Newer
Older
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
/*
* avconv main
* Copyright (c) 2000-2011 The libav developers.
*
* 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
*/
#include "config.h"
#include <ctype.h>
#include <string.h>
#include <math.h>
#include <stdlib.h>
#include <errno.h>
#include <signal.h>
#include <limits.h>
#include <unistd.h>
#include "libavformat/avformat.h"
#include "libavdevice/avdevice.h"
#include "libswscale/swscale.h"
#include "libavutil/opt.h"
#include "libavcodec/audioconvert.h"
#include "libavutil/audioconvert.h"
#include "libavutil/parseutils.h"
#include "libavutil/samplefmt.h"
#include "libavutil/colorspace.h"
#include "libavutil/fifo.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/dict.h"
#include "libavutil/mathematics.h"
#include "libavutil/pixdesc.h"
#include "libavutil/avstring.h"
#include "libavutil/libm.h"
#include "libavutil/imgutils.h"
#include "libavformat/os_support.h"
#if CONFIG_AVFILTER
# include "libavfilter/avfilter.h"
# include "libavfilter/avfiltergraph.h"
# include "libavfilter/buffersrc.h"
# include "libavfilter/vsrc_buffer.h"
#endif
#if HAVE_SYS_RESOURCE_H
#include <sys/types.h>
#include <sys/time.h>
#include <sys/resource.h>
#elif HAVE_GETPROCESSTIMES
#include <windows.h>
#endif
#if HAVE_GETPROCESSMEMORYINFO
#include <windows.h>
#include <psapi.h>
#endif
#if HAVE_SYS_SELECT_H
#include <sys/select.h>
#endif
#include <time.h>
#include "cmdutils.h"
#include "libavutil/avassert.h"
const char program_name[] = "avconv";
const int program_birth_year = 2000;
/* select an input stream for an output stream */
typedef struct StreamMap {
int disabled; /** 1 is this mapping is disabled by a negative map */
int file_index;
int stream_index;
int sync_file_index;
int sync_stream_index;
} StreamMap;
/**
* select an input file for an output file
*/
typedef struct MetadataMap {
int file; ///< file index
char type; ///< type of metadata to copy -- (g)lobal, (s)tream, (c)hapter or (p)rogram
int index; ///< stream/chapter/program number
} MetadataMap;
static const OptionDef options[];
static int video_discard = 0;
static int do_deinterlace = 0;
static int intra_dc_precision = 8;
static int qp_hist = 0;
static int file_overwrite = 0;
static int do_benchmark = 0;
static int do_hex_dump = 0;
static int do_pkt_dump = 0;
static int do_pass = 0;
static char *pass_logfilename_prefix = NULL;
static int video_sync_method= -1;
static int audio_sync_method= 0;
static float audio_drift_threshold= 0.1;
static int copy_ts= 0;
static int opt_shortest = 0;
static char *vstats_filename;
static FILE *vstats_file;
static int audio_volume = 256;
static int exit_on_error = 0;
static int using_stdin = 0;
static int64_t video_size = 0;
static int64_t audio_size = 0;
static int64_t extra_size = 0;
static int nb_frames_dup = 0;
static int nb_frames_drop = 0;
static int input_sync;
static float dts_delta_threshold = 10;
static int print_stats = 1;
static uint8_t *audio_buf;
static uint8_t *audio_out;
static unsigned int allocated_audio_out_size, allocated_audio_buf_size;
#define DEFAULT_PASS_LOGFILENAME_PREFIX "av2pass"
typedef struct FrameBuffer {
uint8_t *base[4];
uint8_t *data[4];
int linesize[4];
int h, w;
enum PixelFormat pix_fmt;
int refcount;
struct InputStream *ist;
struct FrameBuffer *next;
} FrameBuffer;
typedef struct InputStream {
int file_index;
AVStream *st;
int discard; /* true if stream data should be discarded */
int decoding_needed; /* true if the packets must be decoded in 'raw_fifo' */
AVCodec *dec;
AVFrame *decoded_frame;
AVFrame *filtered_frame;
