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Collections - Formulaire de création rapide
19 février 2013, par
Mis à jour : Février 2013
Langue : français
Type : Image
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Personnaliser en ajoutant son logo, sa bannière ou son image de fond
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Publier sur MédiaSpip
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MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
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ffmpeg Transcoding Stops After Few Seconds [migrated]
15 avril 2018, par Salem FI’m trying to do this over week now with no success,
What’s I’m trying to do is transcoding video from live streaming source and downscale it with FFmpeg, but every time I start transcoding it broadcasting for 11 sec and stop.The last command I tried :
ffmpeg -re -i 'http://source.com/1034.ts' -preset ultrafast http://localhost:2052/feed1.ffmI tried to download the .ts file with IDM and it finish downloading the file on the exact 12 Sec that FFmpeg stop trans coding on it.
Does that means that FFmpeg download that file as one segment and not continued reading the source video As what video players does usually. By the way, I tried with source with VLC player and it didn’t stop playing the the same source video.
I decided to pass FFmpeg command via FFserver config file
ffserver.confLaunch ffmpeg -i 'http://source.com/1.ts' -copyinkf -codec copyThe stream works fine for a while but after testing couple sources I notice it’s struggle to trans-coding HD videos.
I guess the issue with my VPS KVM server being very limited CPU and RAM ( 128MB only) ! Since I tried using
ultrafastpreset but din’t solve the issue, another thing, I notice when I enableAVOptionVideo crfsetting onffserver.conftrans-coding runs bit smoothly without frame-rate dropping.
Las my server usesXeon L5520CPU which is outdated CPU specially I gout 1/4 power of V single core (if they count HT it will be 1/8 of the real core) : (# vlc -I dummy 'https://source.com/1034.ts' --sout '#standard{access=http,mux=flv,dst=localhost:2052}'
VLC media player 2.2.8 Weatherwax (revision 2.2.7-14-g3cc1d8cba9)
[09d3fdf0] pulse audio output error: PulseAudio server connection failure: Connection refused
[09d279c0] core interface error: no suitable interface module
[09c9b8f8] core libvlc error: interface "globalhotkeys,none" initialization failed
[09d279c0] dbus interface error: Failed to connect to the D-Bus session daemon: Unable to autolaunch a dbus-daemon without a $DISPLAY for X11
[09d279c0] core interface error: no suitable interface module
[09c9b8f8] core libvlc error: interface "dbus,none" initialization failed
[09d279c0] dummy interface: using the dummy interface module...
[b5e04ae0] access_output_http access out: Consider passing --http-host=IP on the command line instead.
[b5e38ab8] ts demux: MPEG-4 descriptor not found for pid 0x101 type 0xf
[b5e90ae0] packetizer_mpeg4audio decoder: AAC channels: 2 samplerate: 48000
[flv @ 0xb5e33b40] dimensions not set
[b5e06360] avformat mux error: could not write header: Invalid argument
[b5e88ef0] core decoder error: cannot continue streaming due to errors
[b5e90ae0] core decoder error: cannot continue streaming due to errorsHere output with
-loglevel verbose:~# ffmpeg -i http://source.com/1.ts -copyinkf -codec copy -loglevel verbose http://127.0.0.1:8090/feed1.ffm
ffmpeg version 2.6.9 Copyright (c) 2000-2016 the FFmpeg developers
built with gcc 4.9.2 (Debian 4.9.2-10)
