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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 "libavformat/avformat.h"
#include "libavdevice/avdevice.h"
#include "libswscale/swscale.h"
#include "libavutil/opt.h"
#include "libavutil/channel_layout.h"
#include "libavutil/parseutils.h"
#include "libavutil/samplefmt.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"
# include "libavfilter/avfilter.h"
# include "libavfilter/buffersrc.h"
# include "libavfilter/buffersink.h"
#if HAVE_SYS_RESOURCE_H
#include <sys/types.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_PTHREADS
#include <pthread.h>
#endif
#include "cmdutils.h"
#include "libavutil/avassert.h"
const char program_name[] = "avconv";
const int program_birth_year = 2000;
static FILE *vstats_file;
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;
/* signal to input threads that they should exit; set by the main thread */
static int transcoding_finished;
#endif
#define DEFAULT_PASS_LOGFILENAME_PREFIX "av2pass"
InputStream **input_streams = NULL;
int nb_input_streams = 0;
InputFile **input_files = NULL;
int nb_input_files = 0;
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OutputStream **output_streams = NULL;
int nb_output_streams = 0;
OutputFile **output_files = NULL;
int nb_output_files = 0;
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FilterGraph **filtergraphs;
int nb_filtergraphs;
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
}
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static int decode_interrupt_cb(void *ctx)
{
return received_nb_signals > 1;
}
const AVIOInterruptCB int_cb = { decode_interrupt_cb, NULL };
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int i, j;
for (i = 0; i < nb_filtergraphs; i++) {
avfilter_graph_free(&filtergraphs[i]->graph);
for (j = 0; j < filtergraphs[i]->nb_inputs; j++) {
av_freep(&filtergraphs[i]->inputs[j]->name);
av_freep(&filtergraphs[i]->inputs[j]);
for (j = 0; j < filtergraphs[i]->nb_outputs; j++) {
av_freep(&filtergraphs[i]->outputs[j]->name);
av_freep(&filtergraphs[i]->outputs[j]);
av_freep(&filtergraphs[i]->graph_desc);
av_freep(&filtergraphs[i]);
}
av_freep(&filtergraphs);
AVFormatContext *s = output_files[i]->ctx;
if (s && s->oformat && !(s->oformat->flags & AVFMT_NOFILE) && s->pb)
avio_close(s->pb);
avformat_free_context(s);
av_dict_free(&output_files[i]->opts);
av_freep(&output_files[i]);
AVBitStreamFilterContext *bsfc = output_streams[i]->bitstream_filters;
while (bsfc) {
AVBitStreamFilterContext *next = bsfc->next;
av_bitstream_filter_close(bsfc);
bsfc = next;
}
output_streams[i]->bitstream_filters = NULL;
av_frame_free(&output_streams[i]->filtered_frame);
av_parser_close(output_streams[i]->parser);
av_freep(&output_streams[i]->forced_keyframes);
av_freep(&output_streams[i]->avfilter);
av_freep(&output_streams[i]->logfile_prefix);
av_freep(&output_streams[i]);
avformat_close_input(&input_files[i]->ctx);
av_freep(&input_files[i]);
for (i = 0; i < nb_input_streams; i++) {
av_frame_free(&input_streams[i]->decoded_frame);
av_frame_free(&input_streams[i]->filter_frame);
av_dict_free(&input_streams[i]->opts);
av_freep(&input_streams[i]->hwaccel_device);
av_freep(&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);
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av_freep(&output_files);
uninit_opts();
avformat_network_deinit();
if (received_sigterm) {
av_log(NULL, AV_LOG_INFO, "Received signal %d: terminating.\n",
(int) received_sigterm);
exit (255);
}
}
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);
static void abort_codec_experimental(AVCodec *c, int encoder)
{
const char *codec_string = encoder ? "encoder" : "decoder";
