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ffmbc.c
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/*
* ffmpeg main
* Copyright (c) 2000-2003 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU General Public
* License as published by the Free Software Foundation;
* version 2 of the License.
*
* 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
* General Public License for more details.
*
* You should have received a copy of the GNU 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 "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/avcodec.h"
# include "libavfilter/avfilter.h"
# include "libavfilter/avfiltergraph.h"
# include "libavfilter/vsink_buffer.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
#if HAVE_TERMIOS_H
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/time.h>
#include <termios.h>
#elif HAVE_KBHIT
#include <conio.h>
#endif
#include <time.h>
#include "cmdutils.h"
#include "libavutil/avassert.h"
const char program_name[] = "FFmbc";
const int program_birth_year = 2008;
/* select an input stream for an output stream */
typedef struct StreamMap {
int file_index;
int stream_index;
int sync_file_index;
int sync_stream_index;
} StreamMap;
/* select an audio channel to extract */
typedef struct AudioChannelMap {
int file_index;
int stream_index;
int channel_index;
int out_file_index;
int out_stream_index;
int out_channel_index;
} AudioChannelMap;
/**
* 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;
typedef struct ChapterMap {
int in_file;
int out_file;
} ChapterMap;
static const OptionDef options[];
#define MAX_FILES 100
#define MAX_STREAMS 1024 /* arbitrary sanity check value */
static const char *last_asked_format = NULL;
static double *ts_scale;
static int nb_ts_scale;
static AVFormatContext *output_files[MAX_FILES];
static AVDictionary *output_opts[MAX_FILES];
static int nb_output_files = 0;
static StreamMap *stream_maps = NULL;
static int nb_stream_maps;
#define MAX_AUDIO_CHANNEL_MAPS 64
static AudioChannelMap audio_channel_maps[MAX_AUDIO_CHANNEL_MAPS];
static int nb_audio_channel_maps;
/* first item specifies output metadata, second is input */
static MetadataMap (*meta_data_maps)[2] = NULL;
static int nb_meta_data_maps;
static int metadata_streams_autocopy = 0;
static int metadata_chapters_autocopy = 0;
static ChapterMap *chapter_maps = NULL;
static int nb_chapter_maps;
/* 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 AVRational frame_aspect_ratio;
static enum PixelFormat frame_pix_fmt = PIX_FMT_NONE;
static int frame_bits_per_raw_sample = 0;
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 char *video_codec_name = NULL;
static unsigned int video_codec_tag = 0;
static char *video_language = NULL;
static int same_quality = 0;
static int interlaced = 0;
static int intra_dc_precision = 8;
static int loop_input = 0;
static int loop_output = AVFMT_NOOUTPUTLOOP;
static int qp_hist = 0;
#if CONFIG_AVFILTER
static char *vfilters = NULL;
#endif
static int intra_only = 0;
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 char *audio_codec_name = NULL;
static unsigned int audio_codec_tag = 0;
static char *audio_language = NULL;
static int subtitle_disable = 0;
static char *subtitle_codec_name = NULL;
static char *subtitle_language = NULL;
static unsigned int subtitle_codec_tag = 0;
static int data_disable = 0;
static char *data_codec_name = NULL;
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 const char *pass_logfilename_prefix;
static int audio_stream_copy = 0;
static int video_stream_copy = 0;
static int subtitle_stream_copy = 0;
static int data_stream_copy = 0;
static int video_sync_method= 1;
static int audio_sync_method= 1;
static float audio_drift_threshold= 0.1;
static int copy_ts= 0;
static int copy_tb= 0;
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 const char *target;
static int update_interval = 250000;
static int audio_volume = 256;
static int exit_on_error = 0;
static int using_stdin = 0;
static int verbose = 1;
static int run_as_daemon = 0;
static int thread_count= 1;
static int q_pressed = 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 uint64_t limit_filesize = 0;
static int force_fps = 0;
static char *forced_key_frames = NULL;
static float dts_delta_threshold = 10;