configuration: --prefix=/usr --extra-cflags='-g -O2 -fstack-protector-strong -Wformat -Werror=format-security ' --extra-ldflags='-Wl,-z,relro' --cc='ccache cc' --enable-shared --enable-libmp3lame --enable-gpl --enable-nonfree --enable-libvorbis --enable-pthreads --enable-libfaac --enable-libxvid --enable-postproc --enable-x11grab --enable-libgsm --enable-libtheora --enable-libopencore-amrnb --enable-libopencore-amrwb --enable-libx264 --enable-libspeex --enable-nonfree --disable-stripping --enable-libvpx --enable-libschroedinger --disable-encoder=libschroedinger --enable-version3 --enable-libopenjpeg --enable-librtmp --enable-avfilter --enable-libfreetype --enable-libvo-aacenc --disable-decoder=amrnb --enable-libvo-amrwbenc --enable-libaacplus --libdir=/usr/lib/i386-linux-gnu --disable-vda --enable-libbluray --enable-libcdio --enable-gnutls --enable-frei0r --enable-openssl --enable-libass --enable-libopus --enable-fontconfig --enable-libpulse --disable-mips32r2 --disable-mipsdspr1 --disable-mipsdspr2 --enable-libvidstab --enable-libzvbi --enable-avresample --disable-htmlpages --disable-podpages --enable-libutvideo --enable-libfdk-aac --enable-libx265 --enable-libiec61883 --enable-vaapi --enable-libdc1394 --disable-altivec --shlibdir=/usr/lib/i386-linux-gnu
libavutil 54. 20.100 / 54. 20.100
libavcodec 56. 26.100 / 56. 26.100
libavformat 56. 25.101 / 56. 25.101
libavdevice 56. 4.100 / 56. 4.100
libavfilter 5. 11.102 / 5. 11.102
libavresample 2. 1. 0 / 2. 1. 0
libswscale 3. 1.101 / 3. 1.101
libswresample 1. 1.100 / 1. 1.100
libpostproc 53. 3.100 / 53. 3.100
Invalid UE golomb code
Last message repeated 2 times
Input #0, mpegts, from 'http://source.com/1.ts':
Duration: N/A, start: 30472.768167, bitrate: N/A
Program 1
Metadata:
service_name : Service01
service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (High) ([27][0][0][0] / 0x001B), yuv420p, 960x540 (960x544) [SAR 1:1 DAR 16:9], 50 fps, 50 tbr, 90k tbn, 100 tbc
Stream #0:1[0x101]: Audio: aac (LC) ([15][0][0][0] / 0x000F), 48000 Hz, stereo, fltp, 105 kb/s
[graph 0 input from stream 0:1 @ 0x971f2c0] tb:1/48000 samplefmt:fltp samplerate:48000 chlayout:0x3
[audio format for output stream 0:0 @ 0x9844de0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:0'
[auto-inserted resampler 0 @ 0x97115e0] ch:2 chl:stereo fmt:fltp r:48000Hz -> ch:1 chl:mono fmt:fltp r:22050Hz
[graph 1 input from stream 0:0 @ 0x96f5d00] w:960 h:540 pixfmt:yuv420p tb:1/90000 fr:50/1 sar:1/1 sws_param:flags=2
[scaler for output stream 0:1 @ 0x96f5e80] w:352 h:240 flags:'0x4' interl:0
[scaler for output stream 0:1 @ 0x96f5e80] w:960 h:540 fmt:yuv420p sar:1/1 -> w:352 h:240 fmt:yuv420p sar:40/33 flags:0x4
Output #0, ffm, to 'http://127.0.0.1:8090/feed1.ffm':
Metadata:
creation_time : now
encoder : Lavf56.25.101
Stream #0:0: Audio: wmav2, 22050 Hz, mono, fltp, 64 kb/s
Metadata:
encoder : Lavc56.26.100 wmav2
Stream #0:1: Video: msmpeg4v3 (msmpeg4), yuv420p, 352x240 [SAR 40:33 DAR 16:9], q=2-31, 256 kb/s, 50 fps, 1000k tbn, 15 tbc
Metadata:
encoder : Lavc56.26.100 msmpeg4
Stream mapping:
Stream #0:1 -> #0:0 (aac (native) -> wmav2 (native))
Stream #0:0 -> #0:1 (h264 (native) -> msmpeg4v3 (msmpeg4))
Press [q] to stop, [?] for help
Invalid UE golomb code
*** dropping frame 3 from stream 1 at ts 1
Last message repeated 1 times
[msmpeg4 @ 0x970f060] warning, clipping 1 dct coefficients to -127..127
*** dropping frame 4 from stream 1 at ts 2
Last message repeated 1 times
*** dropping frame 5 from stream 1 at ts 3
Last message repeated 1 times
*** dropping frame 5 from stream 1 at ts 4
*** dropping frame 6 from stream 1 at ts 4
Last message repeated 1 times
*** dropping frame 7 from stream 1 at ts 5
Last message repeated 1 times
[msmpeg4 @ 0x970f060] warning, clipping 1 dct coefficients to -127..127
*** dropping frame 8 from stream 1 at ts 6
Last message repeated 1 times
*** dropping frame 8 from stream 1 at ts 7
*** dropping frame 9 from stream 1 at ts 7
Last message repeated 1 times
*** dropping frame 10 from stream 1 at ts 8
Last message repeated 1 times
*** dropping frame 11 from stream 1 at ts 9
Last message repeated 1 times
*** dropping frame 11 from stream 1 at ts 10
*** dropping frame 12 from stream 1 at ts 10
Last message repeated 1 times
*** dropping frame 13 from stream 1 at ts 11
Last message repeated 1 times
*** dropping frame 14 from stream 1 at ts 12
Last message repeated 1 times