AVCodec *codec;
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->name);
codec = encoder ? avcodec_find_encoder(c->id) : avcodec_find_decoder(c->id);
if (!(codec->capabilities & CODEC_CAP_EXPERIMENTAL))
av_log(NULL, AV_LOG_FATAL, "Or use the non experimental %s '%s'.\n",
codec_string, codec->name);
* 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 = INT_MAX, min_inc = INT_MAX;
enum AVSampleFormat dec_fmt = AV_SAMPLE_FMT_NONE;
enum AVSampleFormat inc_fmt = AV_SAMPLE_FMT_NONE;
/* 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 {
enum AVSampleFormat dfmt = av_get_packed_sample_fmt(*p);
enum AVSampleFormat efmt = av_get_packed_sample_fmt(enc->sample_fmt);
int fmt_diff = 32 * abs(dfmt - efmt);
if (av_sample_fmt_is_planar(*p) !=
av_sample_fmt_is_planar(enc->sample_fmt))
fmt_diff++;
if (dfmt == efmt) {
min_inc = fmt_diff;
inc_fmt = *p;
} else if (dfmt > efmt) {
if (fmt_diff < min_inc) {
min_inc = fmt_diff;
inc_fmt = *p;
}
} else {
if (fmt_diff < min_dec) {
min_dec = fmt_diff;
dec_fmt = *p;
}
}
}
}
/* if none match, provide the one that matches quality closest */
dec->request_sample_fmt = min_inc != INT_MAX ? inc_fmt : dec_fmt;
static void write_frame(AVFormatContext *s, AVPacket *pkt, OutputStream *ost)
AVBitStreamFilterContext *bsfc = ost->bitstream_filters;
AVCodecContext *avctx = ost->st->codec;
/*
* Audio encoders may split the packets -- #frames in != #packets out.
* But there is no reordering, so we can limit the number of output packets
* by simply dropping them here.
* Counting encoded video frames needs to be done separately because of
* reordering, see do_video_out()
*/
if (!(avctx->codec_type == AVMEDIA_TYPE_VIDEO && avctx->codec)) {
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if (ost->frame_number >= ost->max_frames) {
av_free_packet(pkt);
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}
ost->frame_number++;
}
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) {
new_pkt.buf = av_buffer_create(new_pkt.data, new_pkt.size,
av_buffer_default_free, NULL, 0);
if (!new_pkt.buf)
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)
if (!(s->oformat->flags & AVFMT_NOTIMESTAMPS) &&
ost->last_mux_dts != AV_NOPTS_VALUE &&
pkt->dts < ost->last_mux_dts + !(s->oformat->flags & AVFMT_TS_NONSTRICT)) {
av_log(NULL, AV_LOG_WARNING, "Non-monotonous DTS in output stream "
"%d:%d; previous: %"PRId64", current: %"PRId64"; ",
ost->file_index, ost->st->index, ost->last_mux_dts, pkt->dts);
if (exit_on_error) {
av_log(NULL, AV_LOG_FATAL, "aborting.\n");
}
av_log(NULL, AV_LOG_WARNING, "changing to %"PRId64". This may result "
"in incorrect timestamps in the output file.\n",
ost->last_mux_dts + 1);
pkt->dts = ost->last_mux_dts + 1;
if (pkt->pts != AV_NOPTS_VALUE)
pkt->pts = FFMAX(pkt->pts, pkt->dts);
}
ost->last_mux_dts = pkt->dts;
pkt->stream_index = ost->index;
ret = av_interleaved_write_frame(s, pkt);
if (ret < 0) {
print_error("av_interleaved_write_frame()", ret);
static int check_recording_time(OutputStream *ost)
{
OutputFile *of = output_files[ost->file_index];
if (of->recording_time != INT64_MAX &&
av_compare_ts(ost->sync_opts - ost->first_pts, ost->st->codec->time_base, of->recording_time,
AV_TIME_BASE_Q) >= 0) {
ost->finished = 1;
return 0;
}
return 1;
}
static void do_audio_out(AVFormatContext *s, OutputStream *ost,
AVFrame *frame)
{
AVCodecContext *enc = ost->st->codec;
AVPacket pkt;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (frame->pts == AV_NOPTS_VALUE || audio_sync_method < 0)
frame->pts = ost->sync_opts;
ost->sync_opts = frame->pts + frame->nb_samples;
if (avcodec_encode_audio2(enc, &pkt, frame, &got_packet) < 0) {
