL, avresample_available(s->avr));
s->pts = pts - avresample_get_delay(s->avr);
avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
ret = avresample_convert(s->avr, NULL, 0, 0, (void**)buf->extended_data,
buf->linesize[0], buf->audio->nb_samples);
fail:
avfilter_unref_buffer(buf);
return ret;
}
AVFilter avfilter_af_asyncts = {

3
libavfilter/af_atempo.c
View File

@@ -1040,7 +1040,7 @@ static void push_samples(ATempoContext *atempo,
atempo->nsamples_out += n_out;
}
static void filter_samples(AVFilterLink *inlink,
static int filter_samples(AVFilterLink *inlink,
AVFilterBufferRef *src_buffer)
{
AVFilterContext *ctx = inlink->dst;
@@ -1074,6 +1074,7 @@ static void filter_samples(AVFilterLink *inlink,
atempo->nsamples_in += n_in;
avfilter_unref_bufferp(&src_buffer);
return 0;
}
static int request_frame(AVFilterLink *outlink)

10
libavfilter/af_channelmap.c
View File

@@ -313,7 +313,7 @@ static int channelmap_query_formats(AVFilterContext *ctx)
return 0;
}
static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
@@ -330,8 +330,10 @@ static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *b
if (nch_out > FF_ARRAY_ELEMS(buf->data)) {
uint8_t **new_extended_data =
av_mallocz(nch_out * sizeof(*buf->extended_data));
if (!new_extended_data)
return;
if (!new_extended_data) {
avfilter_unref_buffer(buf);
return AVERROR(ENOMEM);
}
if (buf->extended_data == buf->data) {
buf->extended_data = new_extended_data;
} else {
@@ -353,7 +355,7 @@ static void channelmap_filter_samples(AVFilterLink *inlink, AVFilterBufferRef *b
memcpy(buf->data, buf->extended_data,
FFMIN(FF_ARRAY_ELEMS(buf->data), nch_out) * sizeof(buf->data[0]));
ff_filter_samples(outlink, buf);
return ff_filter_samples(outlink, buf);
}
static int channelmap_config_input(AVFilterLink *inlink)

15
libavfilter/af_channelsplit.c
View File

@@ -105,24 +105,29 @@ static int query_formats(AVFilterContext *ctx)
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
int i;
int i, ret = 0;
for (i = 0; i < ctx->nb_outputs; i++) {
AVFilterBufferRef *buf_out = avfilter_ref_buffer(buf, ~AV_PERM_WRITE);
if (!buf_out)
return;
if (!buf_out) {
ret = AVERROR(ENOMEM);
break;
}
buf_out->data[0] = buf_out->extended_data[0] = buf_out->extended_data[i];
buf_out->audio->channel_layout =
av_channel_layout_extract_channel(buf->audio->channel_layout, i);
ff_filter_samples(ctx->outputs[i], buf_out);
ret = ff_filter_samples(ctx->outputs[i], buf_out);
if (ret < 0)
break;
}
avfilter_unref_buffer(buf);
return ret;
}
AVFilter avfilter_af_channelsplit = {

7
libavfilter/af_earwax.c
View File

@@ -120,13 +120,15 @@ static inline int16_t *scalarproduct(const int16_t *in, const int16_t *endin, in
return out;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterLink *outlink = inlink->dst->outputs[0];
int16_t *taps, *endin, *in, *out;
AVFilterBufferRef *outsamples =
ff_get_audio_buffer(inlink, AV_PERM_WRITE,
insamples->audio->nb_samples);
int ret;
avfilter_copy_buffer_ref_props(outsamples, insamples);
taps = ((EarwaxContext *)inlink->dst->priv)->taps;
@@ -144,8 +146,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
// save part of input for next round
memcpy(taps, endin, NUMTAPS * sizeof(*taps));
ff_filter_samples(outlink, outsamples);
ret = ff_filter_samples(outlink, outsamples);
avfilter_unref_buffer(insamples);
return ret;
}
AVFilter avfilter_af_earwax = {

8
libavfilter/af_join.c
View File

@@ -92,7 +92,7 @@ static const AVClass join_class = {
.version = LIBAVUTIL_VERSION_INT,
};
static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = link->dst;
JoinContext *s = ctx->priv;
@@ -104,6 +104,8 @@ static void filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
av_assert0(i < ctx->nb_inputs);
av_assert0(!s->input_frames[i]);
s->input_frames[i] = buf;
return 0;
}
static int parse_maps(AVFilterContext *ctx)
@@ -468,11 +470,11 @@ static int join_request_frame(AVFilterLink *outlink)
priv->nb_in_buffers = ctx->nb_inputs;
buf->buf->priv = priv;
ff_filter_samples(outlink, buf);
ret = ff_filter_samples(outlink, buf);
memset(s->input_frames, 0, sizeof(*s->input_frames) * ctx->nb_inputs);
return 0;
return ret;
fail:
avfilter_unref_buffer(buf);

6
libavfilter/af_pan.c
View File

@@ -343,8 +343,9 @@ static int config_props(AVFilterLink *link)
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
int ret;
int n = insamples->audio->nb_samples;
AVFilterLink *const outlink = inlink->dst->outputs[0];
AVFilterBufferRef *outsamples = ff_get_audio_buffer(outlink, AV_PERM_WRITE, n);
@@ -354,8 +355,9 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
avfilter_copy_buffer_ref_props(outsamples, insamples);
outsamples->audio->channel_layout = outlink->channel_layout;
ff_filter_samples(outlink, outsamples);
ret = ff_filter_samples(outlink, outsamples);
avfilter_unref_buffer(insamples);
return ret;
}
static av_cold void uninit(AVFilterContext *ctx)