*** dropping frame 14 from stream 1 at ts 13
*** dropping frame 15 from stream 1 at ts 13
Last message repeated 1 times
*** dropping frame 16 from stream 1 at ts 14
Last message repeated 1 times
*** dropping frame 17 from stream 1 at ts 15
Last message repeated 1 times
*** dropping frame 17 from stream 1 at ts 16
*** dropping frame 18 from stream 1 at ts 16
Last message repeated 1 times
*** dropping frame 19 from stream 1 at ts 17
Last message repeated 1 times
*** dropping frame 20 from stream 1 at ts 18me=00:00:01.33 bitrate= 270.3kbits/s dup=0 drop=39
Last message repeated 1 times
*** dropping frame 20 from stream 1 at ts 19
*** dropping frame 21 from stream 1 at ts 19
Last message repeated 1 times
*** dropping frame 22 from stream 1 at ts 20
Last message repeated 1 times
*** dropping frame 23 from stream 1 at ts 21
Last message repeated 1 times
*** dropping frame 23 from stream 1 at ts 22
*** dropping frame 24 from stream 1 at ts 22
Last message repeated 1 times
*** dropping frame 25 from stream 1 at ts 23
Last message repeated 1 times
*** dropping frame 26 from stream 1 at ts 24
Last message repeated 1 times
*** dropping frame 26 from stream 1 at ts 25
*** dropping frame 27 from stream 1 at ts 25
Last message repeated 1 times
*** dropping frame 28 from stream 1 at ts 26
Last message repeated 1 times
*** dropping frame 29 from stream 1 at ts 27
Last message repeated 1 times
*** dropping frame 29 from stream 1 at ts 28
*** dropping frame 30 from stream 1 at ts 28
Last message repeated 1 times
*** dropping frame 31 from stream 1 at ts 29
Last message repeated 1 times
*** dropping frame 32 from stream 1 at ts 30
Last message repeated 1 times
*** dropping frame 32 from stream 1 at ts 31
*** dropping frame 33 from stream 1 at ts 31
Last message repeated 1 times
*** dropping frame 34 from stream 1 at ts 32
Last message repeated 1 times
*** dropping frame 34 from stream 1 at ts 33
*** dropping frame 35 from stream 1 at ts 33
*** dropping frame 35 from stream 1 at ts 34
*** dropping frame 36 from stream 1 at ts 34
Last message repeated 1 times
*** dropping frame 37 from stream 1 at ts 35
Last message repeated 1 times
Invalid UE golomb code
*** dropping frame 38 from stream 1 at ts 36
Last message repeated 1 times
*** dropping frame 38 from stream 1 at ts 37
*** dropping frame 39 from stream 1 at ts 37
Last message repeated 1 times
*** dropping frame 40 from stream 1 at ts 38
Last message repeated 1 times
*** dropping frame 41 from stream 1 at ts 39me=00:00:02.73 bitrate= 311.7kbits/s dup=0 drop=88
Last message repeated 1 times
*** dropping frame 41 from stream 1 at ts 40
*** dropping frame 42 from stream 1 at ts 40
Last message repeated 1 times
*** dropping frame 43 from stream 1 at ts 41
Last message repeated 1 times
*** dropping frame 44 from stream 1 at ts 42
Last message repeated 1 times
*** dropping frame 44 from stream 1 at ts 43
*** dropping frame 45 from stream 1 at ts 43
Last message repeated 1 times
*** dropping frame 46 from stream 1 at ts 44
Last message repeated 1 times
*** dropping frame 47 from stream 1 at ts 45
Last message repeated 1 times
*** dropping frame 47 from stream 1 at ts 46
*** dropping frame 48 from stream 1 at ts 46
Last message repeated 1 times
*** dropping frame 49 from stream 1 at ts 47
Last message repeated 1 times
*** dropping frame 50 from stream 1 at ts 48
Last message repeated 1 times
*** dropping frame 50 from stream 1 at ts 49
*** dropping frame 51 from stream 1 at ts 49
Last message repeated 1 times
*** dropping frame 52 from stream 1 at ts 50
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*** dropping frame 53 from stream 1 at ts 51
Last message repeated 1 times
[h264 @ 0x9844a00] error while decoding MB 58 12, bytestream -5
[h264 @ 0x9844a00] concealing 1311 DC, 1311 AC, 1311 MV errors in B frame
*** dropping frame 53 from stream 1 at ts 52
No more output streams to write to, finishing.