int64_t start; /* time when read started */
int64_t next_pts; /* synthetic pts for cases where pkt.pts
is not defined */
int64_t pts; /* current pts */
PtsCorrectionContext pts_ctx;
double ts_scale;
int is_start; /* is 1 at the start and after a discontinuity */
int showed_multi_packet_warning;
AVDictionary *opts;
/* a pool of free buffers for decoded data */
FrameBuffer *buffer_pool;
} InputStream;
typedef struct InputFile {
AVFormatContext *ctx;
int eof_reached; /* true if eof reached */
int ist_index; /* index of first stream in ist_table */
int buffer_size; /* current total buffer size */
int64_t ts_offset;
int nb_streams; /* number of stream that avconv is aware of; may be different
from ctx.nb_streams if new streams appear during av_read_frame() */
typedef struct OutputStream {
int file_index; /* file index */
int index; /* stream index in the output file */
int source_index; /* InputStream index */
AVStream *st; /* stream in the output file */
int encoding_needed; /* true if encoding needed for this stream */
int frame_number;
/* input pts and corresponding output pts
for A/V sync */
//double sync_ipts; /* dts from the AVPacket of the demuxer in second units */
struct InputStream *sync_ist; /* input stream to sync against */
int64_t sync_opts; /* output frame counter, could be changed to some true timestamp */ //FIXME look at frame_number
AVBitStreamFilterContext *bitstream_filters;
AVCodec *enc;
/* video only */
int video_resample;
AVFrame pict_tmp; /* temporary image for resampling */
struct SwsContext *img_resample_ctx; /* for image resampling */
int resample_height;
int resample_width;
int resample_pix_fmt;
AVRational frame_rate;
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
float frame_aspect_ratio;
/* forced key frames */
int64_t *forced_kf_pts;
int forced_kf_count;
int forced_kf_index;
/* audio only */
int audio_resample;
ReSampleContext *resample; /* for audio resampling */
int resample_sample_fmt;
int resample_channels;
int resample_sample_rate;
int reformat_pair;
AVAudioConvert *reformat_ctx;
AVFifoBuffer *fifo; /* for compression: one audio fifo per codec */
FILE *logfile;
#if CONFIG_AVFILTER
AVFilterContext *output_video_filter;
AVFilterContext *input_video_filter;
AVFilterBufferRef *picref;
char *avfilter;
AVFilterGraph *graph;
#endif
int64_t sws_flags;
AVDictionary *opts;
int is_past_recording_time;
int stream_copy;
const char *attachment_filename;
int copy_initial_nonkeyframes;
} OutputStream;
Anton Khirnov
committed
typedef struct OutputFile {
AVFormatContext *ctx;
AVDictionary *opts;
int ost_index; /* index of the first stream in output_streams */
int64_t recording_time; /* desired length of the resulting file in microseconds */
int64_t start_time; /* start time in microseconds */
uint64_t limit_filesize;
Anton Khirnov
committed
} OutputFile;
static InputStream *input_streams = NULL;
static int nb_input_streams = 0;
static InputFile *input_files = NULL;
static int nb_input_files = 0;
static OutputStream *output_streams = NULL;
static int nb_output_streams = 0;
Anton Khirnov
committed
static OutputFile *output_files = NULL;
static int nb_output_files = 0;
Anton Khirnov
committed
/* input/output options */
int64_t start_time;
Anton Khirnov
committed
SpecifierOpt *codec_names;
int nb_codec_names;
SpecifierOpt *audio_channels;
int nb_audio_channels;
SpecifierOpt *audio_sample_rate;
int nb_audio_sample_rate;
SpecifierOpt *frame_rates;
int nb_frame_rates;
SpecifierOpt *frame_sizes;
int nb_frame_sizes;
SpecifierOpt *frame_pix_fmts;
int nb_frame_pix_fmts;