45
libavfilter/af_resample.c
View File

@@ -38,6 +38,9 @@ typedef struct ResampleContext {
AVAudioResampleContext *avr;
int64_t next_pts;
/* set by filter_samples() to signal an output frame to request_frame() */
int got_output;
} ResampleContext;
static av_cold void uninit(AVFilterContext *ctx)
@@ -102,12 +105,6 @@ static int config_output(AVFilterLink *outlink)
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;
@@ -130,7 +127,11 @@ static int request_frame(AVFilterLink *outlink)
{
AVFilterContext *ctx = outlink->src;
ResampleContext *s = ctx->priv;
int ret = ff_request_frame(ctx->inputs[0]);
int ret = 0;
s->got_output = 0;
while (ret >= 0 && !s->got_output)
ret = ff_request_frame(ctx->inputs[0]);
/* flush the lavr delay buffer */
if (ret == AVERROR_EOF && s->avr) {
@@ -156,21 +157,21 @@ static int request_frame(AVFilterLink *outlink)
}
buf->pts = s->next_pts;
ff_filter_samples(outlink, buf);
return 0;
return ff_filter_samples(outlink, buf);
}
return ret;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
AVFilterContext *ctx = inlink->dst;
ResampleContext *s = ctx->priv;
AVFilterLink *outlink = ctx->outputs[0];
int ret;
if (s->avr) {
AVFilterBufferRef *buf_out;
int delay, nb_samples, ret;
int delay, nb_samples;
/* maximum possible samples lavr can output */
delay = avresample_get_delay(s->avr);
@@ -179,10 +180,19 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
AV_ROUND_UP);
buf_out = ff_get_audio_buffer(outlink, AV_PERM_WRITE, nb_samples);
if (!buf_out) {
ret = AVERROR(ENOMEM);
goto fail;
}
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);
if (ret < 0) {
avfilter_unref_buffer(buf_out);
goto fail;
}
av_assert0(!avresample_available(s->avr));
@@ -208,11 +218,18 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *buf)
s->next_pts = buf_out->pts + buf_out->audio->nb_samples;
ff_filter_samples(outlink, buf_out);
ret = ff_filter_samples(outlink, buf_out);
s->got_output = 1;
}
fail:
avfilter_unref_buffer(buf);
} else
ff_filter_samples(outlink, buf);
} else {
ret = ff_filter_samples(outlink, buf);
s->got_output = 1;
}
return ret;
}
AVFilter avfilter_af_resample = {

4
libavfilter/af_silencedetect.c
View File

@@ -78,7 +78,7 @@ static av_cold int init(AVFilterContext *ctx, const char *args)
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
int i;
SilenceDetectContext *silence = inlink->dst->priv;
@@ -118,7 +118,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
}
ff_filter_samples(inlink->dst->outputs[0], insamples);
return ff_filter_samples(inlink->dst->outputs[0], insamples);
}
static int query_formats(AVFilterContext *ctx)

4
libavfilter/af_volume.c
View File

@@ -110,7 +110,7 @@ static int query_formats(AVFilterContext *ctx)
return 0;
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
VolumeContext *vol = inlink->dst->priv;
AVFilterLink *outlink = inlink->dst->outputs[0];
@@ -169,7 +169,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
}
}
ff_filter_samples(outlink, insamples);
return ff_filter_samples(outlink, insamples);
}
AVFilter avfilter_af_volume = {

5
libavfilter/asink_anullsink.c
View File

@@ -21,7 +21,10 @@
#include "avfilter.h"
#include "internal.h"
static void null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref) { }
static int null_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
return 0;
}
AVFilter avfilter_asink_anullsink = {
.name = "anullsink",

24
libavfilter/audio.c
View File

@@ -150,19 +150,19 @@ fail:
return NULL;
}
static void default_filter_samples(AVFilterLink *link,
static int default_filter_samples(AVFilterLink *link,
AVFilterBufferRef *samplesref)
{
ff_filter_samples(link->dst->outputs[0], samplesref);
return ff_filter_samples(link->dst->outputs[0], samplesref);
}
void ff_filter_samples_framed(AVFilterLink *link,
AVFilterBufferRef *samplesref)
int ff_filter_samples_framed(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
void (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
int (*filter_samples)(AVFilterLink *, AVFilterBufferRef *);
AVFilterPad *dst = link->dstpad;
int64_t pts;
AVFilterBufferRef *buf_out;
int ret;
FF_TPRINTF_START(NULL, filter_samples); ff_tlog_link(NULL, link, 1);
@@ -193,21 +193,22 @@ void ff_filter_samples_framed(AVFilterLink *link,
link->cur_buf = buf_out;
pts = buf_out->pts;
filter_samples(link, buf_out);
ret = filter_samples(link, buf_out);
ff_update_link_current_pts(link, pts);
return ret;
}
void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
int insamples = samplesref->audio->nb_samples, inpos = 0, nb_samples;
AVFilterBufferRef *pbuf = link->partial_buf;
int nb_channels = av_get_channel_layout_nb_channels(link->channel_layout);
int ret = 0;
if (!link->min_samples ||
(!pbuf &&
insamples >= link->min_samples && insamples <= link->max_samples)) {
ff_filter_samples_framed(link, samplesref);
return;
return ff_filter_samples_framed(link, samplesref);
}
/* Handle framing (min_samples, max_samples) */
while (insamples) {
@@ -218,7 +219,7 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
if (!pbuf) {
av_log(link->dst, AV_LOG_WARNING,
"Samples dropped due to memory allocation failure.\n");
return;
return 0;
}
avfilter_copy_buffer_ref_props(pbuf, samplesref);
pbuf->pts = samplesref->pts +
@@ -234,10 +235,11 @@ void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
insamples -= nb_samples;
pbuf->audio->nb_samples += nb_samples;
if (pbuf->audio->nb_samples >= link->min_samples) {
ff_filter_samples_framed(link, pbuf);
ret = ff_filter_samples_framed(link, pbuf);
pbuf = NULL;
}
}
avfilter_unref_buffer(samplesref);
link->partial_buf = pbuf;
return ret;
}