frame= 55 fps= 42 q=4.3 Lsize= 152kB time=00:00:03.66 bitrate= 339.6kbits/s dup=0 drop=119
video:116kB audio:26kB subtitle:0kB other streams:0kB global headers:0kB muxing overhead: 6.760316%
Input file #0 (http://source.com/1.ts):
Input stream #0:0 (video): 174 packets read (220322 bytes); 174 frames decoded;
Input stream #0:1 (audio): 156 packets read (36657 bytes); 156 frames decoded (159744 samples);
Total: 330 packets (256979 bytes) demuxed
Output file #0 (http://127.0.0.1:8090/feed1.ffm):
Output stream #0:0 (audio): 72 frames encoded (73383 samples); 72 packets muxed (26712 bytes);
Output stream #0:1 (video): 55 frames encoded; 55 packets muxed (119080 bytes);
Total: 127 packets (145792 bytes) muxedHere input URL file info After I download it to my PC with IDM
General
ID : 1 (0x1)
Complete name : D:\1.ts
Format : MPEG-TS
File size : 256 KiB
Duration : 2 s 520 ms
Overall bit rate mode : Variable
Overall bit rate : 788 kb/s
Video
ID : 256 (0x100)
Menu ID : 1 (0x1)
Format : AVC
Format/Info : Advanced Video Codec
Format profile : High@L3.1
Format settings, CABAC : Yes
Format settings, RefFrames : 2 frames
Codec ID : 27
Duration : 2 s 680 ms
Width : 960 pixels
Height : 540 pixels
Display aspect ratio : 16:9
Frame rate : 50.000 FPS
Color space : YUV
Chroma subsampling : 4:2:0
Bit depth : 8 bits
Scan type : Progressive
Audio
ID : 257 (0x101)
Menu ID : 1 (0x1)
Format : AAC
Format/Info : Advanced Audio Codec
Format version : Version 4
Format profile : LC
Muxing mode : ADTS
Codec ID : 15
Duration : 2 s 69 ms
Bit rate mode : Variable
Channel(s) : 2 channels
Channel positions : Front: L R
Sampling rate : 48.0 kHz
Frame rate : 46.875 FPS (1024 SPF)
Compression mode : Lossy
Delay relative to video : -12 ms
Menu
ID : 4096 (0x1000)
Menu ID : 1 (0x1)
Duration : 2 s 520 ms
List : 256 (0x100) (AVC) / 257 (0x101) (AAC)
Service name : Service01
Service provider : FFmpeg
Service type : digital television -
FFMPEG RTSP Server using muxing doc example
11 novembre 2018, par Harshil MakwanaI am trying to develop RTSP server using FFMPEG. For that I slightly modified muxing file located at
doc/example/folder inside FFMPEG repository.Giving my source code of RTSP server example :
#include
#include
#include
#include
#include <libavutil></libavutil>avassert.h>
#include <libavutil></libavutil>channel_layout.h>
#include <libavutil></libavutil>opt.h>
#include <libavutil></libavutil>mathematics.h>
#include <libavutil></libavutil>timestamp.h>
#include <libavformat></libavformat>avformat.h>
#include <libswscale></libswscale>swscale.h>
#include <libswresample></libswresample>swresample.h>
#define STREAM_DURATION 10.0
#define STREAM_FRAME_RATE 25 /* 25 images/s */
#define STREAM_PIX_FMT AV_PIX_FMT_YUV420P /* default pix_fmt */
#define SCALE_FLAGS SWS_BICUBIC
// a wrapper around a single output AVStream
typedef struct OutputStream {
AVStream *st;
AVCodecContext *enc;
/* pts of the next frame that will be generated */
int64_t next_pts;
int samples_count;
AVFrame *frame;
AVFrame *tmp_frame;
float t, tincr, tincr2;
struct SwsContext *sws_ctx;
struct SwrContext *swr_ctx;
} OutputStream;
static void log_packet(const AVFormatContext *fmt_ctx, const AVPacket *pkt)
{
AVRational *time_base = &fmt_ctx->streams[pkt->stream_index]->time_base;
printf("pts:%s pts_time:%s dts:%s dts_time:%s duration:%s duration_time:%s stream_index:%d\n",
av_ts2str(pkt->pts), av_ts2timestr(pkt->pts, time_base),
av_ts2str(pkt->dts), av_ts2timestr(pkt->dts, time_base),
av_ts2str(pkt->duration), av_ts2timestr(pkt->duration, time_base),
pkt->stream_index);
}
static int write_frame(AVFormatContext *fmt_ctx, const AVRational *time_base, AVStream *st, AVPacket *pkt)
{
/* rescale output packet timestamp values from codec to stream timebase */
av_packet_rescale_ts(pkt, *time_base, st->time_base);
pkt->stream_index = st->index;
/* Write the compressed frame to the media file. */
log_packet(fmt_ctx, pkt);
return av_interleaved_write_frame(fmt_ctx, pkt);
}
/* Add an output stream. */
static void add_stream(OutputStream *ost, AVFormatContext *oc,
AVCodec **codec,
enum AVCodecID codec_id)
{
AVCodecContext *c;
int i;
/* find the encoder */
*codec = avcodec_find_encoder(codec_id);
if (!(*codec)) {
fprintf(stderr, "Could not find encoder for '%s'\n",
avcodec_get_name(codec_id));
exit(1);
}
ost->st = avformat_new_stream(oc, NULL);
if (!ost->st) {
fprintf(stderr, "Could not allocate stream\n");
exit(1);
}
ost->st->id = oc->nb_streams-1;
c = avcodec_alloc_context3(*codec);
if (!c) {
fprintf(stderr, "Could not alloc an encoding context\n");
exit(1);
}
ost->enc = c;
switch ((*codec)->type) {
case AVMEDIA_TYPE_AUDIO:
c->sample_fmt = (*codec)->sample_fmts ?