Anton Khirnov
committed
/* input options */
int64_t input_ts_offset;
Anton Khirnov
committed
SpecifierOpt *ts_scale;
int nb_ts_scale;
SpecifierOpt *dump_attachment;
int nb_dump_attachment;
/* output options */
StreamMap *stream_maps;
int nb_stream_maps;
/* first item specifies output metadata, second is input */
MetadataMap (*meta_data_maps)[2];
int nb_meta_data_maps;
int metadata_global_manual;
int metadata_streams_manual;
int metadata_chapters_manual;
const char **attachments;
int nb_attachments;
Anton Khirnov
committed
int chapters_input_file;
Anton Khirnov
committed
int64_t recording_time;
float mux_preload;
float mux_max_delay;
int video_disable;
int audio_disable;
int subtitle_disable;
int data_disable;
/* indexed by output file stream index */
int *streamid_map;
int nb_streamid_map;
SpecifierOpt *metadata;
int nb_metadata;
SpecifierOpt *max_frames;
int nb_max_frames;
SpecifierOpt *bitstream_filters;
int nb_bitstream_filters;
SpecifierOpt *codec_tags;
int nb_codec_tags;
SpecifierOpt *sample_fmts;
int nb_sample_fmts;
SpecifierOpt *qscale;
int nb_qscale;
SpecifierOpt *forced_key_frames;
int nb_forced_key_frames;
SpecifierOpt *force_fps;
int nb_force_fps;
SpecifierOpt *frame_aspect_ratios;
int nb_frame_aspect_ratios;
SpecifierOpt *rc_overrides;
int nb_rc_overrides;
SpecifierOpt *intra_matrices;
int nb_intra_matrices;
SpecifierOpt *inter_matrices;
int nb_inter_matrices;
SpecifierOpt *top_field_first;
int nb_top_field_first;
SpecifierOpt *metadata_map;
int nb_metadata_map;
SpecifierOpt *copy_initial_nonkeyframes;
int nb_copy_initial_nonkeyframes;
#if CONFIG_AVFILTER
SpecifierOpt *filters;
int nb_filters;
#endif
#define MATCH_PER_STREAM_OPT(name, type, outvar, fmtctx, st)\
{\
int i, ret;\
for (i = 0; i < o->nb_ ## name; i++) {\
char *spec = o->name[i].specifier;\
if ((ret = check_stream_specifier(fmtctx, st, spec)) > 0)\
outvar = o->name[i].u.type;\
else if (ret < 0)\
exit_program(1);\
}\
}
static void reset_options(OptionsContext *o)
{
const OptionDef *po = options;
/* all OPT_SPEC and OPT_STRING can be freed in generic way */
while (po->name) {
void *dst = (uint8_t*)o + po->u.off;
if (po->flags & OPT_SPEC) {
SpecifierOpt **so = dst;
int i, *count = (int*)(so + 1);
for (i = 0; i < *count; i++) {
av_freep(&(*so)[i].specifier);
if (po->flags & OPT_STRING)
av_freep(&(*so)[i].u.str);
}
av_freep(so);
*count = 0;
} else if (po->flags & OPT_OFFSET && po->flags & OPT_STRING)
av_freep(dst);
po++;
}
av_freep(&o->stream_maps);
av_freep(&o->meta_data_maps);
av_freep(&o->streamid_map);
Anton Khirnov
committed
o->mux_max_delay = 0.7;
Anton Khirnov
committed
o->recording_time = INT64_MAX;
o->limit_filesize = UINT64_MAX;
o->chapters_input_file = INT_MAX;
Anton Khirnov
committed
uninit_opts();
init_opts();
}
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
static int alloc_buffer(InputStream *ist, FrameBuffer **pbuf)
{
AVCodecContext *s = ist->st->codec;
FrameBuffer *buf = av_mallocz(sizeof(*buf));
int ret;
const int pixel_size = av_pix_fmt_descriptors[s->pix_fmt].comp[0].step_minus1+1;
int h_chroma_shift, v_chroma_shift;
int edge = 32; // XXX should be avcodec_get_edge_width(), but that fails on svq1
int w = s->width, h = s->height;
if (!buf)
return AVERROR(ENOMEM);
if (!(s->flags & CODEC_FLAG_EMU_EDGE)) {
w += 2*edge;
h += 2*edge;
}
avcodec_align_dimensions(s, &w, &h);
if ((ret = av_image_alloc(buf->base, buf->linesize, w, h,
s->pix_fmt, 32)) < 0) {
av_freep(&buf);
return ret;
}
/* XXX this shouldn't be needed, but some tests break without this line
* those decoders are buggy and need to be fixed.