static int64_t timer_start;
static uint8_t *input_tmp;
static uint8_t *audio_buf;
static uint8_t *audio_out;
static unsigned int allocated_audio_out_size, allocated_audio_buf_size, input_tmp_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 "ffmpeg2pass"
struct InputStream;
typedef struct {
uint8_t *buf;
unsigned buf_size;
unsigned last_sample_pos; /* last valid sample pos in buf */
unsigned buf_index[MAX_AUDIO_CHANNEL_MAPS];
unsigned sample_size; /* size of one sample */
unsigned out_channels;
} AudioMergeContext;
typedef struct OutputStream {
int file_index; /* file index */
int index; /* stream index in the output file */
int source_index[MAX_AUDIO_CHANNEL_MAPS]; /* AVInputStream index */
int nb_source_indexes; /* number of source indexes */
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 */
int is_start; /* is 1 at the start and after a discontinuity */
int64_t sync_opts; /* output frame counter, could be changed to some true timestamp */ //FIXME look at frame_number
AVFrame prev_frame;
int free_prev_frame;
AVBitStreamFilterContext *bitstream_filters;
AVCodec *enc;
/* video only */
int video_resample;
AVFrame resample_frame; /* temporary frame for image resampling */
struct SwsContext *img_resample_ctx; /* for image resampling */
int resample_height;
int resample_width;
int resample_pix_fmt;
AVRational frame_rate;
int force_fps;
AVRational frame_aspect_ratio;
const char *target;
/* forced key frames */
int64_t *forced_kf_pts;
int forced_kf_count;
int forced_kf_index;
/* audio only */
AudioChannelMap *audio_channel_maps[MAX_AUDIO_CHANNEL_MAPS];
int nb_audio_channel_maps;
uint8_t *audio_merge_buf;
AudioMergeContext audiomerge;
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;
AVFilterBufferRef *prev_picref;
char *avfilter;
AVFilterGraph *graph;
#endif
int sws_flags;
AVDictionary *opts;
} OutputStream;
static OutputStream **output_streams_for_file[MAX_FILES] = { NULL };
static int nb_output_streams_for_file[MAX_FILES] = { 0 };
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 */
double ts_scale;
int64_t dts; /* current dts */
int dts_is_reordered_pts;/* set this to 1 if format is avi/wmv/raw, ie format has no pts */
int showed_multi_packet_warning;
int is_past_recording_time;
AVDictionary *opts;
uint8_t *pkt_data_to_free;
AVRational frame_rate;
} 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;
} InputFile;
#if HAVE_TERMIOS_H
/* init terminal so that we can grab keys */
static struct termios oldtty;
#endif
static InputStream *input_streams = NULL;
static int nb_input_streams = 0;
static InputFile *input_files = NULL;
static int nb_input_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;
enum PixelFormat pix_fmts[] = { codec->pix_fmt, PIX_FMT_NONE };
AVRational sample_aspect_ratio;
AVDictionaryEntry *t;
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 if (ist->st->codec->sample_aspect_ratio.num)
sample_aspect_ratio = ist->st->codec->sample_aspect_ratio;
else
sample_aspect_ratio = (AVRational){1,1};
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, avfilter_get_by_name("buffersink"),
"out", NULL, pix_fmts, ost->graph);
if (ret < 0)
return ret;
last_filter = ost->input_video_filter;
if (t = av_dict_get(ist->st->metadata, "rotate", NULL, 0)) {
AVFilterContext *filter;
snprintf(args, 255, "%d", atoi(t->value));
av_log(NULL, AV_LOG_INFO, "Auto-inserting rotate filter of %d degrees\n",
atoi(t->value));
if ((ret = avfilter_open(&filter, avfilter_get_by_name("rotate"),
"auto-rotate")) < 0)
return ret;
if ((ret = avfilter_init_filter(filter, args, NULL)) < 0)
return ret;
if ((ret = avfilter_link(last_filter, 0, filter, 0)) < 0)
return ret;
last_filter = filter;
avfilter_graph_add_filter(ost->graph, last_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"),
"auto-inserted scaler",
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 = avfilter_inout_alloc();
AVFilterInOut *inputs = avfilter_inout_alloc();
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)) < 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.num ? // overriden by the -aspect cli option
av_mul_q(ost->frame_aspect_ratio, (AVRational){ codec->height, codec->width }) :
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, "%s", "");
#if HAVE_TERMIOS_H
if(!run_as_daemon)