(*codec)->sample_fmts[0] : AV_SAMPLE_FMT_FLTP;
c->bit_rate = 64000;
c->sample_rate = 44100;
if ((*codec)->supported_samplerates) {
c->sample_rate = (*codec)->supported_samplerates[0];
for (i = 0; (*codec)->supported_samplerates[i]; i++) {
if ((*codec)->supported_samplerates[i] == 44100)
c->sample_rate = 44100;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
c->channel_layout = AV_CH_LAYOUT_STEREO;
if ((*codec)->channel_layouts) {
c->channel_layout = (*codec)->channel_layouts[0];
for (i = 0; (*codec)->channel_layouts[i]; i++) {
if ((*codec)->channel_layouts[i] == AV_CH_LAYOUT_STEREO)
c->channel_layout = AV_CH_LAYOUT_STEREO;
}
}
c->channels = av_get_channel_layout_nb_channels(c->channel_layout);
ost->st->time_base = (AVRational){ 1, c->sample_rate };
break;
case AVMEDIA_TYPE_VIDEO:
c->codec_id = codec_id;
c->bit_rate = 400000;
/* Resolution must be a multiple of two. */
c->width = 352;
c->height = 288;
/* timebase: This is the fundamental unit of time (in seconds) in terms
* of which frame timestamps are represented. For fixed-fps content,
* timebase should be 1/framerate and timestamp increments should be
* identical to 1. */
ost->st->time_base = (AVRational){ 1, STREAM_FRAME_RATE };
c->time_base = ost->st->time_base;
c->gop_size = 12; /* emit one intra frame every twelve frames at most */
c->pix_fmt = STREAM_PIX_FMT;
if (c->codec_id == AV_CODEC_ID_MPEG2VIDEO) {
/* just for testing, we also add B-frames */
c->max_b_frames = 2;
}
if (c->codec_id == AV_CODEC_ID_MPEG1VIDEO) {
/* Needed to avoid using macroblocks in which some coeffs overflow.
* This does not happen with normal video, it just happens here as
* the motion of the chroma plane does not match the luma plane. */
c->mb_decision = 2;
}
break;
default:
break;
}
/* Some formats want stream headers to be separate. */
if (oc->oformat->flags & AVFMT_GLOBALHEADER)
c->flags |= AV_CODEC_FLAG_GLOBAL_HEADER;
}
/**************************************************************/
/* audio output */
static AVFrame *alloc_audio_frame(enum AVSampleFormat sample_fmt,
uint64_t channel_layout,
int sample_rate, int nb_samples)
{
AVFrame *frame = av_frame_alloc();
int ret;
if (!frame) {
fprintf(stderr, "Error allocating an audio frame\n");
exit(1);
}
frame->format = sample_fmt;
frame->channel_layout = channel_layout;
frame->sample_rate = sample_rate;
frame->nb_samples = nb_samples;
if (nb_samples) {
ret = av_frame_get_buffer(frame, 0);
if (ret < 0) {
fprintf(stderr, "Error allocating an audio buffer\n");
exit(1);
}
}
return frame;
}
static void open_audio(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
AVCodecContext *c;
int nb_samples;
int ret;
AVDictionary *opt = NULL;
c = ost->enc;
/* open it */
av_dict_copy(&opt, opt_arg, 0);
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open audio codec: %s\n", av_err2str(ret));
exit(1);
}
/* init signal generator */
ost->t = 0;
ost->tincr = 2 * M_PI * 110.0 / c->sample_rate;
/* increment frequency by 110 Hz per second */
ost->tincr2 = 2 * M_PI * 110.0 / c->sample_rate / c->sample_rate;
if (c->codec->capabilities & AV_CODEC_CAP_VARIABLE_FRAME_SIZE)
nb_samples = 10000;
else
nb_samples = c->frame_size;
ost->frame = alloc_audio_frame(c->sample_fmt, c->channel_layout,
c->sample_rate, nb_samples);
ost->tmp_frame = alloc_audio_frame(AV_SAMPLE_FMT_S16, c->channel_layout,
c->sample_rate, nb_samples);
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
exit(1);
}
/* create resampler context */
ost->swr_ctx = swr_alloc();
if (!ost->swr_ctx) {
fprintf(stderr, "Could not allocate resampler context\n");
exit(1);
}
/* set options */
av_opt_set_int (ost->swr_ctx, "in_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "in_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "in_sample_fmt", AV_SAMPLE_FMT_S16, 0);
av_opt_set_int (ost->swr_ctx, "out_channel_count", c->channels, 0);
av_opt_set_int (ost->swr_ctx, "out_sample_rate", c->sample_rate, 0);
av_opt_set_sample_fmt(ost->swr_ctx, "out_sample_fmt", c->sample_fmt, 0);
/* initialize the resampling context */
if ((ret = swr_init(ost->swr_ctx)) < 0) {
fprintf(stderr, "Failed to initialize the resampling context\n");
exit(1);
}
}
/* Prepare a 16 bit dummy audio frame of 'frame_size' samples and
* 'nb_channels' channels. */
static AVFrame *get_audio_frame(OutputStream *ost)
