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avconv.c
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/*
* 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 "libavformat/os_support.h"
#if CONFIG_AVFILTER
# include "libavfilter/avfilter.h"
# include "libavfilter/avfiltergraph.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 const char *last_asked_format = NULL;
static AVDictionary *ts_scale;
static StreamMap *stream_maps = NULL;
static int nb_stream_maps;
static AVDictionary *codec_names;
/* first item specifies output metadata, second is input */
static MetadataMap (*meta_data_maps)[2] = NULL;
static int nb_meta_data_maps;
static int metadata_global_autocopy = 1;
static int metadata_streams_autocopy = 1;
static int metadata_chapters_autocopy = 1;
static int chapters_input_file = INT_MAX;
/* indexed by output file stream index */
static int *streamid_map = NULL;
static int nb_streamid_map = 0;
static int frame_width = 0;
static int frame_height = 0;
static float frame_aspect_ratio = 0;
static enum PixelFormat frame_pix_fmt = PIX_FMT_NONE;
static enum AVSampleFormat audio_sample_fmt = AV_SAMPLE_FMT_NONE;
static int max_frames[4] = {INT_MAX, INT_MAX, INT_MAX, INT_MAX};
static AVRational frame_rate;
static float video_qscale = 0;
static uint16_t *intra_matrix = NULL;
static uint16_t *inter_matrix = NULL;
static const char *video_rc_override_string=NULL;
static int video_disable = 0;
static int video_discard = 0;
static unsigned int video_codec_tag = 0;
static char *video_language = NULL;
static int same_quant = 0;
static int do_deinterlace = 0;
static int top_field_first = -1;
static int me_threshold = 0;
static int intra_dc_precision = 8;
static int qp_hist = 0;
#if CONFIG_AVFILTER
static char *vfilters = NULL;
#endif
static int audio_sample_rate = 0;
#define QSCALE_NONE -99999
static float audio_qscale = QSCALE_NONE;
static int audio_disable = 0;
static int audio_channels = 0;
static unsigned int audio_codec_tag = 0;
static char *audio_language = NULL;
static int subtitle_disable = 0;
static char *subtitle_language = NULL;
static unsigned int subtitle_codec_tag = 0;
static int data_disable = 0;
static unsigned int data_codec_tag = 0;
static float mux_preload= 0.5;
static float mux_max_delay= 0.7;
static int64_t recording_time = INT64_MAX;
static int64_t start_time = 0;
static int64_t input_ts_offset = 0;
static int file_overwrite = 0;
static AVDictionary *metadata;
static int do_benchmark = 0;
static int do_hex_dump = 0;
static int do_pkt_dump = 0;
static int do_psnr = 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 copy_tb;
static int opt_shortest = 0;
static char *vstats_filename;
static FILE *vstats_file;
static int opt_programid = 0;
static int copy_initial_nonkeyframes = 0;
static int rate_emu = 0;
static int audio_volume = 256;
static int exit_on_error = 0;
static int using_stdin = 0;
static int verbose = 1;
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 uint64_t limit_filesize = UINT64_MAX;
static int force_fps = 0;
static char *forced_key_frames = NULL;
static float dts_delta_threshold = 10;
static uint8_t *audio_buf;
static uint8_t *audio_out;
static unsigned int allocated_audio_out_size, allocated_audio_buf_size;
static short *samples;
static AVBitStreamFilterContext *video_bitstream_filters=NULL;
static AVBitStreamFilterContext *audio_bitstream_filters=NULL;
static AVBitStreamFilterContext *subtitle_bitstream_filters=NULL;
#define DEFAULT_PASS_LOGFILENAME_PREFIX "av2pass"
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;
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;
} 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() */
} InputFile;
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;
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
int sws_flags;
AVDictionary *opts;
int is_past_recording_time;
} OutputStream;
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;
} 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;
static OutputFile *output_files = NULL;
static int nb_output_files = 0;
#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,
ost->sws_flags);
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", ost->sws_flags);
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
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
}