7
libavfilter/audio.h
View File

@@ -70,14 +70,17 @@ AVFilterBufferRef *ff_get_audio_buffer(AVFilterLink *link, int perms,
* @param samplesref a reference to the buffer of audio samples being sent. The
* receiving filter will free this reference when it no longer
* needs it or pass it on to the next filter.
*
* @return >= 0 on success, a negative AVERROR on error. The receiving filter
* is responsible for unreferencing samplesref in case of error.
*/
void ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
int ff_filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref);
/**
* Send a buffer of audio samples to the next link, without checking
* min_samples.
*/
void ff_filter_samples_framed(AVFilterLink *link,
int ff_filter_samples_framed(AVFilterLink *link,
AVFilterBufferRef *samplesref);
#endif /* AVFILTER_AUDIO_H */

3
libavfilter/avf_showwaves.c
View File

@@ -180,7 +180,7 @@ static int request_frame(AVFilterLink *outlink)
#define MAX_INT16 ((1<<15) -1)
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
@@ -225,6 +225,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
}
avfilter_unref_buffer(insamples);
return 0;
}
AVFilter avfilter_avf_showwaves = {

6
libavfilter/avfilter.h
View File

@@ -301,8 +301,12 @@ struct AVFilterPad {
* and should do its processing.
*
* Input audio pads only.
*
* @return >= 0 on success, a negative AVERROR on error. This function
* must ensure that samplesref is properly unreferenced on error if it
* hasn't been passed on to another filter.
*/
void (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
/**
* Frame poll callback. This returns the number of immediately available

8
libavfilter/buffersink.c
View File

@@ -56,6 +56,12 @@ static void start_frame(AVFilterLink *link, AVFilterBufferRef *buf)
link->cur_buf = NULL;
};
static int filter_samples(AVFilterLink *link, AVFilterBufferRef *buf)
{
start_frame(link, buf);
return 0;
}
int av_buffersink_read(AVFilterContext *ctx, AVFilterBufferRef **buf)
{
BufferSinkContext *s = ctx->priv;
@@ -160,7 +166,7 @@ AVFilter avfilter_asink_abuffer = {
.inputs = (AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_AUDIO,
.filter_samples = start_frame,
.filter_samples = filter_samples,
.min_perms = AV_PERM_READ,
.needs_fifo = 1 },
{ .name = NULL }},

5
libavfilter/buffersrc.c
View File

@@ -408,6 +408,7 @@ static int request_frame(AVFilterLink *link)
{
BufferSourceContext *c = link->src->priv;
AVFilterBufferRef *buf;
int ret = 0;
if (!av_fifo_size(c->fifo)) {
if (c->eof)
@@ -424,7 +425,7 @@ static int request_frame(AVFilterLink *link)
ff_end_frame(link);
break;
case AVMEDIA_TYPE_AUDIO:
ff_filter_samples(link, avfilter_ref_buffer(buf, ~0));
ret = ff_filter_samples(link, avfilter_ref_buffer(buf, ~0));
break;
default:
return AVERROR(EINVAL);
@@ -432,7 +433,7 @@ static int request_frame(AVFilterLink *link)
avfilter_unref_buffer(buf);
return 0;
return ret;
}
static int poll_frame(AVFilterLink *link)

4
libavfilter/f_settb.c
View File

@@ -117,7 +117,7 @@ static void start_frame(AVFilterLink *inlink, AVFilterBufferRef *picref)
ff_start_frame(outlink, picref2);
}
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
{
AVFilterContext *ctx = inlink->dst;
AVFilterLink *outlink = ctx->outputs[0];
@@ -132,7 +132,7 @@ static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *insamples)
avfilter_unref_buffer(insamples);
}
ff_filter_samples(outlink, outsamples);
return ff_filter_samples(outlink, outsamples);
}
#if CONFIG_SETTB_FILTER