{
AVFrame *frame = ost->tmp_frame;
int j, i, v;
int16_t *q = (int16_t*)frame->data[0];
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, ost->enc->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
for (j = 0; j nb_samples; j++) {
v = (int)(sin(ost->t) * 10000);
for (i = 0; i < ost->enc->channels; i++)
*q++ = v;
ost->t += ost->tincr;
ost->tincr += ost->tincr2;
}
frame->pts = ost->next_pts;
ost->next_pts += frame->nb_samples;
return frame;
}
/*
* encode one audio frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_audio_frame(AVFormatContext *oc, OutputStream *ost)
{
AVCodecContext *c;
AVPacket pkt = { 0 }; // data and size must be 0;
AVFrame *frame;
int ret;
int got_packet;
int dst_nb_samples;
av_init_packet(&pkt);
c = ost->enc;
frame = get_audio_frame(ost);
if (frame) {
/* convert samples from native format to destination codec format, using the resampler */
/* compute destination number of samples */
dst_nb_samples = av_rescale_rnd(swr_get_delay(ost->swr_ctx, c->sample_rate) + frame->nb_samples,
c->sample_rate, c->sample_rate, AV_ROUND_UP);
av_assert0(dst_nb_samples == frame->nb_samples);
/* when we pass a frame to the encoder, it may keep a reference to it
* internally;
* make sure we do not overwrite it here
*/
ret = av_frame_make_writable(ost->frame);
if (ret < 0)
exit(1);
/* convert to destination format */
ret = swr_convert(ost->swr_ctx,
ost->frame->data, dst_nb_samples,
(const uint8_t **)frame->data, frame->nb_samples);
if (ret < 0) {
fprintf(stderr, "Error while converting\n");
exit(1);
}
frame = ost->frame;
frame->pts = av_rescale_q(ost->samples_count, (AVRational){1, c->sample_rate}, c->time_base);
ost->samples_count += dst_nb_samples;
}
ret = avcodec_encode_audio2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding audio frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
if (ret < 0) {
fprintf(stderr, "Error while writing audio frame: %s\n",
av_err2str(ret));
exit(1);
}
}
return (frame || got_packet) ? 0 : 1;
}
/**************************************************************/
/* video output */
static AVFrame *alloc_picture(enum AVPixelFormat pix_fmt, int width, int height)
{
AVFrame *picture;
int ret;
picture = av_frame_alloc();
if (!picture)
return NULL;
picture->format = pix_fmt;
picture->width = width;
picture->height = height;
/* allocate the buffers for the frame data */
ret = av_frame_get_buffer(picture, 32);
if (ret < 0) {
fprintf(stderr, "Could not allocate frame data.\n");
exit(1);
}
return picture;
}
static void open_video(AVFormatContext *oc, AVCodec *codec, OutputStream *ost, AVDictionary *opt_arg)
{
int ret;
AVCodecContext *c = ost->enc;
AVDictionary *opt = NULL;
av_dict_copy(&opt, opt_arg, 0);
/* open the codec */
ret = avcodec_open2(c, codec, &opt);
av_dict_free(&opt);
if (ret < 0) {
fprintf(stderr, "Could not open video codec: %s\n", av_err2str(ret));
exit(1);
}
/* allocate and init a re-usable frame */
ost->frame = alloc_picture(c->pix_fmt, c->width, c->height);
if (!ost->frame) {
fprintf(stderr, "Could not allocate video frame\n");
exit(1);
}
/* If the output format is not YUV420P, then a temporary YUV420P
* picture is needed too. It is then converted to the required
* output format. */
ost->tmp_frame = NULL;
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
ost->tmp_frame = alloc_picture(AV_PIX_FMT_YUV420P, c->width, c->height);
if (!ost->tmp_frame) {
fprintf(stderr, "Could not allocate temporary picture\n");
exit(1);
}
}
/* copy the stream parameters to the muxer */
ret = avcodec_parameters_from_context(ost->st->codecpar, c);
if (ret < 0) {
fprintf(stderr, "Could not copy the stream parameters\n");
exit(1);
}
}
/* Prepare a dummy image. */
static void fill_yuv_image(AVFrame *pict, int frame_index,
int width, int height)
{
int x, y, i;
i = frame_index;
/* Y */
for (y = 0; y < height; y++)
for (x = 0; x < width; x++)
pict->data[0][y * pict->linesize[0] + x] = x + y + i * 3;
/* Cb and Cr */
for (y = 0; y < height / 2; y++) {
for (x = 0; x < width / 2; x++) {
pict->data[1][y * pict->linesize[1] + x] = 128 + y + i * 2;
pict->data[2][y * pict->linesize[2] + x] = 64 + x + i * 5;
}
}
}
static AVFrame *get_video_frame(OutputStream *ost)
{
AVCodecContext *c = ost->enc;
/* check if we want to generate more frames */
