avconv.c

/*
 * 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[];

#define MAX_FILES 100

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;

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 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 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 int thread_count= 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 = 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 *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"

struct InputStream;

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;
} 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 */
    PtsCorrectionContext pts_ctx;
    double ts_scale;
    int is_start;            /* is 1 at the start and after a discontinuity */
    int showed_multi_packet_warning;
    int is_past_recording_time;
    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;
} InputFile;

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;
    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 ? // overriden 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];
        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(meta_data_maps);

    av_freep(&input_streams);
    av_freep(&input_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;
    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){
            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);
    }
}

#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;
    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;
}

/* we begin to correct av delay at this threshold */
#define AV_DELAY_MAX 0.100

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++) {