* the following tests fail:
* bethsoft-vid, cdgraphics, ansi, aasc, fraps-v1, qtrle-1bit
*/
memset(buf->base[0], 128, ret);
avcodec_get_chroma_sub_sample(s->pix_fmt, &h_chroma_shift, &v_chroma_shift);
for (int i = 0; i < FF_ARRAY_ELEMS(buf->data); i++) {
const int h_shift = i==0 ? 0 : h_chroma_shift;
const int v_shift = i==0 ? 0 : v_chroma_shift;
if (s->flags & CODEC_FLAG_EMU_EDGE)
buf->data[i] = buf->base[i];
else
buf->data[i] = buf->base[i] +
FFALIGN((buf->linesize[i]*edge >> v_shift) +
(pixel_size*edge >> h_shift), 32);
}
buf->w = s->width;
buf->h = s->height;
buf->pix_fmt = s->pix_fmt;
buf->ist = ist;
*pbuf = buf;
return 0;
}
static void free_buffer_pool(InputStream *ist)
{
FrameBuffer *buf = ist->buffer_pool;
while (buf) {
ist->buffer_pool = buf->next;
av_freep(&buf->base[0]);
av_free(buf);
buf = ist->buffer_pool;
}
}
static void unref_buffer(InputStream *ist, FrameBuffer *buf)
{
av_assert0(buf->refcount);
buf->refcount--;
if (!buf->refcount) {
buf->next = ist->buffer_pool;
ist->buffer_pool = buf;
}
}
static int codec_get_buffer(AVCodecContext *s, AVFrame *frame)
{
InputStream *ist = s->opaque;
FrameBuffer *buf;
int ret, i;
if (!ist->buffer_pool && (ret = alloc_buffer(ist, &ist->buffer_pool)) < 0)
return ret;
buf = ist->buffer_pool;
ist->buffer_pool = buf->next;
buf->next = NULL;
if (buf->w != s->width || buf->h != s->height || buf->pix_fmt != s->pix_fmt) {
av_freep(&buf->base[0]);
av_free(buf);
if ((ret = alloc_buffer(ist, &buf)) < 0)
return ret;
}
buf->refcount++;
frame->opaque = buf;
frame->type = FF_BUFFER_TYPE_USER;
frame->extended_data = frame->data;
frame->pkt_pts = s->pkt ? s->pkt->pts : AV_NOPTS_VALUE;
for (i = 0; i < FF_ARRAY_ELEMS(buf->data); i++) {
frame->base[i] = buf->base[i]; // XXX h264.c uses base though it shouldn't
frame->data[i] = buf->data[i];
frame->linesize[i] = buf->linesize[i];
}
return 0;
}
static void codec_release_buffer(AVCodecContext *s, AVFrame *frame)
{
InputStream *ist = s->opaque;
FrameBuffer *buf = frame->opaque;
int i;
for (i = 0; i < FF_ARRAY_ELEMS(frame->data); i++)
frame->data[i] = NULL;
unref_buffer(ist, buf);
}
static void filter_release_buffer(AVFilterBuffer *fb)
{
FrameBuffer *buf = fb->priv;
av_free(fb);
unref_buffer(buf->ist, buf);
}
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
#if CONFIG_AVFILTER
static int configure_video_filters(InputStream *ist, OutputStream *ost)
{
AVFilterContext *last_filter, *filter;
/** filter graph containing all filters including input & output */
AVCodecContext *codec = ost->st->codec;
AVCodecContext *icodec = ist->st->codec;
FFSinkContext ffsink_ctx = { .pix_fmt = codec->pix_fmt };
AVRational sample_aspect_ratio;
char args[255];
int ret;
ost->graph = avfilter_graph_alloc();
if (ist->st->sample_aspect_ratio.num){
sample_aspect_ratio = ist->st->sample_aspect_ratio;
}else
sample_aspect_ratio = ist->st->codec->sample_aspect_ratio;
snprintf(args, 255, "%d:%d:%d:%d:%d:%d:%d", ist->st->codec->width,
ist->st->codec->height, ist->st->codec->pix_fmt, 1, AV_TIME_BASE,
sample_aspect_ratio.num, sample_aspect_ratio.den);
ret = avfilter_graph_create_filter(&ost->input_video_filter, avfilter_get_by_name("buffer"),
"src", args, NULL, ost->graph);
if (ret < 0)
return ret;
ret = avfilter_graph_create_filter(&ost->output_video_filter, &ffsink,
"out", NULL, &ffsink_ctx, ost->graph);
if (ret < 0)