tcsetattr (0, TCSANOW, &oldtty);
#endif
}
static volatile int received_sigterm = 0;
static void
sigterm_handler(int sig)
{
received_sigterm = sig;
q_pressed++;
if (q_pressed > 3)
exit(1);
term_exit();
}
static void term_init(void)
{
#if HAVE_TERMIOS_H
if(!run_as_daemon){
struct termios tty;
tcgetattr (0, &tty);
oldtty = tty;
atexit(term_exit);
tty.c_iflag &= ~(IGNBRK|BRKINT|PARMRK|ISTRIP
|INLCR|IGNCR|ICRNL|IXON);
tty.c_oflag |= OPOST;
tty.c_lflag &= ~(ECHO|ECHONL|ICANON|IEXTEN);
tty.c_cflag &= ~(CSIZE|PARENB);
tty.c_cflag |= CS8;
tty.c_cc[VMIN] = 1;
tty.c_cc[VTIME] = 0;
tcsetattr (0, TCSANOW, &tty);
signal(SIGQUIT, sigterm_handler); /* Quit (POSIX). */
}
#endif
signal(SIGINT , sigterm_handler); /* Interrupt (ANSI). */
signal(SIGTERM, sigterm_handler); /* Termination (ANSI). */
#ifdef SIGXCPU
signal(SIGXCPU, sigterm_handler);
#endif
}
/* read a key without blocking */
static int read_key(void)
{
#if HAVE_TERMIOS_H
int n = 1;
unsigned char ch;
struct timeval tv;
fd_set rfds;
if(run_as_daemon)
return -1;
FD_ZERO(&rfds);
FD_SET(0, &rfds);
tv.tv_sec = 0;
tv.tv_usec = 0;
n = select(1, &rfds, NULL, NULL, &tv);
if (n > 0) {
n = read(0, &ch, 1);
if (n == 1)
return ch;
return n;
}
#elif HAVE_KBHIT
if(kbhit())
return(getch());
#endif
return -1;
}
static int decode_interrupt_cb(void)
{
q_pressed += read_key() == 'q';
return q_pressed > 1;
}
static int ffmpeg_exit(int ret)
{
int i;
/* close files */
for(i=0;i<nb_output_files;i++) {
AVFormatContext *s = output_files[i];
if (!(s->oformat->flags & AVFMT_NOFILE) && s->pb)
avio_close(s->pb);
avformat_free_context(s);
av_free(output_streams_for_file[i]);
av_dict_free(&output_opts[i]);
}
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(streamid_map);
av_free(stream_maps);
av_free(meta_data_maps);
av_freep(&input_streams);
av_freep(&input_files);
av_free(video_codec_name);
av_free(audio_codec_name);
av_free(subtitle_codec_name);
av_free(data_codec_name);
uninit_opts();
av_free(audio_buf);
av_free(audio_out);
av_free(input_tmp);
allocated_audio_buf_size= allocated_audio_out_size= 0;
av_free(samples);
#if CONFIG_AVFILTER
avfilter_uninit();
#endif
if (received_sigterm) {
av_log(NULL, AV_LOG_INFO,
"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);
ffmpeg_exit(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);
ffmpeg_exit(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) {
av_log(NULL, AV_LOG_ERROR, "Array too big.\n");
ffmpeg_exit(1);
}
if (*size < new_size) {
uint8_t *tmp = av_realloc(array, new_size*elem_size);
if (!tmp) {
av_log(NULL, AV_LOG_ERROR, "Could not alloc buffer.\n");
ffmpeg_exit(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) {
if((codec->capabilities & CODEC_CAP_LOSSLESS) && av_get_sample_fmt_name(st->codec->sample_fmt) > av_get_sample_fmt_name(codec->sample_fmts[0]))
av_log(NULL, AV_LOG_ERROR, "Convertion will not be lossless'\n");
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];
}
}
}
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;
enum PixelFormat pix_fmt = codec->pix_fmts[0];
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 == PIX_FMT_YUV422P &&
(st->codec->pix_fmt == PIX_FMT_YUV422P16 ||
st->codec->pix_fmt == PIX_FMT_YUV422P10))
pix_fmt = PIX_FMT_YUV422P;
}
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[pix_fmt].name);
st->codec->pix_fmt = pix_fmt;
}
}
}
static OutputStream *new_output_stream(AVFormatContext *oc, int file_idx, AVCodec *codec)
{
OutputStream *ost;
AVStream *st = av_new_stream(oc, oc->nb_streams < nb_streamid_map ? streamid_map[oc->nb_streams] : 0);
int idx = oc->nb_streams - 1;
if (!st) {
av_log(NULL, AV_LOG_ERROR, "Could not alloc stream.\n");
ffmpeg_exit(1);
}
output_streams_for_file[file_idx] =
grow_array(output_streams_for_file[file_idx],
sizeof(*output_streams_for_file[file_idx]),
&nb_output_streams_for_file[file_idx],
oc->nb_streams);
ost = output_streams_for_file[file_idx][idx] =
av_mallocz(sizeof(OutputStream));
if (!ost) {
av_log(NULL, AV_LOG_ERROR, "Could not alloc output stream\n");
ffmpeg_exit(1);
}
ost->file_index = file_idx;
ost->index = idx;
ost->st = st;
ost->enc = codec;
if (codec)
ost->opts = filter_codec_opts(codec_opts, codec->id, 1);
avcodec_get_context_defaults3(st->codec, codec);
ost->sws_flags = av_get_int(sws_opts, "sws_flags", NULL);
