[FFmpeg-devel] is it possible that avfilter graph updates its setting in realtime?

[qw]

Hi,


I have one question about some rare usage for avfilter graph:


Sometimes, I want to change the setting in avfilter graph. For example, fps 
filter is used to set output frame rate, and it's expected that fps can be 
changed in accordance to real needs. Sometimes, fps is set to 25, and later 15 
as network bandwidth is not good. Is it possible that avfilter graph updates 
its setting in realtime, such as fps in fps filter, and widthxheight in scale 
filter?


Thanks!


Regards


Andrew
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[FFmpeg-devel] why the option 'timeout' in native rtmp plugin doesn't work

[qw]

Hi,


1)
I want to open and read some rtmp live stream, and don't like wait indefinite 
time for some rtmp stream when there is no rtmp stream to come. Therefore, the 
option of 'timeout' is used to set the longest waiting time. But it didn't 
work, and avformat_open_input() reported as follows:


[rtmp @ 0x1430580] Cannot open connection 
tcp://localhost:1935?listen&listen_timeout=5000


The source code is shown as follows:


AVDictionary *pFormatOpts = NULL;
AVFormatContext *pIfmtCtx;
char pInRtmpUrl[MAX_RTMP_URL_LENGTH];


av_dict_set(&pFormatOpts, "rtmp_live", "-1", 0);
av_dict_set(&pFormatOpts, "rtmp_listen", "1", 0);
av_dict_set(&pFormatOpts, "timeout", "5", 0);


avformat_open_input(&pIfmtCtx, pInRtmpUrl, NULL, &pFormatOpts)


Why the rtmp setting doesn't work?


2)
If I want to set the timeout for av_read_frame()/av_interleaved_write_frame() 
to prevent the function from waiting indefinite time for reading/writing 
avpackets, is there some options for the native rtmp plugin?




Thanks!


Regards


Andrew

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[FFmpeg-devel] why the option 'timeout' in native rtmp plugin doesn't work

[qw]

Hi,


1)
I want to open and read some rtmp live stream, and don't like wait indefinite 
time for some rtmp stream when there is no rtmp stream to come. Therefore, the 
option of 'timeout' is used to set the longest waiting time. But it didn't 
work, and avformat_open_input() reported as follows:


[rtmp @ 0x1430580] Cannot open connection 
tcp://localhost:1935?listen&listen_timeout=5000


The source code is shown as follows:


AVDictionary *pFormatOpts = NULL;
AVFormatContext *pIfmtCtx;
char pInRtmpUrl[MAX_RTMP_URL_LENGTH];


av_dict_set(&pFormatOpts, "rtmp_live", "-1", 0);
av_dict_set(&pFormatOpts, "rtmp_listen", "1", 0);
av_dict_set(&pFormatOpts, "timeout", "5", 0);


avformat_open_input(&pIfmtCtx, pInRtmpUrl, NULL, &pFormatOpts)


Why the rtmp setting doesn't work?


2)
If I want to set the timeout for av_read_frame()/av_interleaved_write_frame() 
to prevent the function from waiting indefinite time for reading/writing 
avpackets, is there some options for the native rtmp plugin?




Thanks!


Regards


Andrew






 
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[FFmpeg-devel] one bug of getting wrong sample rate when receiving rtmp stream

[qw]

Hi,


I found one bug, where ffmpeg functions can't get precise audio sample rate 
when receiving rtmp stream.


1st method:
If avformat_open_input() and avformat_find_stream_info() is used to parse rtmp 
stream, audio sample rate can be calculated correctly.


2nd method:
But if av_format_set_audio_codec() is used just like ffmpeg program does before 
avformat_open_input() and avformat_find_stream_info(), ffmpeg functions can't 
calculate precise sample rate.


For example, if there is one audio rtmp stream, where sample rate is 48000Hz 
and audio codec is aac. If aac codec is preset by using 
av_format_set_audio_codec(), ffmpeg will return wrong sample rate of 44100 Hz. 
The reason is that rtmp stream should be sent to flv demuxer, which only 
support four kinds of sample rate and doesn't support 48000Hz. Therefore, 
ffmpeg functions can't get correct sample rate by using the 2nd method.


Is it a bug? how to fix it?


Thanks


Regards


Andrew





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[FFmpeg-devel] what's the general setting for network reading and writing?

[qw]

Hi,

If input or output is set to file, it's easy to set AVFormatContext. But for 
network application, there are many issues that should be considered, such as 
network delay, no network connection, and corrupted data. How to set 
AVFormatContext to deal with those general network issues?

Thanks!