av_log(NULL, AV_LOG_FATAL, "Audio encoding failed\n");
}
if (got_packet) {
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base);
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base);
if (pkt.duration > 0)
pkt.duration = av_rescale_q(pkt.duration, enc->time_base, ost->st->time_base);
write_frame(s, &pkt, ost);
audio_size += pkt.size;
}
}
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) {
av_log(NULL, AV_LOG_ERROR, "Subtitle packets must have a pts\n");
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 == AV_CODEC_ID_DVB_SUBTITLE)
nb = 2;
else
nb = 1;
ost->sync_opts = av_rescale_q(pts, ist->st->time_base, enc->time_base);
if (!check_recording_time(ost))
return;
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);
sub->end_display_time -= sub->start_display_time;
sub->start_display_time = 0;
subtitle_out_size = avcodec_encode_subtitle(enc, subtitle_out,
subtitle_out_max_size, sub);
if (subtitle_out_size < 0) {
av_log(NULL, AV_LOG_FATAL, "Subtitle encoding failed\n");
}
av_init_packet(&pkt);
pkt.data = subtitle_out;
pkt.size = subtitle_out_size;
pkt.pts = av_rescale_q(sub->pts, AV_TIME_BASE_Q, ost->st->time_base);
if (enc->codec_id == AV_CODEC_ID_DVB_SUBTITLE) {
/* XXX: the pts correction is handled here. Maybe handling
it in the codec would be better */
if (i == 0)
pkt.pts += 90 * sub->start_display_time;
else
pkt.pts += 90 * sub->end_display_time;
}
write_frame(s, &pkt, ost);
static void do_video_out(AVFormatContext *s,
OutputStream *ost,
AVFrame *in_picture,
int ret, format_video_sync;
AVPacket pkt;
AVCodecContext *enc = ost->st->codec;
format_video_sync = video_sync_method;
if (format_video_sync == VSYNC_AUTO)
format_video_sync = (s->oformat->flags & AVFMT_NOTIMESTAMPS) ? VSYNC_PASSTHROUGH :
(s->oformat->flags & AVFMT_VARIABLE_FPS) ? VSYNC_VFR : VSYNC_CFR;
if (format_video_sync != VSYNC_PASSTHROUGH &&
ost->frame_number &&
in_picture->pts != AV_NOPTS_VALUE &&
in_picture->pts < ost->sync_opts) {
nb_frames_drop++;
av_log(NULL, AV_LOG_VERBOSE, "*** drop!\n");
if (in_picture->pts == AV_NOPTS_VALUE)
in_picture->pts = ost->sync_opts;
ost->sync_opts = in_picture->pts;
ost->first_pts = in_picture->pts;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
if (ost->frame_number >= ost->max_frames)
enc->codec->id == AV_CODEC_ID_RAWVIDEO) {
/* raw pictures are written as AVPicture structure to
avoid any copies. We support temporarily the older
method. */
enc->coded_frame->interlaced_frame = in_picture->interlaced_frame;
enc->coded_frame->top_field_first = in_picture->top_field_first;
pkt.data = (uint8_t *)in_picture;
pkt.size = sizeof(AVPicture);
pkt.pts = av_rescale_q(in_picture->pts, enc->time_base, ost->st->time_base);
pkt.flags |= AV_PKT_FLAG_KEY;
write_frame(s, &pkt, ost);
} else {
int got_packet;
if (ost->st->codec->flags & (CODEC_FLAG_INTERLACED_DCT|CODEC_FLAG_INTERLACED_ME) &&
ost->top_field_first >= 0)
in_picture->top_field_first = !!ost->top_field_first;
in_picture->quality = ost->st->codec->global_quality;
in_picture->pict_type = 0;
in_picture->pts >= ost->forced_kf_pts[ost->forced_kf_index]) {
in_picture->pict_type = AV_PICTURE_TYPE_I;
ret = avcodec_encode_video2(enc, &pkt, in_picture, &got_packet);
if (ret < 0) {
av_log(NULL, AV_LOG_FATAL, "Video encoding failed\n");
if (got_packet) {
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base);
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base);
write_frame(s, &pkt, ost);
*frame_size = pkt.size;
video_size += pkt.size;
/* if two pass, output log */
if (ost->logfile && enc->stats_out) {
fprintf(ost->logfile, "%s", enc->stats_out);
}
ost->sync_opts++;
/*
* For video, number of frames in == number of packets out.