return ret;
last_filter = ost->input_video_filter;
if (codec->width != icodec->width || codec->height != icodec->height) {
snprintf(args, 255, "%d:%d:flags=0x%X",
codec->width,
codec->height,
if ((ret = avfilter_graph_create_filter(&filter, avfilter_get_by_name("scale"),
NULL, args, NULL, ost->graph)) < 0)
return ret;
if ((ret = avfilter_link(last_filter, 0, filter, 0)) < 0)
return ret;
last_filter = filter;
}
snprintf(args, sizeof(args), "flags=0x%X", (unsigned)ost->sws_flags);
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
ost->graph->scale_sws_opts = av_strdup(args);
if (ost->avfilter) {
AVFilterInOut *outputs = av_malloc(sizeof(AVFilterInOut));
AVFilterInOut *inputs = av_malloc(sizeof(AVFilterInOut));
outputs->name = av_strdup("in");
outputs->filter_ctx = last_filter;
outputs->pad_idx = 0;
outputs->next = NULL;
inputs->name = av_strdup("out");
inputs->filter_ctx = ost->output_video_filter;
inputs->pad_idx = 0;
inputs->next = NULL;
if ((ret = avfilter_graph_parse(ost->graph, ost->avfilter, inputs, outputs, NULL)) < 0)
return ret;
av_freep(&ost->avfilter);
} else {
if ((ret = avfilter_link(last_filter, 0, ost->output_video_filter, 0)) < 0)
return ret;
}
if ((ret = avfilter_graph_config(ost->graph, NULL)) < 0)
return ret;
codec->width = ost->output_video_filter->inputs[0]->w;
codec->height = ost->output_video_filter->inputs[0]->h;
codec->sample_aspect_ratio = ost->st->sample_aspect_ratio =
ost->frame_aspect_ratio ? // overridden by the -aspect cli option
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
av_d2q(ost->frame_aspect_ratio*codec->height/codec->width, 255) :
ost->output_video_filter->inputs[0]->sample_aspect_ratio;
return 0;
}
#endif /* CONFIG_AVFILTER */
static void term_exit(void)
{
av_log(NULL, AV_LOG_QUIET, "");
}
static volatile int received_sigterm = 0;
static volatile int received_nb_signals = 0;
static void
sigterm_handler(int sig)
{
received_sigterm = sig;
received_nb_signals++;
term_exit();
}
static void term_init(void)
{
signal(SIGINT , sigterm_handler); /* Interrupt (ANSI). */
signal(SIGTERM, sigterm_handler); /* Termination (ANSI). */
#ifdef SIGXCPU
signal(SIGXCPU, sigterm_handler);
#endif
}
Martin Storsjö
committed
static int decode_interrupt_cb(void *ctx)
{
return received_nb_signals > 1;
}
Martin Storsjö
committed
static const AVIOInterruptCB int_cb = { decode_interrupt_cb, NULL };
void exit_program(int ret)
{
int i;
/* close files */
for(i=0;i<nb_output_files;i++) {
Anton Khirnov
committed
AVFormatContext *s = output_files[i].ctx;
if (!(s->oformat->flags & AVFMT_NOFILE) && s->pb)
avio_close(s->pb);
avformat_free_context(s);
Anton Khirnov
committed
av_dict_free(&output_files[i].opts);
}
for(i=0;i<nb_input_files;i++) {
avformat_close_input(&input_files[i].ctx);
for (i = 0; i < nb_input_streams; i++) {
av_freep(&input_streams[i].decoded_frame);
av_freep(&input_streams[i].filtered_frame);
av_dict_free(&input_streams[i].opts);
free_buffer_pool(&input_streams[i]);
if (vstats_file)
fclose(vstats_file);
av_free(vstats_filename);
av_freep(&input_streams);
av_freep(&input_files);
av_freep(&output_streams);
Anton Khirnov
committed
av_freep(&output_files);
uninit_opts();
av_free(audio_buf);
av_free(audio_out);
allocated_audio_buf_size= allocated_audio_out_size= 0;
#if CONFIG_AVFILTER
avfilter_uninit();
#endif
avformat_network_deinit();