Regards

Andrew
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[FFmpeg-devel] why ffmpeg 3.1.1 still uses AVStream::codec

[qw]

Hi,

I have read part of source code of ffmpeg 3.1.1. In the structure of 'struct 
AVStream', 'AVCodecContext *codec' is declared as deprecated. But in ffmpeg.c, 
'AVCodecContext *codec' is still used in some functions. Why?

Thanks!

Regards

Andrew


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Re: [FFmpeg-devel] why ffmpeg 3.1.1 still uses AVStream::codec

[qw]

Hi,

If AVStream::codec is declared as deprecated, I think, there should be another 
way to open and close avcodec instead of using avcodec_open2() and 
avcodec_close(). Is that right?



Thanks!

Regards


Andrew



At 2016-07-19 19:30:58, "Timo Rothenpieler"  wrote:
>> Hi,
>> 
>> I have read part of source code of ffmpeg 3.1.1. In the structure of 'struct 
>> AVStream', 'AVCodecContext *codec' is declared as deprecated. But in 
>> ffmpeg.c, 'AVCodecContext *codec' is still used in some functions. Why?
>
>Because nobody felt like changing that yet.
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[FFmpeg-devel] why are audio/video encoding/decoding functions marked as attribute_deprecated?

[qw]

Hi,

avcodec_decode_video2(), avcodec_decode_audio4(), avcodec_encode_video2(), and 
avcodec_encode_audio2(). The following url links are ffmpeg 3.1.1 description:

http://ffmpeg.org/doxygen/3.1/group__lavc__decoding.html#ga3ac51525b7ad8bca4ced9f3446e96532
http://ffmpeg.org/doxygen/3.1/group__lavc__decoding.html#gaaa1fbe477c04455cdc7a994090100db4
http://ffmpeg.org/doxygen/3.1/group__lavc__encoding.html#ga2c08a4729f72f9bdac41b5533c4f2642
http://ffmpeg.org/doxygen/3.1/group__lavc__encoding.html#gae4b5ba4d50b1264a5cf5408ff32fb622

Why are the four functions marked as attribute_deprecated?

The four encoding/decoding functions are still used in ffmpeg.c and 
transcoding.c.

Thanks!

Regards

Andrew


 
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[FFmpeg-devel] one issue for reading incoming rtmp stream

[qw]

Hi,

I'm using avformat_open_input() and avformat_find_stream_info() open incoming 
rtmp stream, where there are two substream, i.e. audio and video streams.

ffmpeg's native rtmp plugin is expected to detect both audio and video streams. 
Sometimes ffmpeg's native rtmp plugin can detect only audio stream, but not 
video stream.

Why? Is it a bug in native rtmp plugin?

Thanks!

Regards

Andrew
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[FFmpeg-devel] why prebuilt windows ffmpeg use rtmpdump instead of its native rtmp plugin

[qw]

Hi,

Previously, I sent email to ask which rtmp plugin is better. The answer is 
ffmpeg's native rtmp plugin is better than rtmpdump.

But why does official prebuilt windows ffmpeg use rtmpdump instead of its 
native rtmp plugin?

Thanks!

Regards

andrew
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[FFmpeg-devel] questions about multi-thread issues for ffmpeg

[qw]

Hi,


I have made one simple transcoding program by using ffmpeg lib, where several 
av transcoding tasks are done in one process and each task corresponds to one 
thread.


But some ffmpeg function will report error, and sometimes the multi-thread 
version of transcoding program will crash abnormally.


For example, av_guess_format("mp4", NULL, NULL) returns error accidentally with 
several concurrent tasks. And transcoding program will crash, when calling 
av_interleaved_write_frame ().


does ffmpeg support multi-thread in this case? If not, which ffmpeg functions 
should be synchronized by application developer?


Thanks!


Regards


Andrew



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Re: [FFmpeg-devel] questions about multi-thread issues for ffmpeg

[qw]

hi Ronald,


>does ffmpeg support multi-thread in this case? If not, which ffmpeg
>> functions should be synchronized by application developer?
>
>
>Yes, ffmpeg supports multi-threading (internally), it is tested
>continuously and is considered very safe.
>
>Can you show some code? I'm wondering if your use is unusual.



In my side, ffmpeg 2.8.6 is used to make the simple transcoding program, while 
x264 as video encoder with 2 pass rate control algorithm.