static int decode_interrupt_cb(void)
{
return received_nb_signals > 1;
}
static int exit_program(int ret)
{
int i;
/* close files */
for(i=0;i<nb_output_files;i++) {
AVFormatContext *s = output_files[i].ctx;
if (!(s->oformat->flags & AVFMT_NOFILE) && s->pb)
avio_close(s->pb);
avformat_free_context(s);
av_dict_free(&output_files[i].opts);
}
for(i=0;i<nb_input_files;i++) {
av_close_input_file(input_files[i].ctx);
}
for (i = 0; i < nb_input_streams; i++)
av_dict_free(&input_streams[i].opts);
av_free(intra_matrix);
av_free(inter_matrix);
if (vstats_file)
fclose(vstats_file);
av_free(vstats_filename);
av_free(meta_data_maps);
av_freep(&input_streams);
av_freep(&input_files);
av_freep(&output_streams);
av_freep(&output_files);
uninit_opts();
av_free(audio_buf);
av_free(audio_out);
allocated_audio_buf_size= allocated_audio_out_size= 0;
av_free(samples);
#if CONFIG_AVFILTER
avfilter_uninit();
#endif
if (received_sigterm) {
fprintf(stderr,
"Received signal %d: terminating.\n",
(int) received_sigterm);
exit (255);
}
exit(ret); /* not all OS-es handle main() return value */
return ret;
}
static void assert_avoptions(AVDictionary *m)
{
AVDictionaryEntry *t;
if ((t = av_dict_get(m, "", NULL, AV_DICT_IGNORE_SUFFIX))) {
av_log(NULL, AV_LOG_ERROR, "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_ERROR, "%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_ERROR, "Or use the non experimental %s '%s'.\n",
codec_string, codec->name);
exit_program(1);
}
}
/* similar to ff_dynarray_add() and av_fast_realloc() */
static void *grow_array(void *array, int elem_size, int *size, int new_size)
{
if (new_size >= INT_MAX / elem_size) {
fprintf(stderr, "Array too big.\n");
exit_program(1);
}
if (*size < new_size) {
uint8_t *tmp = av_realloc(array, new_size*elem_size);
if (!tmp) {
fprintf(stderr, "Could not alloc buffer.\n");
exit_program(1);
}
memset(tmp + *size*elem_size, 0, (new_size-*size) * elem_size);
*size = new_size;
return tmp;
}
return array;
}
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!=-1; p++){
if(*p == st->codec->pix_fmt)
break;
}
if (*p == -1) {
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){
fprintf(stderr, "%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 do_audio_out(AVFormatContext *s,
OutputStream *ost,
InputStream *ist,
unsigned char *buf, int size)
{
uint8_t *buftmp;
int64_t audio_out_size, audio_buf_size;
int64_t allocated_for_size= 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);
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){
fprintf(stderr, "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){
fprintf(stderr, "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)
fprintf(stderr, "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) {
fprintf(stderr, "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) {
fprintf(stderr, "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 * 2);
double idelta= delta*dec->sample_rate / enc->sample_rate;
int byte_delta= ((int)idelta)*2*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;
if(verbose > 2)
fprintf(stderr, "discarding %d audio samples\n", (int)-delta);
if(!size)
return;
ist->is_start=0;
}else{
static uint8_t *input_tmp= NULL;
input_tmp= av_realloc(input_tmp, byte_delta + size);
if(byte_delta > allocated_for_size - size){
allocated_for_size= byte_delta + (int64_t)size;
goto need_realloc;
}
ist->is_start=0;
memset(input_tmp, 0, byte_delta);
memcpy(input_tmp + byte_delta, buf, size);
buf= input_tmp;
size += byte_delta;
if(verbose > 2)
fprintf(stderr, "adding %d audio samples of silence\n", (int)delta);
}
}else if(audio_sync_method>1){
int comp= av_clip(delta, -audio_sync_method, audio_sync_method);
av_assert0(ost->audio_resample);
if(verbose > 2)
fprintf(stderr, "compensating audio timestamp drift:%f compensation:%d in:%d\n", delta, comp, enc->sample_rate);
// fprintf(stderr, "drift:%f len:%d opts:%"PRId64" ipts:%"PRId64" fifo:%d\n", delta, -1, ost->sync_opts, (int64_t)(get_sync_ipts(ost) * enc->sample_rate), av_fifo_size(ost->fifo)/(ost->st->codec->channels * 2));
av_resample_compensate(*(struct AVResampleContext**)ost->resample, comp, enc->sample_rate);
}
}
}else
ost->sync_opts= lrintf(get_sync_ipts(ost) * enc->sample_rate)
- av_fifo_size(ost->fifo)/(enc->channels * 2); //FIXME wrong
if (ost->audio_resample) {
buftmp = audio_buf;