if (received_sigterm) {
av_log(NULL, AV_LOG_INFO, "Received signal %d: terminating.\n",
(int) received_sigterm);
}
static void assert_avoptions(AVDictionary *m)
{
AVDictionaryEntry *t;
if ((t = av_dict_get(m, "", NULL, AV_DICT_IGNORE_SUFFIX))) {
av_log(NULL, AV_LOG_FATAL, "Option %s not found.\n", t->key);
exit_program(1);
}
}
static void assert_codec_experimental(AVCodecContext *c, int encoder)
{
const char *codec_string = encoder ? "encoder" : "decoder";
AVCodec *codec;
if (c->codec->capabilities & CODEC_CAP_EXPERIMENTAL &&
c->strict_std_compliance > FF_COMPLIANCE_EXPERIMENTAL) {
av_log(NULL, AV_LOG_FATAL, "%s '%s' is experimental and might produce bad "
"results.\nAdd '-strict experimental' if you want to use it.\n",
codec_string, c->codec->name);
codec = encoder ? avcodec_find_encoder(c->codec->id) : avcodec_find_decoder(c->codec->id);
if (!(codec->capabilities & CODEC_CAP_EXPERIMENTAL))
av_log(NULL, AV_LOG_FATAL, "Or use the non experimental %s '%s'.\n",
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
codec_string, codec->name);
exit_program(1);
}
}
static void choose_sample_fmt(AVStream *st, AVCodec *codec)
{
if(codec && codec->sample_fmts){
const enum AVSampleFormat *p= codec->sample_fmts;
for(; *p!=-1; p++){
if(*p == st->codec->sample_fmt)
break;
}
if (*p == -1) {
av_log(NULL, AV_LOG_WARNING,
"Incompatible sample format '%s' for codec '%s', auto-selecting format '%s'\n",
av_get_sample_fmt_name(st->codec->sample_fmt),
codec->name,
av_get_sample_fmt_name(codec->sample_fmts[0]));
st->codec->sample_fmt = codec->sample_fmts[0];
}
}
}
/**
* Update the requested input sample format based on the output sample format.
* This is currently only used to request float output from decoders which
* support multiple sample formats, one of which is AV_SAMPLE_FMT_FLT.
* Ideally this will be removed in the future when decoders do not do format
* conversion and only output in their native format.
*/
static void update_sample_fmt(AVCodecContext *dec, AVCodec *dec_codec,
AVCodecContext *enc)
{
/* if sample formats match or a decoder sample format has already been
requested, just return */
if (enc->sample_fmt == dec->sample_fmt ||
dec->request_sample_fmt > AV_SAMPLE_FMT_NONE)
return;
/* if decoder supports more than one output format */
if (dec_codec && dec_codec->sample_fmts &&
dec_codec->sample_fmts[0] != AV_SAMPLE_FMT_NONE &&
dec_codec->sample_fmts[1] != AV_SAMPLE_FMT_NONE) {
const enum AVSampleFormat *p;
int min_dec = -1, min_inc = -1;
/* find a matching sample format in the encoder */
for (p = dec_codec->sample_fmts; *p != AV_SAMPLE_FMT_NONE; p++) {
if (*p == enc->sample_fmt) {
dec->request_sample_fmt = *p;
return;
} else if (*p > enc->sample_fmt) {
min_inc = FFMIN(min_inc, *p - enc->sample_fmt);
} else
min_dec = FFMIN(min_dec, enc->sample_fmt - *p);
}
/* if none match, provide the one that matches quality closest */
dec->request_sample_fmt = min_inc > 0 ? enc->sample_fmt + min_inc :
enc->sample_fmt - min_dec;
}
}
static void choose_sample_rate(AVStream *st, AVCodec *codec)
{
if(codec && codec->supported_samplerates){
const int *p= codec->supported_samplerates;
int best=0;
int best_dist=INT_MAX;
for(; *p; p++){
int dist= abs(st->codec->sample_rate - *p);
if(dist < best_dist){
best_dist= dist;
best= *p;
}
}
if(best_dist){