* But there may be reordering, so we can't throw away frames on encoder
* flush, we need to limit them here, before they go into encoder.
*/
ost->frame_number++;
static double psnr(double d)
{
return -10.0 * log(d) / log(10.0);
static void do_video_stats(OutputStream *ost, int frame_size)
{
AVCodecContext *enc;
int frame_number;
double ti1, bitrate, avg_bitrate;
/* this is executed just the first time do_video_stats is called */
if (!vstats_file) {
vstats_file = fopen(vstats_filename, "w");
if (!vstats_file) {
perror("fopen");
}
}
enc = ost->st->codec;
if (enc->codec_type == AVMEDIA_TYPE_VIDEO) {
frame_number = ost->frame_number;
fprintf(vstats_file, "frame= %5d q= %2.1f ", frame_number, enc->coded_frame->quality / (float)FF_QP2LAMBDA);
if (enc->flags&CODEC_FLAG_PSNR)
fprintf(vstats_file, "PSNR= %6.2f ", psnr(enc->coded_frame->error[0] / (enc->width * enc->height * 255.0 * 255.0)));
fprintf(vstats_file,"f_size= %6d ", frame_size);
/* compute pts value */
ti1 = ost->sync_opts * av_q2d(enc->time_base);
if (ti1 < 0.01)
ti1 = 0.01;
bitrate = (frame_size * 8) / av_q2d(enc->time_base) / 1000.0;
avg_bitrate = (double)(video_size * 8) / ti1 / 1000.0;
fprintf(vstats_file, "s_size= %8.0fkB time= %0.3f br= %7.1fkbits/s avg_br= %7.1fkbits/s ",
(double)video_size / 1024, ti1, bitrate, avg_bitrate);
fprintf(vstats_file, "type= %c\n", av_get_picture_type_char(enc->coded_frame->pict_type));
}
}
* Read one frame for lavfi output for ost and encode it.
*/
static int poll_filter(OutputStream *ost)
OutputFile *of = output_files[ost->file_index];
if (!ost->filtered_frame && !(ost->filtered_frame = av_frame_alloc())) {
filtered_frame = ost->filtered_frame;
if (ost->enc->type == AVMEDIA_TYPE_AUDIO &&
!(ost->enc->capabilities & CODEC_CAP_VARIABLE_FRAME_SIZE))
ret = av_buffersink_get_samples(ost->filter->filter, filtered_frame,
ost->st->codec->frame_size);
else
ret = av_buffersink_get_frame(ost->filter->filter, filtered_frame);
if (filtered_frame->pts != AV_NOPTS_VALUE) {
int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
filtered_frame->pts = av_rescale_q(filtered_frame->pts,
ost->filter->filter->inputs[0]->time_base,
ost->st->codec->time_base) -
av_rescale_q(start_time,
AV_TIME_BASE_Q,
ost->st->codec->time_base);
switch (ost->filter->filter->inputs[0]->type) {
case AVMEDIA_TYPE_VIDEO:
if (!ost->frame_aspect_ratio)
ost->st->codec->sample_aspect_ratio = filtered_frame->sample_aspect_ratio;
do_video_out(of->ctx, ost, filtered_frame, &frame_size);
if (vstats_filename && frame_size)
do_video_stats(ost, frame_size);
break;
case AVMEDIA_TYPE_AUDIO:
do_audio_out(of->ctx, ost, filtered_frame);
break;
default:
// TODO support subtitle filters
av_assert0(0);
}
av_frame_unref(filtered_frame);
* Read as many frames from possible from lavfi and encode them.
*
* Always read from the active stream with the lowest timestamp. If no frames
* are available for it then return EAGAIN and wait for more input. This way we
* can use lavfi sources that generate unlimited amount of frames without memory
* usage exploding.