size_out = audio_resample(ost->resample,
(short *)buftmp, (short *)buf,
size / (dec->channels * isize));
size_out = size_out * enc->channels * osize;
} else {
buftmp = buf;
size_out = size;
}
if (!ost->audio_resample && dec->sample_fmt!=enc->sample_fmt) {
const void *ibuf[6]= {buftmp};
void *obuf[6]= {audio_buf};
int istride[6]= {isize};
int ostride[6]= {osize};
int len= size_out/istride[0];
if (av_audio_convert(ost->reformat_ctx, obuf, ostride, ibuf, istride, len)<0) {
printf("av_audio_convert() failed\n");
if (exit_on_error)
exit_program(1);
return;
}
buftmp = audio_buf;
size_out = len*osize;
}
/* now encode as many frames as possible */
if (enc->frame_size > 1) {
/* output resampled raw samples */
if (av_fifo_realloc2(ost->fifo, av_fifo_size(ost->fifo) + size_out) < 0) {
fprintf(stderr, "av_fifo_realloc2() failed\n");
exit_program(1);
}
av_fifo_generic_write(ost->fifo, buftmp, size_out, NULL);
frame_bytes = enc->frame_size * osize * enc->channels;
while (av_fifo_size(ost->fifo) >= frame_bytes) {
AVPacket pkt;
av_init_packet(&pkt);
av_fifo_generic_read(ost->fifo, audio_buf, frame_bytes, NULL);
//FIXME pass ost->sync_opts as AVFrame.pts in avcodec_encode_audio()
ret = avcodec_encode_audio(enc, audio_out, audio_out_size,
(short *)audio_buf);
if (ret < 0) {
fprintf(stderr, "Audio encoding failed\n");
exit_program(1);
}
audio_size += ret;
pkt.stream_index= ost->index;
pkt.data= audio_out;
pkt.size= ret;
if(enc->coded_frame && enc->coded_frame->pts != AV_NOPTS_VALUE)
pkt.pts= av_rescale_q(enc->coded_frame->pts, enc->time_base, ost->st->time_base);
pkt.flags |= AV_PKT_FLAG_KEY;
write_frame(s, &pkt, enc, ost->bitstream_filters);
ost->sync_opts += enc->frame_size;
}
} else {
AVPacket pkt;
av_init_packet(&pkt);
ost->sync_opts += size_out / (osize * enc->channels);
/* output a pcm frame */
/* determine the size of the coded buffer */
size_out /= osize;
if (coded_bps)
size_out = size_out*coded_bps/8;
if(size_out > audio_out_size){
fprintf(stderr, "Internal error, buffer size too small\n");
exit_program(1);
}
//FIXME pass ost->sync_opts as AVFrame.pts in avcodec_encode_audio()
ret = avcodec_encode_audio(enc, audio_out, size_out,
(short *)buftmp);
if (ret < 0) {
fprintf(stderr, "Audio encoding failed\n");
exit_program(1);
}
audio_size += ret;
pkt.stream_index= ost->index;
pkt.data= audio_out;
pkt.size= ret;
if(enc->coded_frame && enc->coded_frame->pts != AV_NOPTS_VALUE)
pkt.pts= av_rescale_q(enc->coded_frame->pts, enc->time_base, ost->st->time_base);
pkt.flags |= AV_PKT_FLAG_KEY;
write_frame(s, &pkt, enc, ost->bitstream_filters);
}
}
static void pre_process_video_frame(InputStream *ist, AVPicture *picture, void **bufp)
{
AVCodecContext *dec;
AVPicture *picture2;
AVPicture picture_tmp;
uint8_t *buf = 0;
dec = ist->st->codec;
/* deinterlace : must be done before any resize */
if (do_deinterlace) {
int size;
/* create temporary picture */
size = avpicture_get_size(dec->pix_fmt, dec->width, dec->height);
buf = av_malloc(size);
if (!buf)
return;
picture2 = &picture_tmp;
avpicture_fill(picture2, buf, dec->pix_fmt, dec->width, dec->height);
if(avpicture_deinterlace(picture2, picture,
dec->pix_fmt, dec->width, dec->height) < 0) {
/* if error, do not deinterlace */
fprintf(stderr, "Deinterlacing failed\n");
av_free(buf);
buf = NULL;
picture2 = picture;
}
} else {
picture2 = picture;
}
if (picture != picture2)
*picture = *picture2;
*bufp = buf;
}
static void do_subtitle_out(AVFormatContext *s,
OutputStream *ost,
InputStream *ist,
AVSubtitle *sub,
int64_t pts)
{
static uint8_t *subtitle_out = NULL;
int subtitle_out_max_size = 1024 * 1024;
int subtitle_out_size, nb, i;
AVCodecContext *enc;
AVPacket pkt;
if (pts == AV_NOPTS_VALUE) {
fprintf(stderr, "Subtitle packets must have a pts\n");
if (exit_on_error)
exit_program(1);
return;
}
enc = ost->st->codec;
if (!subtitle_out) {
subtitle_out = av_malloc(subtitle_out_max_size);
}
/* Note: DVB subtitle need one packet to draw them and one other
packet to clear them */
/* XXX: signal it in the codec context ? */
if (enc->codec_id == CODEC_ID_DVB_SUBTITLE)
nb = 2;
else
nb = 1;
for(i = 0; i < nb; i++) {
sub->pts = av_rescale_q(pts, ist->st->time_base, AV_TIME_BASE_Q);
// start_display_time is required to be 0
sub->pts += av_rescale_q(sub->start_display_time, (AVRational){1, 1000}, AV_TIME_BASE_Q);