Because of using 2-pass rate control, AVFormatContext is created with setting 
AVFMT_NOFILE for 1st pass, which just encodes video stream. After encoding the 
total video stream, video and audio streams are encoded for 2nd pass. If two 
transocding tasks are started simultaneously, and each corresponds to one 
thread, AVFormatContext::oformat->flags is changed to unset AVFMT_NOFILE at the 
end of 1st pass encoding, i.e. flushing bufferred video from x264 encoder.


If AVFMT_NOFILE is unset in AVFormatContext::oformat->flags for 1st pass, there 
is no crash issue any more.


Thanks!


Regards


Andrew



At 2016-09-08 17:56:07, "Ronald S. Bultje"  wrote:
>Hi Andrew,
>
>On Thu, Sep 8, 2016 at 3:59 AM, qw  wrote:
>
>> I have made one simple transcoding program by using ffmpeg lib, where
>> several av transcoding tasks are done in one process and each task
>> corresponds to one thread.
>>
>> But some ffmpeg function will report error, and sometimes the multi-thread
>> version of transcoding program will crash abnormally.
>>
>> For example, av_guess_format("mp4", NULL, NULL) returns error accidentally
>> with several concurrent tasks. And transcoding program will crash, when
>> calling av_interleaved_write_frame ().
>>
>
>Have you tried running this under valgrind? It is typically very helpful to
>see what causes the crash.
>
>does ffmpeg support multi-thread in this case? If not, which ffmpeg
>> functions should be synchronized by application developer?
>
>
>Yes, ffmpeg supports multi-threading (internally), it is tested
>continuously and is considered very safe.
>
>Can you show some code? I'm wondering if your use is unusual.
>
>Ronald
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[FFmpeg-devel] how to build ffmpeg with -O0 instead of -O1

[qw]

Hi,
 
I want to build ffmpeg without optimization options, which means compiling 
flags uses O0 instead of O1-O3. I use the following command to build ffmpeg, 
where O1 is used in config.mak.
 
./configure --enable-debug --disable-optimizations ..

I want to disable all optimization options to debug ffmpeg/ffmpeg_g program. If 
I change optimization options in config.mak, i.e. using O0 instead of O1, there 
will be some error messages during the building process:
 
libavutil/libavutil.a(cpu.o): In function `av_get_cpu_flags':
/export/qiang/sourceCode/ffmpeg-2.8.3/libavutil/cpu.c:82: undefined reference 
to `ff_get_cpu_flags_aarch64'
/export/qiang/sourceCode/ffmpeg-2.8.3/libavutil/cpu.c:84: undefined reference 
to `ff_get_cpu_flags_arm'
/export/qiang/sourceCode/ffmpeg-2.8.3/libavutil/cpu.c:86: undefined reference 
to `ff_get_cpu_flags_ppc'

How to build ffmpeg with O0 option?
 
B.R.
 
Andrew
 




 





 
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[FFmpeg-devel] one question about aac bitstream in mov container

[qw]

Hi,
 
I am reading ffmpeg source code, and confused about the following logic in 
movenc.c,
 
if (enc->codec_id == AV_CODEC_ID_AAC && pkt->size > 2 &&
(AV_RB16(pkt->data) & 0xfff0) == 0xfff0) {
if (!s->streams[pkt->stream_index]->nb_frames) {
av_log(s, AV_LOG_ERROR, "Malformed AAC bitstream detected: "
   "use the audio bitstream filter 'aac_adtstoasc' to fix it "
   "('-bsf:a aac_adtstoasc' option with ffmpeg)\n");
return -1;
}
av_log(s, AV_LOG_WARNING, "aac bitstream error\n");
}

If I used default setting to encode audio samples into packets and moved those 
packets into mp4 container, my transcoding program will report:
 
[mov @ 0xe30a00] Malformed AAC bitstream detected: use the audio bitstream 
filter 'aac_adtstoasc' to fix it ('-bsf:a aac_adtstoasc' option with ffmpeg)

how to set fdk_aac encoder to code audio samples without the error message?
 
Thanks!
 
B.R.
 
Andrew
 
 
 
 
 
 
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[FFmpeg-devel] one question about aac bitstream in mov container

[qw]

Hi,
 
I am reading ffmpeg source code, and confused about the following logic in 
movenc.c,
 
if (enc->codec_id == AV_CODEC_ID_AAC && pkt->size > 2 &&
(AV_RB16(pkt->data) & 0xfff0) == 0xfff0) {
if (!s->streams[pkt->stream_index]->nb_frames) {
av_log(s, AV_LOG_ERROR, "Malformed AAC bitstream detected: "
   "use the audio bitstream filter 'aac_adtstoasc' to fix it "
   "('-bsf:a aac_adtstoasc' option with ffmpeg)\n");
return -1;
}
av_log(s, AV_LOG_WARNING, "aac bitstream error\n");
}

If I used default setting to encode audio samples into packets and moved those 
packets into mp4 container, my transcoding program will report:
 
[mov @ 0xe30a00] Malformed AAC bitstream detected: use the audio bitstream 
filter 'aac_adtstoasc' to fix it ('-bsf:a aac_adtstoasc' option with ffmpeg)

why movenc.c can't support ADTS format? how to set fdk_aac encoder to code 
audio samples without the error message?
 