av_log(st->codec, AV_LOG_WARNING, "Requested sampling rate unsupported using closest supported (%d)\n", best);
}
st->codec->sample_rate= best;
}
}
static void choose_pixel_fmt(AVStream *st, AVCodec *codec)
{
if(codec && codec->pix_fmts){
const enum PixelFormat *p= codec->pix_fmts;
if(st->codec->strict_std_compliance <= FF_COMPLIANCE_UNOFFICIAL){
if(st->codec->codec_id==CODEC_ID_MJPEG){
p= (const enum PixelFormat[]){PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUV420P, PIX_FMT_YUV422P, PIX_FMT_NONE};
}else if(st->codec->codec_id==CODEC_ID_LJPEG){
p= (const enum PixelFormat[]){PIX_FMT_YUVJ420P, PIX_FMT_YUVJ422P, PIX_FMT_YUVJ444P, PIX_FMT_YUV420P, PIX_FMT_YUV422P, PIX_FMT_YUV444P, PIX_FMT_BGRA, PIX_FMT_NONE};
}
}
for (; *p != PIX_FMT_NONE; p++) {
if(*p == st->codec->pix_fmt)
break;
}
if(st->codec->pix_fmt != PIX_FMT_NONE)
av_log(NULL, AV_LOG_WARNING,
"Incompatible pixel format '%s' for codec '%s', auto-selecting format '%s'\n",
av_pix_fmt_descriptors[st->codec->pix_fmt].name,
codec->name,
av_pix_fmt_descriptors[codec->pix_fmts[0]].name);
st->codec->pix_fmt = codec->pix_fmts[0];
}
}
}
static double
get_sync_ipts(const OutputStream *ost)
{
const InputStream *ist = ost->sync_ist;
OutputFile *of = &output_files[ost->file_index];
return (double)(ist->pts - of->start_time)/AV_TIME_BASE;
}
static void write_frame(AVFormatContext *s, AVPacket *pkt, AVCodecContext *avctx, AVBitStreamFilterContext *bsfc){
int ret;
while(bsfc){
AVPacket new_pkt= *pkt;
int a= av_bitstream_filter_filter(bsfc, avctx, NULL,
&new_pkt.data, &new_pkt.size,
pkt->data, pkt->size,
pkt->flags & AV_PKT_FLAG_KEY);
if(a>0){
av_free_packet(pkt);
new_pkt.destruct= av_destruct_packet;
} else if(a<0){
av_log(NULL, AV_LOG_ERROR, "%s failed for stream %d, codec %s",
bsfc->filter->name, pkt->stream_index,
avctx->codec ? avctx->codec->name : "copy");
print_error("", a);
if (exit_on_error)
exit_program(1);
}
*pkt= new_pkt;
bsfc= bsfc->next;
}
ret= av_interleaved_write_frame(s, pkt);
if(ret < 0){
print_error("av_interleaved_write_frame()", ret);
exit_program(1);
}
}
static void generate_silence(uint8_t* buf, enum AVSampleFormat sample_fmt, size_t size)
{
int fill_char = 0x00;
if (sample_fmt == AV_SAMPLE_FMT_U8)
fill_char = 0x80;
memset(buf, fill_char, size);
}
static void do_audio_out(AVFormatContext *s, OutputStream *ost,
InputStream *ist, AVFrame *decoded_frame)
{
uint8_t *buftmp;
int64_t audio_out_size, audio_buf_size;
int size_out, frame_bytes, ret, resample_changed;
AVCodecContext *enc= ost->st->codec;
AVCodecContext *dec= ist->st->codec;
int osize = av_get_bytes_per_sample(enc->sample_fmt);
int isize = av_get_bytes_per_sample(dec->sample_fmt);
const int coded_bps = av_get_bits_per_sample(enc->codec->id);
uint8_t *buf = decoded_frame->data[0];
int size = decoded_frame->nb_samples * dec->channels * isize;
int64_t allocated_for_size = size;
need_realloc:
audio_buf_size= (allocated_for_size + isize*dec->channels - 1) / (isize*dec->channels);
audio_buf_size= (audio_buf_size*enc->sample_rate + dec->sample_rate) / dec->sample_rate;
audio_buf_size= audio_buf_size*2 + 10000; //safety factors for the deprecated resampling API
audio_buf_size= FFMAX(audio_buf_size, enc->frame_size);
audio_buf_size*= osize*enc->channels;
audio_out_size= FFMAX(audio_buf_size, enc->frame_size * osize * enc->channels);
if(coded_bps > 8*osize)
audio_out_size= audio_out_size * coded_bps / (8*osize);