*/
static int poll_filters(void)
{
while (ret >= 0 && !received_sigterm) {
OutputStream *ost = NULL;
int64_t min_pts = INT64_MAX;
/* choose output stream with the lowest timestamp */
for (i = 0; i < nb_output_streams; i++) {
int64_t pts = output_streams[i]->sync_opts;
if (!output_streams[i]->filter || output_streams[i]->finished)
continue;
pts = av_rescale_q(pts, output_streams[i]->st->codec->time_base,
AV_TIME_BASE_Q);
if (pts < min_pts) {
min_pts = pts;
ost = output_streams[i];
}
}
if (!ost)
break;
ret = poll_filter(ost);
if (ret == AVERROR_EOF) {
OutputFile *of = output_files[ost->file_index];
ost->finished = 1;
if (of->shortest) {
for (j = 0; j < of->ctx->nb_streams; j++)
output_streams[of->ost_index + j]->finished = 1;
}
ret = 0;
} else if (ret == AVERROR(EAGAIN))
return 0;
}
return ret;
}
static void print_report(int is_last_report, int64_t timer_start)
{
char buf[1024];
OutputStream *ost;
AVFormatContext *oc;
int64_t total_size;
AVCodecContext *enc;
int frame_number, vid, i;
double bitrate, ti1, pts;
static int64_t last_time = -1;
static int qp_histogram[52];
if (!print_stats && !is_last_report)
return;
if (!is_last_report) {
int64_t cur_time;
/* display the report every 0.5 seconds */
cur_time = av_gettime();
if (last_time == -1) {
last_time = cur_time;
return;
}
if ((cur_time - last_time) < 500000)
return;
last_time = cur_time;
}
oc = output_files[0]->ctx;
total_size = avio_size(oc->pb);
if (total_size <= 0) // FIXME improve avio_size() so it works with non seekable output too
if (total_size < 0) {
char errbuf[128];
av_strerror(total_size, errbuf, sizeof(errbuf));
av_log(NULL, AV_LOG_VERBOSE, "Bitrate not available, "
"avio_tell() failed: %s\n", errbuf);
total_size = 0;
}
buf[0] = '\0';
ti1 = 1e10;
vid = 0;
for (i = 0; i < nb_output_streams; i++) {
ost = output_streams[i];
if (!ost->stream_copy && enc->coded_frame)
q = enc->coded_frame->quality / (float)FF_QP2LAMBDA;
if (vid && enc->codec_type == AVMEDIA_TYPE_VIDEO) {
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "q=%2.1f ", q);
}
if (!vid && enc->codec_type == AVMEDIA_TYPE_VIDEO) {
float t = (av_gettime() - timer_start) / 1000000.0;
frame_number = ost->frame_number;
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "frame=%5d fps=%3d q=%3.1f ",
frame_number, (t > 1) ? (int)(frame_number / t + 0.5) : 0, q);
if (is_last_report)
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "L");
int j;
int qp = lrintf(q);
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "%X", (int)lrintf(log2(qp_histogram[j] + 1)));
double error, error_sum = 0;
double scale, scale_sum = 0;
char type[3] = { 'Y','U','V' };
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "PSNR=");
for (j = 0; j < 3; j++) {
if (is_last_report) {
error = enc->error[j];
scale = enc->width * enc->height * 255.0 * 255.0 * frame_number;
} else {
error = enc->coded_frame->error[j];
scale = enc->width * enc->height * 255.0 * 255.0;
error_sum += error;
scale_sum += scale;
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "%c:%2.2f ", type[j], psnr(error / scale));
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), "*:%2.2f ", psnr(error_sum / scale_sum));
}
vid = 1;
}
/* compute min output value */
pts = (double)ost->st->pts.val * av_q2d(ost->st->time_base);
if ((pts < ti1) && (pts > 0))
ti1 = pts;
}
if (ti1 < 0.01)
ti1 = 0.01;
bitrate = (double)(total_size * 8) / ti1 / 1000.0;
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf),
"size=%8.0fkB time=%0.2f bitrate=%6.1fkbits/s",