Thanks!
 
B.R.
 
Andrew
 
 
 
 
 
 




 
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[FFmpeg-devel] how to know via ffmpeg lib whether disk is full or not

[qw]

Hi,

In some rare case, when there are huge AV transcoding tasks, disk will become 
full if old files are not deleted immediately. Then those transcoding tasks 
can't work correctly.

In general transcoding application, av_interleaved_write_frame() and 
av_read_frame() are used to make media files, such as mp4, flv, and mkv.

Do the two functions or other ffmpeg functions support the function that ffmpeg 
lib can detect the disk status and report error when disk is full? And how to 
know via ffmpeg lib whether disk is full or not?

Thanks!

B.R.

Andrew
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[FFmpeg-devel] why ffmpeg reports error when decoding rm or rmvb files

[qw]

Hi,

I'm using the following command to decode some rm and rmvb files, and ffmpeg 
reports error as shown below:

[root@PT-18376 test-clips]# ffmpeg -i hanma.rm -xerror -f null /dev/null
ffmpeg version 2.8.3 Copyright (c) 2000-2015 the FFmpeg developers
  built with icc (ICC) 14.0.2 20140120
  configuration: --cc=/opt/intel/bin/icc --enable-version3 --enable-asm 
--enable-yasm --enable-avfilter --enable-libvidstab --disable-static 
--enable-shared --enable-libx264 --enable-gpl --prefix=/usr/local/ 
--extra-cflags=-I/usr/local/include --extra-ldflags=-L/usr/local/lib 
--enable-libfdk_aac --enable-nonfree --enable-libass --enable-libfreetype 
--extra-libs=-lfreetype
  libavutil  54. 31.100 / 54. 31.100
  libavcodec 56. 60.100 / 56. 60.100
  libavformat56. 40.101 / 56. 40.101
  libavdevice56.  4.100 / 56.  4.100
  libavfilter 5. 40.101 /  5. 40.101
  libswscale  3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc53.  3.100 / 53.  3.100
[rm @ 0x1d703a0] Invalid stream index 2 for index at pos 7110127
Input #0, rm, from 'hanma.rm':
  Metadata:
title   :
author  :
copyright   :
comment :
ASMRuleBook : #($Bandwidth >= 0),Stream1Bandwidth = 64082, 
Stream0Bandwidth = 285918;
Audiences   : 350k Download (VBR);
audioMode   : music
Creation Date   : 4/19/2016 14:41:27
Generated By: Helix Producer SDK 11.0 for Windows, Build 11.0.0.2013
Modification Date: 4/19/2016 14:41:27
videoMode   : sharp
  Duration: 00:02:39.75, start: 0.00, bitrate: 356 kb/s
Stream #0:0: Video: rv40 (RV40 / 0x30345652), yuv420p, 448x336, 286 kb/s, 
30 fps, 30 tbr, 1k tbn, 1k tbc
Stream #0:1: Audio: cook (cook / 0x6B6F6F63), 44100 Hz, stereo, fltp, 64 
kb/s
Output #0, null, to '/dev/null':
  Metadata:
title   :
author  :
copyright   :
comment :
ASMRuleBook : #($Bandwidth >= 0),Stream1Bandwidth = 64082, 
Stream0Bandwidth = 285918;
Audiences   : 350k Download (VBR);
audioMode   : music
Creation Date   : 4/19/2016 14:41:27
Generated By: Helix Producer SDK 11.0 for Windows, Build 11.0.0.2013
Modification Date: 4/19/2016 14:41:27
videoMode   : sharp
encoder : Lavf56.40.101
Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 448x336, q=2-31, 
200 kb/s, 30 fps, 30 tbn, 30 tbc
Metadata:
  encoder : Lavc56.60.100 rawvideo
Stream #0:1: Audio: pcm_s16le, 44100 Hz, stereo, s16, 1411 kb/s
Metadata:
  encoder : Lavc56.60.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (rv40 (native) -> rawvideo (native))
  Stream #0:1 -> #0:1 (cook (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
hanma.rm: Input/output errorN/A   
Conversion failed!