26
libavfilter/fifo.c
View File

@@ -72,13 +72,25 @@ static av_cold void uninit(AVFilterContext *ctx)
avfilter_unref_buffer(fifo->buf_out);
}
static void add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf)
static int add_to_queue(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
FifoContext *fifo = inlink->dst->priv;
fifo->last->next = av_mallocz(sizeof(Buf));
if (!fifo->last->next) {
avfilter_unref_buffer(buf);
return AVERROR(ENOMEM);
}
fifo->last = fifo->last->next;
fifo->last->buf = buf;
return 0;
}
static void start_frame(AVFilterLink *inlink, AVFilterBufferRef *buf)
{
add_to_queue(inlink, buf);
}
static void queue_pop(FifoContext *s)
@@ -210,15 +222,13 @@ static int return_audio_frame(AVFilterContext *ctx)
buf_out = s->buf_out;
s->buf_out = NULL;
}
ff_filter_samples(link, buf_out);
return 0;
return ff_filter_samples(link, buf_out);
}
static int request_frame(AVFilterLink *outlink)
{
FifoContext *fifo = outlink->src->priv;
int ret;
int ret = 0;
if (!fifo->root.next) {
if ((ret = ff_request_frame(outlink->src->inputs[0])) < 0)
@@ -238,7 +248,7 @@ static int request_frame(AVFilterLink *outlink)
if (outlink->request_samples) {
return return_audio_frame(outlink->src);
} else {
ff_filter_samples(outlink, fifo->root.next->buf);
ret = ff_filter_samples(outlink, fifo->root.next->buf);
queue_pop(fifo);
}
break;
@@ -246,7 +256,7 @@ static int request_frame(AVFilterLink *outlink)
return AVERROR(EINVAL);
}
return 0;
return ret;
}
AVFilter avfilter_vf_fifo = {
@@ -261,7 +271,7 @@ AVFilter avfilter_vf_fifo = {
.inputs = (const AVFilterPad[]) {{ .name = "default",
.type = AVMEDIA_TYPE_VIDEO,
.get_video_buffer= ff_null_get_video_buffer,
.start_frame = add_to_queue,
.start_frame = start_frame,
.draw_slice = draw_slice,
.end_frame = end_frame,
.rej_perms = AV_PERM_REUSE2, },

6
libavfilter/internal.h
View File

@@ -135,8 +135,12 @@ struct AVFilterPad {
* and should do its processing.
*
* Input audio pads only.
*
* @return >= 0 on success, a negative AVERROR on error. This function
* must ensure that samplesref is properly unreferenced on error if it
* hasn't been passed on to another filter.
*/
void (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
int (*filter_samples)(AVFilterLink *link, AVFilterBufferRef *samplesref);
/**
* Frame poll callback. This returns the number of immediately available

3
libavfilter/sink_buffer.c
View File

@@ -244,9 +244,10 @@ AVFilter avfilter_vsink_buffersink = {
#if CONFIG_ABUFFERSINK_FILTER
static void filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
static int filter_samples(AVFilterLink *link, AVFilterBufferRef *samplesref)
{
end_frame(link);
return 0;
}
static av_cold int asink_init(AVFilterContext *ctx, const char *args)

12
libavfilter/split.c
View File

@@ -110,15 +110,19 @@ AVFilter avfilter_vf_split = {
.outputs = (AVFilterPad[]) {{ .name = NULL}},
};
static void filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
static int filter_samples(AVFilterLink *inlink, AVFilterBufferRef *samplesref)
{
AVFilterContext *ctx = inlink->dst;
int i;
int i, ret = 0;
for (i = 0; i < ctx->nb_outputs; i++)
ff_filter_samples(inlink->dst->outputs[i],
for (i = 0; i < ctx->nb_outputs; i++) {
ret = ff_filter_samples(inlink->dst->outputs[i],
avfilter_ref_buffer(samplesref, ~AV_PERM_WRITE));
if (ret < 0)
break;
}
avfilter_unref_buffer(samplesref);
return ret;
}
AVFilter avfilter_af_asplit = {

5
libavformat/matroskadec.c
View File

@@ -842,8 +842,11 @@ static int ebml_parse_id(MatroskaDemuxContext *matroska, EbmlSyntax *syntax,
matroska->num_levels > 0 &&
matroska->levels[matroska->num_levels-1].length == 0xffffffffffffff)
return 0; // we reached the end of an unknown size cluster
if (!syntax[i].id && id != EBML_ID_VOID && id != EBML_ID_CRC32)
if (!syntax[i].id && id != EBML_ID_VOID && id != EBML_ID_CRC32) {
av_log(matroska->ctx, AV_LOG_INFO, "Unknown entry 0x%X\n", id);
if (matroska->ctx->error_recognition & AV_EF_EXPLODE)
return AVERROR_INVALIDDATA;
}
return ebml_parse_elem(matroska, &syntax[i], data);
}

11
libavformat/tcp.c
View File

@@ -43,7 +43,7 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
char buf[256];
int ret;
socklen_t optlen;
int timeout = 50;
int timeout = 50, listen_timeout = -1;
char hostname[1024],proto[1024],path[1024];
char portstr[10];
@@ -59,6 +59,9 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
if (av_find_info_tag(buf, sizeof(buf), "timeout", p)) {
timeout = strtol(buf, NULL, 10);
}
if (av_find_info_tag(buf, sizeof(buf), "listen_timeout", p)) {
listen_timeout = strtol(buf, NULL, 10);
}
}
hints.ai_family = AF_UNSPEC;
hints.ai_socktype = SOCK_STREAM;
@@ -87,6 +90,7 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
if (listen_socket) {
int fd1;
int reuse = 1;
struct pollfd lp = { fd, POLLIN, 0 };
setsockopt(fd, SOL_SOCKET, SO_REUSEADDR, &reuse, sizeof(reuse));
ret = bind(fd, cur_ai->ai_addr, cur_ai->ai_addrlen);
if (ret) {
@@ -98,6 +102,11 @@ static int tcp_open(URLContext *h, const char *uri, int flags)
ret = ff_neterrno();
goto fail1;
}
ret = poll(&lp, 1, listen_timeout >= 0 ? listen_timeout : -1);
if (ret <= 0) {
ret = AVERROR(ETIMEDOUT);
goto fail1;
}
fd1 = accept(fd, NULL, NULL);
if (fd1 < 0) {
ret = ff_neterrno();