audio_out_size += FF_MIN_BUFFER_SIZE;
if(audio_out_size > INT_MAX || audio_buf_size > INT_MAX){
av_log(NULL, AV_LOG_FATAL, "Buffer sizes too large\n");
exit_program(1);
}
av_fast_malloc(&audio_buf, &allocated_audio_buf_size, audio_buf_size);
av_fast_malloc(&audio_out, &allocated_audio_out_size, audio_out_size);
if (!audio_buf || !audio_out){
av_log(NULL, AV_LOG_FATAL, "Out of memory in do_audio_out\n");
exit_program(1);
}
if (enc->channels != dec->channels || enc->sample_rate != dec->sample_rate)
ost->audio_resample = 1;
resample_changed = ost->resample_sample_fmt != dec->sample_fmt ||
ost->resample_channels != dec->channels ||
ost->resample_sample_rate != dec->sample_rate;
if ((ost->audio_resample && !ost->resample) || resample_changed) {
if (resample_changed) {
av_log(NULL, AV_LOG_INFO, "Input stream #%d:%d frame changed from rate:%d fmt:%s ch:%d to rate:%d fmt:%s ch:%d\n",
ist->file_index, ist->st->index,
ost->resample_sample_rate, av_get_sample_fmt_name(ost->resample_sample_fmt), ost->resample_channels,
dec->sample_rate, av_get_sample_fmt_name(dec->sample_fmt), dec->channels);
ost->resample_sample_fmt = dec->sample_fmt;
ost->resample_channels = dec->channels;
ost->resample_sample_rate = dec->sample_rate;
if (ost->resample)
audio_resample_close(ost->resample);
}
/* if audio_sync_method is >1 the resampler is needed for audio drift compensation */
if (audio_sync_method <= 1 &&
ost->resample_sample_fmt == enc->sample_fmt &&
ost->resample_channels == enc->channels &&
ost->resample_sample_rate == enc->sample_rate) {
ost->resample = NULL;
ost->audio_resample = 0;
} else if (ost->audio_resample) {
if (dec->sample_fmt != AV_SAMPLE_FMT_S16)
av_log(NULL, AV_LOG_WARNING, "Using s16 intermediate sample format for resampling\n");
ost->resample = av_audio_resample_init(enc->channels, dec->channels,
enc->sample_rate, dec->sample_rate,
enc->sample_fmt, dec->sample_fmt,
16, 10, 0, 0.8);
if (!ost->resample) {
av_log(NULL, AV_LOG_FATAL, "Can not resample %d channels @ %d Hz to %d channels @ %d Hz\n",
dec->channels, dec->sample_rate,
enc->channels, enc->sample_rate);
exit_program(1);
}
}
}
#define MAKE_SFMT_PAIR(a,b) ((a)+AV_SAMPLE_FMT_NB*(b))
if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt &&
MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt)!=ost->reformat_pair) {
if (ost->reformat_ctx)
av_audio_convert_free(ost->reformat_ctx);
ost->reformat_ctx = av_audio_convert_alloc(enc->sample_fmt, 1,
dec->sample_fmt, 1, NULL, 0);
if (!ost->reformat_ctx) {
av_log(NULL, AV_LOG_FATAL, "Cannot convert %s sample format to %s sample format\n",
av_get_sample_fmt_name(dec->sample_fmt),
av_get_sample_fmt_name(enc->sample_fmt));
exit_program(1);
}
ost->reformat_pair=MAKE_SFMT_PAIR(enc->sample_fmt,dec->sample_fmt);
}
if(audio_sync_method){
double delta = get_sync_ipts(ost) * enc->sample_rate - ost->sync_opts
- av_fifo_size(ost->fifo)/(enc->channels * osize);
int idelta = delta * dec->sample_rate / enc->sample_rate;
int byte_delta = idelta * isize * dec->channels;
//FIXME resample delay
if(fabs(delta) > 50){
if(ist->is_start || fabs(delta) > audio_drift_threshold*enc->sample_rate){
if(byte_delta < 0){
byte_delta= FFMAX(byte_delta, -size);
size += byte_delta;
buf -= byte_delta;
av_log(NULL, AV_LOG_VERBOSE, "discarding %d audio samples\n",
-byte_delta / (isize * dec->channels));