(double)total_size / 1024, ti1, bitrate);
if (nb_frames_dup || nb_frames_drop)
snprintf(buf + strlen(buf), sizeof(buf) - strlen(buf), " dup=%d drop=%d",
nb_frames_dup, nb_frames_drop);
av_log(NULL, AV_LOG_INFO, "%s \r", buf);
int64_t raw = audio_size + video_size + extra_size;
float percent = 0.0;
if (raw)
percent = 100.0 * (total_size - raw) / raw;
av_log(NULL, AV_LOG_INFO, "\n");
av_log(NULL, AV_LOG_INFO, "video:%1.0fkB audio:%1.0fkB global headers:%1.0fkB muxing overhead %f%%\n",
video_size / 1024.0,
audio_size / 1024.0,
extra_size / 1024.0,
static void flush_encoders(void)
{
int i, ret;
for (i = 0; i < nb_output_streams; i++) {
OutputStream *ost = output_streams[i];
AVCodecContext *enc = ost->st->codec;
AVFormatContext *os = output_files[ost->file_index]->ctx;
if (!ost->encoding_needed)
continue;
if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO && enc->frame_size <= 1)
if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO && (os->oformat->flags & AVFMT_RAWPICTURE) && enc->codec->id == AV_CODEC_ID_RAWVIDEO)
continue;
int (*encode)(AVCodecContext*, AVPacket*, const AVFrame*, int*) = NULL;
const char *desc;
int64_t *size;
switch (ost->st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
encode = avcodec_encode_audio2;
desc = "Audio";
size = &audio_size;
break;
case AVMEDIA_TYPE_VIDEO:
encode = avcodec_encode_video2;
desc = "Video";
size = &video_size;
break;
default:
stop_encoding = 1;
}
if (encode) {
AVPacket pkt;
int got_packet;
av_init_packet(&pkt);
pkt.data = NULL;
pkt.size = 0;
ret = encode(enc, &pkt, NULL, &got_packet);
if (ret < 0) {
av_log(NULL, AV_LOG_FATAL, "%s encoding failed\n", desc);
if (ost->logfile && enc->stats_out) {
fprintf(ost->logfile, "%s", enc->stats_out);
}
stop_encoding = 1;
break;
}
if (pkt.pts != AV_NOPTS_VALUE)
pkt.pts = av_rescale_q(pkt.pts, enc->time_base, ost->st->time_base);
if (pkt.dts != AV_NOPTS_VALUE)
pkt.dts = av_rescale_q(pkt.dts, enc->time_base, ost->st->time_base);
if (pkt.duration > 0)
pkt.duration = av_rescale_q(pkt.duration, enc->time_base, ost->st->time_base);
}
}
}
/*
* Check whether a packet from ist should be written into ost at this time
*/
static int check_output_constraints(InputStream *ist, OutputStream *ost)
{
OutputFile *of = output_files[ost->file_index];
int ist_index = input_files[ist->file_index]->ist_index + ist->st->index;
if (ost->source_index != ist_index)
return 0;
if (of->start_time != AV_NOPTS_VALUE && ist->last_dts < of->start_time)
return 0;
return 1;
}
static void do_streamcopy(InputStream *ist, OutputStream *ost, const AVPacket *pkt)
{
OutputFile *of = output_files[ost->file_index];
InputFile *f = input_files [ist->file_index];
int64_t start_time = (of->start_time == AV_NOPTS_VALUE) ? 0 : of->start_time;
int64_t ost_tb_start_time = av_rescale_q(start_time, AV_TIME_BASE_Q, ost->st->time_base);
AVPacket opkt;
av_init_packet(&opkt);
if ((!ost->frame_number && !(pkt->flags & AV_PKT_FLAG_KEY)) &&
!ost->copy_initial_nonkeyframes)
return;
if (of->recording_time != INT64_MAX &&
ist->last_dts >= of->recording_time + start_time) {
ost->finished = 1;
if (f->recording_time != INT64_MAX) {
start_time = f->ctx->start_time;
if (f->start_time != AV_NOPTS_VALUE)
start_time += f->start_time;
if (ist->last_dts >= f->recording_time + start_time) {
ost->finished = 1;
return;
}
}
/* force the input stream PTS */
if (ost->st->codec->codec_type == AVMEDIA_TYPE_AUDIO)
audio_size += pkt->size;
else if (ost->st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
video_size += pkt->size;