8
libavresample/audio_mix.c
View File

@@ -305,6 +305,14 @@ int ff_audio_mix_init(AVAudioResampleContext *avr)
{
int ret;
if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
"mixing: %s\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
return AVERROR(EINVAL);
}
/* build matrix if the user did not already set one */
if (!avr->am->matrix) {
int i, j;

7
libavresample/avresample.h
View File

@@ -45,6 +45,13 @@ enum AVMixCoeffType {
AV_MIX_COEFF_TYPE_NB, /** Number of coeff types. Not part of ABI */
};
/** Resampling Filter Types */
enum AVResampleFilterType {
AV_RESAMPLE_FILTER_TYPE_CUBIC, /**< Cubic */
AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL, /**< Blackman Nuttall Windowed Sinc */
AV_RESAMPLE_FILTER_TYPE_KAISER, /**< Kaiser Windowed Sinc */
};
/**
* Return the LIBAVRESAMPLE_VERSION_INT constant.
*/

2
libavresample/internal.h
View File

@@ -50,6 +50,8 @@ struct AVAudioResampleContext {
int phase_shift; /**< log2 of the number of entries in the resampling polyphase filterbank */
int linear_interp; /**< if 1 then the resampling FIR filter will be linearly interpolated */
double cutoff; /**< resampling cutoff frequency. 1.0 corresponds to half the output sample rate */
enum AVResampleFilterType filter_type; /**< resampling filter type */
int kaiser_beta; /**< beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
int in_channels; /**< number of input channels */
int out_channels; /**< number of output channels */

7
libavresample/options.c
View File

@@ -39,7 +39,7 @@ static const AVOption options[] = {
{ "out_channel_layout", "Output Channel Layout", OFFSET(out_channel_layout), AV_OPT_TYPE_INT64, { 0 }, INT64_MIN, INT64_MAX, PARAM },
{ "out_sample_fmt", "Output Sample Format", OFFSET(out_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_S16 }, AV_SAMPLE_FMT_U8, AV_SAMPLE_FMT_NB-1, PARAM },
{ "out_sample_rate", "Output Sample Rate", OFFSET(out_sample_rate), AV_OPT_TYPE_INT, { 48000 }, 1, INT_MAX, PARAM },
{ "internal_sample_fmt", "Internal Sample Format", OFFSET(internal_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_FLTP }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, PARAM },
{ "internal_sample_fmt", "Internal Sample Format", OFFSET(internal_sample_fmt), AV_OPT_TYPE_INT, { AV_SAMPLE_FMT_NONE }, AV_SAMPLE_FMT_NONE, AV_SAMPLE_FMT_NB-1, PARAM },
{ "mix_coeff_type", "Mixing Coefficient Type", OFFSET(mix_coeff_type), AV_OPT_TYPE_INT, { AV_MIX_COEFF_TYPE_FLT }, AV_MIX_COEFF_TYPE_Q8, AV_MIX_COEFF_TYPE_NB-1, PARAM, "mix_coeff_type" },
{ "q8", "16-bit 8.8 Fixed-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q8 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
{ "q15", "32-bit 17.15 Fixed-Point", 0, AV_OPT_TYPE_CONST, { AV_MIX_COEFF_TYPE_Q15 }, INT_MIN, INT_MAX, PARAM, "mix_coeff_type" },
@@ -56,6 +56,11 @@ static const AVOption options[] = {
{ "none", "None", 0, AV_OPT_TYPE_CONST, { AV_MATRIX_ENCODING_NONE }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "dolby", "Dolby", 0, AV_OPT_TYPE_CONST, { AV_MATRIX_ENCODING_DOLBY }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "dplii", "Dolby Pro Logic II", 0, AV_OPT_TYPE_CONST, { AV_MATRIX_ENCODING_DPLII }, INT_MIN, INT_MAX, PARAM, "matrix_encoding" },
{ "filter_type", "Filter Type", OFFSET(filter_type), AV_OPT_TYPE_INT, { AV_RESAMPLE_FILTER_TYPE_KAISER }, AV_RESAMPLE_FILTER_TYPE_CUBIC, AV_RESAMPLE_FILTER_TYPE_KAISER, PARAM, "filter_type" },
{ "cubic", "Cubic", 0, AV_OPT_TYPE_CONST, { AV_RESAMPLE_FILTER_TYPE_CUBIC }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "blackman_nuttall", "Blackman Nuttall Windowed Sinc", 0, AV_OPT_TYPE_CONST, { AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "kaiser", "Kaiser Windowed Sinc", 0, AV_OPT_TYPE_CONST, { AV_RESAMPLE_FILTER_TYPE_KAISER }, INT_MIN, INT_MAX, PARAM, "filter_type" },
{ "kaiser_beta", "Kaiser Window Beta", OFFSET(kaiser_beta), AV_OPT_TYPE_INT, { 9 }, 2, 16, PARAM },
{ NULL },
};