if (av_compare_ts(ost->next_pts, c->time_base,
STREAM_DURATION, (AVRational){ 1, 1 }) >= 0)
return NULL;
/* when we pass a frame to the encoder, it may keep a reference to it
* internally; make sure we do not overwrite it here */
if (av_frame_make_writable(ost->frame) < 0)
exit(1);
if (c->pix_fmt != AV_PIX_FMT_YUV420P) {
/* as we only generate a YUV420P picture, we must convert it
* to the codec pixel format if needed */
if (!ost->sws_ctx) {
ost->sws_ctx = sws_getContext(c->width, c->height,
AV_PIX_FMT_YUV420P,
c->width, c->height,
c->pix_fmt,
SCALE_FLAGS, NULL, NULL, NULL);
if (!ost->sws_ctx) {
fprintf(stderr,
"Could not initialize the conversion context\n");
exit(1);
}
}
fill_yuv_image(ost->tmp_frame, ost->next_pts, c->width, c->height);
sws_scale(ost->sws_ctx,
(const uint8_t * const *)ost->tmp_frame->data, ost->tmp_frame->linesize,
0, c->height, ost->frame->data, ost->frame->linesize);
} else {
fill_yuv_image(ost->frame, ost->next_pts, c->width, c->height);
}
ost->frame->pts = ost->next_pts++;
return ost->frame;
}
/*
* encode one video frame and send it to the muxer
* return 1 when encoding is finished, 0 otherwise
*/
static int write_video_frame(AVFormatContext *oc, OutputStream *ost)
{
int ret;
AVCodecContext *c;
AVFrame *frame;
int got_packet = 0;
AVPacket pkt = { 0 };
c = ost->enc;
frame = get_video_frame(ost);
av_init_packet(&pkt);
/* encode the image */
ret = avcodec_encode_video2(c, &pkt, frame, &got_packet);
if (ret < 0) {
fprintf(stderr, "Error encoding video frame: %s\n", av_err2str(ret));
exit(1);
}
if (got_packet) {
ret = write_frame(oc, &c->time_base, ost->st, &pkt);
} else {
ret = 0;
}
if (ret < 0) {
fprintf(stderr, "Error while writing video frame: %s\n", av_err2str(ret));
exit(1);
}
return (frame || got_packet) ? 0 : 1;
}
static void close_stream(AVFormatContext *oc, OutputStream *ost)
{
avcodec_free_context(&ost->enc);
av_frame_free(&ost->frame);
av_frame_free(&ost->tmp_frame);
sws_freeContext(ost->sws_ctx);
swr_free(&ost->swr_ctx);
}
/**************************************************************/
/* media file output */
int main(int argc, char **argv)
{
OutputStream video_st = { 0 }, audio_st = { 0 };
const char *filename;
AVOutputFormat *fmt;
AVFormatContext *oc;
AVCodec *audio_codec, *video_codec;
int ret;
int have_video = 0, have_audio = 0;
int encode_video = 0, encode_audio = 0;
AVDictionary *opt = NULL;
int i;
/* Initialize libavcodec, and register all codecs and formats. */
av_register_all();
avformat_network_init();
if (argc < 2) {
printf("usage: %s output_file\n"
"API example program to output a media file with libavformat.\n"
"This program generates a synthetic audio and video stream, encodes and\n"
"muxes them into a file named output_file.\n"
"The output format is automatically guessed according to the file extension.\n"
"Raw images can also be output by using '%%d' in the filename.\n"
"\n", argv[0]);
return 1;
}
filename = argv[1];
for (i = 2; i+1 < argc; i+=2) {
if (!strcmp(argv[i], "-flags") || !strcmp(argv[i], "-fflags"))
av_dict_set(&opt, argv[i]+1, argv[i+1], 0);
}
/* allocate the output media context */
avformat_alloc_output_context2(&oc, NULL, "rtsp", filename);
if (!oc) {
printf("Could not deduce output format from file extension: using MPEG.\n");
avformat_alloc_output_context2(&oc, NULL, "mpeg", filename);
}
if (!oc)
return 1;
fmt = oc->oformat;
/* Add the audio and video streams using the default format codecs
* and initialize the codecs. */
if (fmt->video_codec != AV_CODEC_ID_NONE) {
add_stream(&video_st, oc, &video_codec, fmt->video_codec);
have_video = 1;
encode_video = 1;
}
if (fmt->audio_codec != AV_CODEC_ID_NONE) {
add_stream(&audio_st, oc, &audio_codec, fmt->audio_codec);
have_audio = 1;
encode_audio = 1;
}
/* Now that all the parameters are set, we can open the audio and
* video codecs and allocate the necessary encode buffers. */
if (have_video)
open_video(oc, video_codec, &video_st, opt);
if (have_audio)
open_audio(oc, audio_codec, &audio_st, opt);
av_dump_format(oc, 0, filename, 1);
/* open the output file, if needed */
if (!(fmt->flags & AVFMT_NOFILE)) {
ret = avio_open(&oc->pb, filename, AVIO_FLAG_WRITE);
if (ret < 0) {
fprintf(stderr, "Could not open '%s': %s\n", filename,
av_err2str(ret));
return 1;