return ost;
}
static double
get_sync_ipts(const OutputStream *ost)
{
const InputStream *ist = ost->sync_ist;
return (double)(ist->pts - 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)
ffmpeg_exit(1);
}
*pkt= new_pkt;
bsfc= bsfc->next;
}
ret= av_interleaved_write_frame(s, pkt);
if(ret < 0){
print_error("av_interleaved_write_frame()", ret);
ffmpeg_exit(1);
}
}
static int audiomerge_init(AudioMergeContext *a, int out_channels, int sample_size)
{
int i;
if (sample_size <= 0 || out_channels <= 0)
return -1;
a->out_channels = out_channels;
a->sample_size = sample_size;
a->buf_size = out_channels*sample_size*48000; // 1 sec at 48khz
a->buf = av_malloc(a->buf_size);
if (!a->buf)
return AVERROR(ENOMEM);
for (i = 0; i < out_channels; i++)
a->buf_index[i] = i*sample_size;
return 0;
}
static int audiomerge_add_channel(AudioMergeContext *a, uint8_t *input,
unsigned in_channel, unsigned out_channel,
unsigned in_channels, unsigned samples)
{
if (out_channel >= a->out_channels)
return -1;
//fprintf(stderr, "in channel %d out channel %d samples %d\n", in_channel, out_channel, samples);
if (a->buf_index[out_channel] +
(int64_t)samples*a->sample_size*a->out_channels >= a->buf_size) {
uint8_t *buf;
if (a->buf_size + (int64_t)samples*a->sample_size*a->out_channels >= UINT_MAX)
goto error;
a->buf_size += (int64_t)samples*a->sample_size*a->out_channels;
buf = av_realloc(a->buf, a->buf_size);
if (!buf) {
error:
av_log(NULL, AV_LOG_ERROR, "error reallocating audiomerge buffer\n");
return -1;
}
a->buf = buf;
}
input += a->sample_size*in_channel;
while (samples--) {
memcpy(a->buf + a->buf_index[out_channel], input, a->sample_size);
a->buf_index[out_channel] += a->sample_size*a->out_channels;
input += a->sample_size*in_channels;
}
a->last_sample_pos = FFMAX(a->buf_index[out_channel], a->last_sample_pos);
return 0;
}
static unsigned audiomerge_complete_size(AudioMergeContext *a)
{
int i, min = INT_MAX;
int align = a->out_channels*a->sample_size;
for (i = 0; i < a->out_channels; i++)
min = FFMIN(a->buf_index[i], min);
return (min / align) * align;
}
static unsigned audiomerge_get_buffered_samples(const OutputStream *ost, const InputStream *ist)
{
int i;
if (!ost->nb_audio_channel_maps)
return 0;
for (i = 0; i < ost->nb_audio_channel_maps; i++) {
if (ost->audio_channel_maps[i]->file_index == ist->file_index &&
ost->audio_channel_maps[i]->stream_index == ist->st->index) {
return ost->audiomerge.buf_index[ost->audio_channel_maps[i]->out_channel_index] /
(ost->audiomerge.sample_size*ost->audiomerge.out_channels);
}
}
av_log(NULL, AV_LOG_ERROR, "error, could not find corresponding channel mapping\n");
return 0;
}
static void audiomerge_drain_complete_size(AudioMergeContext *a)
{
int i;
unsigned complete_size = audiomerge_complete_size(a);
if (!complete_size)
return;
memmove(a->buf, a->buf + complete_size,
a->last_sample_pos - complete_size);
a->last_sample_pos -= complete_size;
for (i = 0; i < a->out_channels; i++)
a->buf_index[i] -= complete_size;
}
#define MAX_AUDIO_PACKET_SIZE (128 * 1024)
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;
int size_out, frame_bytes, ret, resample_changed, i, in_channels;
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);
if (ost->nb_audio_channel_maps > 0)
in_channels = enc->channels; // do not mix channels
else
in_channels = dec->channels;
if (enc->channels != in_channels)
ost->audio_resample = 1;
resample_changed = ost->resample_sample_fmt != dec->sample_fmt ||
(!ost->nb_audio_channel_maps &&
ost->resample_channels != dec->channels) ||
ost->resample_sample_rate != dec->sample_rate;
if ((ost->audio_resample && !ost->resample) || resample_changed) {
if (resample_changed) {
if (ost->nb_audio_channel_maps > 0) {
av_log(NULL, AV_LOG_ERROR, "Input stream audio configuration changed, aborting\n");
ffmpeg_exit(1);
}
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 {
ost->audio_resample = 1;
if (dec->sample_fmt != AV_SAMPLE_FMT_S16 && enc->sample_fmt != AV_SAMPLE_FMT_S16)
av_log(NULL, AV_LOG_ERROR, "Warning, using s16 intermediate sample format for resampling\n");
ost->resample = av_audio_resample_init(enc->channels, in_channels,
enc->sample_rate, dec->sample_rate,
enc->sample_fmt, dec->sample_fmt,
32, 10, 0, 0.97);
if (!ost->resample) {