180
libavresample/resample.c
View File

@@ -24,37 +24,10 @@
#include "internal.h"
#include "audio_data.h"
#ifdef CONFIG_RESAMPLE_FLT
/* float template */
#define FILTER_SHIFT 0
#define FELEM float
#define FELEM2 float
#define FELEML float
#define WINDOW_TYPE 24
#elifdef CONFIG_RESAMPLE_S32
/* s32 template */
#define FILTER_SHIFT 30
#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define FELEM_MAX INT32_MAX
#define FELEM_MIN INT32_MIN
#define WINDOW_TYPE 12
#else
/* s16 template */
#define FILTER_SHIFT 15
#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define FELEM_MAX INT16_MAX
#define FELEM_MIN INT16_MIN
#define WINDOW_TYPE 9
#endif
struct ResampleContext {
AVAudioResampleContext *avr;
AudioData *buffer;
FELEM *filter_bank;
uint8_t *filter_bank;
int filter_length;
int ideal_dst_incr;
int dst_incr;
@@ -65,9 +38,35 @@ struct ResampleContext {
int phase_shift;
int phase_mask;
int linear;
enum AVResampleFilterType filter_type;
int kaiser_beta;
double factor;
void (*set_filter)(void *filter, double *tab, int phase, int tap_count);
void (*resample_one)(struct ResampleContext *c, int no_filter, void *dst0,
int dst_index, const void *src0, int src_size,
int index, int frac);
};
/* double template */
#define CONFIG_RESAMPLE_DBL
#include "resample_template.c"
#undef CONFIG_RESAMPLE_DBL
/* float template */
#define CONFIG_RESAMPLE_FLT
#include "resample_template.c"
#undef CONFIG_RESAMPLE_FLT
/* s32 template */
#define CONFIG_RESAMPLE_S32
#include "resample_template.c"
#undef CONFIG_RESAMPLE_S32
/* s16 template */
#include "resample_template.c"
/**
* 0th order modified bessel function of the first kind.
*/
@@ -95,17 +94,17 @@ static double bessel(double x)
* @param tap_count tap count
* @param phase_count phase count
* @param scale wanted sum of coefficients for each filter
* @param type 0->cubic
* 1->blackman nuttall windowed sinc
* 2..16->kaiser windowed sinc beta=2..16
* @param filter_type filter type
* @param kaiser_beta kaiser window beta
* @return 0 on success, negative AVERROR code on failure
*/
static int build_filter(FELEM *filter, double factor, int tap_count,
int phase_count, int scale, int type)
static int build_filter(ResampleContext *c)
{
int ph, i;
double x, y, w;
double x, y, w, factor;
double *tab;
int tap_count = c->filter_length;
int phase_count = 1 << c->phase_shift;
const int center = (tap_count - 1) / 2;
tab = av_malloc(tap_count * sizeof(*tab));
@@ -113,8 +112,7 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
return AVERROR(ENOMEM);
/* if upsampling, only need to interpolate, no filter */
if (factor > 1.0)
factor = 1.0;
factor = FFMIN(c->factor, 1.0);
for (ph = 0; ph < phase_count; ph++) {
double norm = 0;
@@ -122,39 +120,34 @@ static int build_filter(FELEM *filter, double factor, int tap_count,
x = M_PI * ((double)(i - center) - (double)ph / phase_count) * factor;
if (x == 0) y = 1.0;
else y = sin(x) / x;
switch (type) {
case 0: {
switch (c->filter_type) {
case AV_RESAMPLE_FILTER_TYPE_CUBIC: {
const float d = -0.5; //first order derivative = -0.5
x = fabs(((double)(i - center) - (double)ph / phase_count) * factor);
if (x < 1.0) y = 1 - 3 * x*x + 2 * x*x*x + d * ( -x*x + x*x*x);
else y = d * (-4 + 8 * x - 5 * x*x + x*x*x);
break;
}
case 1:
case AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL:
w = 2.0 * x / (factor * tap_count) + M_PI;
y *= 0.3635819 - 0.4891775 * cos( w) +
0.1365995 * cos(2 * w) -
0.0106411 * cos(3 * w);
break;
default:
case AV_RESAMPLE_FILTER_TYPE_KAISER:
w = 2.0 * x / (factor * tap_count * M_PI);
y *= bessel(type * sqrt(FFMAX(1 - w * w, 0)));
y *= bessel(c->kaiser_beta * sqrt(FFMAX(1 - w * w, 0)));
break;
}
tab[i] = y;
norm += y;
}
/* normalize so that an uniform color remains the same */
for (i = 0; i < tap_count; i++) {
#ifdef CONFIG_RESAMPLE_FLT
filter[ph * tap_count + i] = tab[i] / norm;
#else
filter[ph * tap_count + i] = av_clip(lrintf(tab[i] * scale / norm),
FELEM_MIN, FELEM_MAX);
#endif
}
for (i = 0; i < tap_count; i++)
tab[i] = tab[i] / norm;
c->set_filter(c->filter_bank, tab, ph, tap_count);
}
av_free(tab);
@@ -168,9 +161,12 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
int in_rate = avr->in_sample_rate;
double factor = FFMIN(out_rate * avr->cutoff / in_rate, 1.0);
int phase_count = 1 << avr->phase_shift;
int felem_size;
/* TODO: add support for s32 and float internal formats */
if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