}
}
/* Write the stream header, if any. */
ret = avformat_write_header(oc, &opt);
if (ret < 0) {
fprintf(stderr, "Error occurred when opening output file: %s\n",
av_err2str(ret));
return 1;
}
while (encode_video || encode_audio) {
/* select the stream to encode */
if (encode_video &&
(!encode_audio || av_compare_ts(video_st.next_pts, video_st.enc->time_base,
audio_st.next_pts, audio_st.enc->time_base) <= 0)) {
encode_video = !write_video_frame(oc, &video_st);
} else {
encode_audio = !write_audio_frame(oc, &audio_st);
}
}
/* Write the trailer, if any. The trailer must be written before you
* close the CodecContexts open when you wrote the header; otherwise
* av_write_trailer() may try to use memory that was freed on
* av_codec_close(). */
av_write_trailer(oc);
/* Close each codec. */
if (have_video)
close_stream(oc, &video_st);
if (have_audio)
close_stream(oc, &audio_st);
if (!(fmt->flags & AVFMT_NOFILE))
/* Close the output file. */
avio_closep(&oc->pb);
/* free the stream */
avformat_free_context(oc);
return 0;
}After compiling it, I am running binary :
$ ./muxing rtsp://127.0.0.1/test
Output #0, rtsp, to 'rtsp://127.0.0.1/test':
Stream #0:0: Video: mpeg4, yuv420p, 352x288, q=2-31, 400 kb/s, 25 tbn
Stream #0:1: Audio: aac (LC), 44100 Hz, stereo, fltp, 64 kb/s
[tcp @ 0x2b9d220] Connection to tcp://127.0.0.1:554?timeout=0 failed: Connection refused
Error occurred when opening output file: Connection refusedBut getting Connection refused error,
-
FFMPEG stereo track stops capturing at random times during a capture session
26 mai 2022, par mrwassenI am currently working on building a workflow to capture and archive a large stash of family and friends PAL and NTSC VHS tapes. The hardware setup is as follows :


- 

- JVC HR-7860S VCR
- s-video / RCA audio >
- ADVC-3000 converter
- SDI / BNC cable >
- Blackmagic Decklink Mini Recorder 4K PCIe card
- installed in a fairly hi-spec windows machine : AMD Ryzen 9 5900X 3.7 Ghz base 12 core, GEFORCE RTX 3060 12 gB, 32 gB ram














The plan is to capture to lossless AVI, then drop into an NLE (Vegas Pro v.16) to do a minimal amount of cleanup / trimming, then render to a more compressed video format (TBD) for upload to AWS S3 accessible through a family website.


The issue I am having is that when I run the capture using ffmpeg/directshow e.g. for a perfectly fine 90 min. PAL tape, at some random point of time during the capture one of the 2 stereo channels just stops capturing. This has happened with all of the tapes I have tested so far, and it happens at different times during the same video. I have examined the frames surrounding points in time when this happens, and it doesn't correlate to any transitions or jitter, but often just randomly in the middle of a perfectly smooth scene. Once the one channel stops capturing it never starts back up again during that capture session.


The ADVC-3000 and the VCR are both showing both stereo channels playing normally throughout the capture. The windows machine running the capture hardly breaks a sweat at any time, and the transfer easily keeps up constantly showing a speed = 1x which I assume means nothing lagging. Also there are no video/audio sync issues at any point in time even towards the end of long tapes e.g. 90 mins.


I am fairly new at ffmpeg, so I have spent extensive amounts of time reading up on forum posts and experimenting and have ended up with the following syntax :


ffmpeg -y -f dshow -rtbufsize 2000M -i video="Blackmagic WDM Capture":audio="Blackmagic WDM Capture" -codec:v v210 -pix_fmt yuv422p -codec:a pcm_s16le -b:a 128k -t 02:00:00 -r 25 -threads 4 -maxrate 2500k -filter:a "volume=1.5" output_v210_audio.avi


The capture runs without a single dropped frame, the only error I am getting when launching (and perhaps this is a smoking gun ?) is :




"Non-monotonous DTS in output stream 0:1 ; previous : 0, current : -30 ;
changing to 1. This may result in incorrect timestamps in the output
file."




I have tried to troubleshoot this in the hopes that it is tied to my issue but so far without luck.


Hoping somebody can help correct or modify my command line or perhaps other ideas to help resolve the issue.


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