if (avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_S32P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_DBLP) {
av_log(avr, AV_LOG_ERROR, "Unsupported internal format for "
"resampling: %s\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
@@ -186,18 +182,40 @@ ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr)
c->linear = avr->linear_interp;
c->factor = factor;
c->filter_length = FFMAX((int)ceil(avr->filter_size / factor), 1);
c->filter_type = avr->filter_type;
c->kaiser_beta = avr->kaiser_beta;
c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * sizeof(FELEM));
switch (avr->internal_sample_fmt) {
case AV_SAMPLE_FMT_DBLP:
c->resample_one = resample_one_dbl;
c->set_filter = set_filter_dbl;
break;
case AV_SAMPLE_FMT_FLTP:
c->resample_one = resample_one_flt;
c->set_filter = set_filter_flt;
break;
case AV_SAMPLE_FMT_S32P:
c->resample_one = resample_one_s32;
c->set_filter = set_filter_s32;
break;
case AV_SAMPLE_FMT_S16P:
c->resample_one = resample_one_s16;
c->set_filter = set_filter_s16;
break;
}
felem_size = av_get_bytes_per_sample(avr->internal_sample_fmt);
c->filter_bank = av_mallocz(c->filter_length * (phase_count + 1) * felem_size);
if (!c->filter_bank)
goto error;
if (build_filter(c->filter_bank, factor, c->filter_length, phase_count,
1 << FILTER_SHIFT, WINDOW_TYPE) < 0)
if (build_filter(c) < 0)
goto error;
memcpy(&c->filter_bank[c->filter_length * phase_count + 1],
c->filter_bank, (c->filter_length - 1) * sizeof(FELEM));
c->filter_bank[c->filter_length * phase_count] = c->filter_bank[c->filter_length - 1];
memcpy(&c->filter_bank[(c->filter_length * phase_count + 1) * felem_size],
c->filter_bank, (c->filter_length - 1) * felem_size);
memcpy(&c->filter_bank[c->filter_length * phase_count * felem_size],
&c->filter_bank[(c->filter_length - 1) * felem_size], felem_size);
c->compensation_distance = 0;
if (!av_reduce(&c->src_incr, &c->dst_incr, out_rate,
@@ -311,10 +329,10 @@ reinit_fail:
return ret;
}
static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
static int resample(ResampleContext *c, void *dst, const void *src,
int *consumed, int src_size, int dst_size, int update_ctx)
{
int dst_index, i;
int dst_index;
int index = c->index;
int frac = c->frac;
int dst_incr_frac = c->dst_incr % c->src_incr;
@@ -334,7 +352,7 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
if (dst) {
for(dst_index = 0; dst_index < dst_size; dst_index++) {
dst[dst_index] = src[index2 >> 32];
c->resample_one(c, 1, dst, dst_index, src, 0, index2 >> 32, 0);
index2 += incr;
}
} else {
@@ -345,42 +363,14 @@ static int resample(ResampleContext *c, int16_t *dst, const int16_t *src,
frac = (frac + dst_index * (int64_t)dst_incr_frac) % c->src_incr;
} else {
for (dst_index = 0; dst_index < dst_size; dst_index++) {
FELEM *filter = c->filter_bank +
c->filter_length * (index & c->phase_mask);
int sample_index = index >> c->phase_shift;
if (!dst && (sample_index + c->filter_length > src_size ||
-sample_index >= src_size))
if (sample_index + c->filter_length > src_size ||
-sample_index >= src_size)
break;
if (dst) {
FELEM2 val = 0;
if (sample_index < 0) {
for (i = 0; i < c->filter_length; i++)
val += src[FFABS(sample_index + i) % src_size] *
(FELEM2)filter[i];
} else if (sample_index + c->filter_length > src_size) {
break;
} else if (c->linear) {
FELEM2 v2 = 0;
for (i = 0; i < c->filter_length; i++) {
val += src[abs(sample_index + i)] * (FELEM2)filter[i];
v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
}
val += (v2 - val) * (FELEML)frac / c->src_incr;
} else {
for (i = 0; i < c->filter_length; i++)
val += src[sample_index + i] * (FELEM2)filter[i];
}
#ifdef CONFIG_RESAMPLE_FLT
dst[dst_index] = av_clip_int16(lrintf(val));
#else
val = (val + (1<<(FILTER_SHIFT-1)))>>FILTER_SHIFT;
dst[dst_index] = av_clip_int16(val);
#endif
}
if (dst)
c->resample_one(c, 0, dst, dst_index, src, src_size, index, frac);
frac += dst_incr_frac;
index += dst_incr;
@@ -451,8 +441,8 @@ int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src,
/* resample each channel plane */
for (ch = 0; ch < c->buffer->channels; ch++) {
out_samples = resample(c, (int16_t *)dst->data[ch],
(const int16_t *)c->buffer->data[ch], consumed,
out_samples = resample(c, (void *)dst->data[ch],
(const void *)c->buffer->data[ch], consumed,
c->buffer->nb_samples, dst->allocated_samples,
ch + 1 == c->buffer->channels);
}

102
libavresample/resample_template.c Normal file
View File

@@ -0,0 +1,102 @@
/*
* Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at>
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#if defined(CONFIG_RESAMPLE_DBL)
#define SET_TYPE(func) func ## _dbl
#define FELEM double
#define FELEM2 double
#define FELEML double
#define OUT(d, v) d = v
#define DBL_TO_FELEM(d, v) d = v
#elif defined(CONFIG_RESAMPLE_FLT)
#define SET_TYPE(func) func ## _flt
#define FELEM float
#define FELEM2 float
#define FELEML float
#define OUT(d, v) d = v
#define DBL_TO_FELEM(d, v) d = v
#elif defined(CONFIG_RESAMPLE_S32)
#define SET_TYPE(func) func ## _s32
#define FELEM int32_t
#define FELEM2 int64_t
#define FELEML int64_t
#define OUT(d, v) d = av_clipl_int32((v + (1 << 29)) >> 30)
#define DBL_TO_FELEM(d, v) d = av_clipl_int32(llrint(v * (1 << 30)));
#else
#define SET_TYPE(func) func ## _s16
#define FELEM int16_t
#define FELEM2 int32_t
#define FELEML int64_t
#define OUT(d, v) d = av_clip_int16((v + (1 << 14)) >> 15)
#define DBL_TO_FELEM(d, v) d = av_clip_int16(lrint(v * (1 << 15)))
#endif
static void SET_TYPE(resample_one)(ResampleContext *c, int no_filter,
void *dst0, int dst_index, const void *src0,
int src_size, int index, int frac)
{
FELEM *dst = dst0;
const FELEM *src = src0;
if (no_filter) {
dst[dst_index] = src[index];
} else {
int i;
int sample_index = index >> c->phase_shift;
FELEM2 val = 0;
FELEM *filter = ((FELEM *)c->filter_bank) +
c->filter_length * (index & c->phase_mask);
if (sample_index < 0) {
for (i = 0; i < c->filter_length; i++)
val += src[FFABS(sample_index + i) % src_size] *
(FELEM2)filter[i];
} else if (c->linear) {
FELEM2 v2 = 0;
for (i = 0; i < c->filter_length; i++) {
val += src[abs(sample_index + i)] * (FELEM2)filter[i];
v2 += src[abs(sample_index + i)] * (FELEM2)filter[i + c->filter_length];
}
val += (v2 - val) * (FELEML)frac / c->src_incr;
} else {
for (i = 0; i < c->filter_length; i++)
val += src[sample_index + i] * (FELEM2)filter[i];
}
OUT(dst[dst_index], val);
}
}
static void SET_TYPE(set_filter)(void *filter0, double *tab, int phase,
int tap_count)
{
int i;
FELEM *filter = ((FELEM *)filter0) + phase * tap_count;
for (i = 0; i < tap_count; i++) {
DBL_TO_FELEM(filter[i], tab[i]);
}
}
#undef SET_TYPE
#undef FELEM
#undef FELEM2
#undef FELEML
#undef OUT
#undef DBL_TO_FELEM

43
libavresample/utils.c
View File

@@ -57,18 +57,43 @@ int avresample_open(AVAudioResampleContext *avr)
avr->resample_needed = avr->in_sample_rate != avr->out_sample_rate ||
avr->force_resampling;
/* set sample format conversion parameters */
/* override user-requested internal format to avoid unexpected failures
TODO: support more internal formats */
if (avr->resample_needed && avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P) {
av_log(avr, AV_LOG_WARNING, "Using s16p as internal sample format\n");
/* select internal sample format if not specified by the user */
if (avr->internal_sample_fmt == AV_SAMPLE_FMT_NONE &&
(avr->mixing_needed || avr->resample_needed)) {
enum AVSampleFormat in_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
enum AVSampleFormat out_fmt = av_get_planar_sample_fmt(avr->out_sample_fmt);
int max_bps = FFMAX(av_get_bytes_per_sample(in_fmt),
av_get_bytes_per_sample(out_fmt));
if (max_bps <= 2) {
avr->internal_sample_fmt = AV_SAMPLE_FMT_S16P;
} else if (avr->mixing_needed &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_S16P &&
avr->internal_sample_fmt != AV_SAMPLE_FMT_FLTP) {
av_log(avr, AV_LOG_WARNING, "Using fltp as internal sample format\n");
} else if (avr->mixing_needed) {
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
} else {
if (max_bps <= 4) {
if (in_fmt == AV_SAMPLE_FMT_S32P ||
out_fmt == AV_SAMPLE_FMT_S32P) {
if (in_fmt == AV_SAMPLE_FMT_FLTP ||
out_fmt == AV_SAMPLE_FMT_FLTP) {
/* if one is s32 and the other is flt, use dbl */
avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
} else {
/* if one is s32 and the other is s32, s16, or u8, use s32 */
avr->internal_sample_fmt = AV_SAMPLE_FMT_S32P;
}
} else {
/* if one is flt and the other is flt, s16 or u8, use flt */
avr->internal_sample_fmt = AV_SAMPLE_FMT_FLTP;
}
} else {
/* if either is dbl, use dbl */
avr->internal_sample_fmt = AV_SAMPLE_FMT_DBLP;
}
}
av_log(avr, AV_LOG_DEBUG, "Using %s as internal sample format\n",
av_get_sample_fmt_name(avr->internal_sample_fmt));
}
/* set sample format conversion parameters */
if (avr->in_channels == 1)
avr->in_sample_fmt = av_get_planar_sample_fmt(avr->in_sample_fmt);
if (avr->out_channels == 1)

8
tests/fate-run.sh
View File

@@ -75,7 +75,13 @@ probefmt(){
}
ffmpeg(){
run ffmpeg -nostats -threads $threads -thread_type $thread_type -cpuflags $cpuflags "$@"
dec_opts="-threads $threads -thread_type $thread_type"
ffmpeg_args="-nostats -cpuflags $cpuflags"
for arg in $@; do
[ ${arg} = -i ] && ffmpeg_args="${ffmpeg_args} ${dec_opts}"
ffmpeg_args="${ffmpeg_args} ${arg}"
done
run ffmpeg ${ffmpeg_args}
}
framecrc(){