From the above message, ffmpeg reports error when reaching the end of input 
files, because av_read_frame() return a value that is not AVERROR_EOF. I use 
the same command to decode other clips, such mkv and mp4.

Why ffmpeg reports error when decoding rm and rmvb files?

rm and rmvb samples can be downloaded from
http://pan.baidu.com/s/1hs113Bu
http://pan.baidu.com/s/1qYKNY7y
 
Thanks!

B.R.

andrew
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Re: [FFmpeg-devel] why ffmpeg reports error when decoding rm or rmvb files

[qw]

Hi Moritz,

Thanks for your reply. What does 'the thread went OT' mean?


How to know whether my presented question has been resolved, if someone has 
sent some solution on ffmpeg forum, like 
'http://ffmpeg.org/pipermail/ffmpeg-devel/2016-May/194083.html'?

Thanks!

B.R.

Andrew


At 2016-05-16 23:04:05, "Moritz Barsnick"  wrote:
>On Mon, May 16, 2016 at 09:46:15 -0400, compn wrote:
>> > I'm using the following command to decode some rm and rmvb files, and
>> > ffmpeg reports error as shown below:
>> 
>> hi, does error still occur with latest 2016 ffmpeg?
>
>I believe this patch was Carl Eugen's suggestion for a fix, but the
>thread went OT...
>http://ffmpeg.org/pipermail/ffmpeg-devel/2016-May/194083.html
>
>Moritz
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[FFmpeg-devel] does the latest ffmpeg support h264 SVC

[qw]

Hi,

I have one question about the function of the latest ffmepg. Does the latest 
ffmpeg support H.264 SVC? Why ffmpeg doesn't support H.264 SVC?

Thanks!

B.R.

Andrew
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[FFmpeg-devel] how to deal with the situation of full disk

[qw]

Hi,

In some rare case, when there are huge AV transcoding tasks, disk will become 
full if old files are not deleted immediately. Then those running transcoding 
tasks may not work correctly.

In general transcoding application, av_interleaved_write_frame() and 
av_read_frame() are used to write out av packets and make media files, such as 
mp4, flv, and mkv.

Can the two functions or other ffmpeg functions detect the disk status and 
report error when disk is full? And how to know via ffmpeg lib whether disk is 
full or not?

Thanks!

B.R.

Andrew
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[FFmpeg-devel] One questions about 3D video

[qw]

Hi,

My quesions are shown below:

1) Is there some way to check whether media files contain 3d Video or not?

Thanks!

B.R.

andrew
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[FFmpeg-devel] one question about h.264 svc decoder

[qw]

Hi,

I found ffmpeg has supported openh264 now. Does ffmpeg support h.264 SVC 
deocding, i.e. capability of decoding temporal, spatial and quality scalability 
bitstream?

Thanks!

B.R.

Andrew
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[FFmpeg-devel] why ffmpeg doesn't support vp9 svc?

[qw]

Hi,

I have one question about vp9. vp9 supports svc encoding/decoding. Why ffmpeg 
doesn't support vp9 svc?

Thanks!

B.R.

Andrew
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[FFmpeg-devel] how to use two rtmp plugins in ffmpeg

[qw]

Hi,

ffmpeg has its own built-in rtmp plugin, and also support to use rtmpdump as 
another rtmp plugin.

How to use specified rtmp plugin, i.e. build-in rtmp plugin or rtmpdump plugin, 
if rtmpdump is enabled and built within ffmpeg?

Thanks!

Regards

Andrew
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[FFmpeg-devel] how to parse audio/video information for rtmp protocal

[qw]

Hi,

FFmpeg has its native rtmp plugin, and also can use rtmpdump instead if 
enable-librtmp is chosen when building ffmpeg.

I have reviewed rtmp specification, where client and server can notify its peer 
of audio/video information via metadata message, and of audio/video codec via 
connect command.

For live-streamming application, what is the common way to get audio/video 
information, such as bit rate, audio channel number, sample rate, and frame 
rate? Are there some ffmpeg functions to get those information?

Thanks!

Regards

Andrew
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[FFmpeg-devel] which rtmpdump is beter for fmpeg's native rtmp lib and rtmpdump

[qw]

Hi,

I found ffmpeg support two rtmp libs, where one is ffmpeg's native rtmp lib, 
and the other is rtmpdump. Which is better?

Thanks!

Regards

Andrew
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[FFmpeg-devel] does ffmpeg support AEC?

[qw]

Hi,

Does ffmpeg support AEC, i.e. Acoustic Echo Cancellation? Or Does ffmpeg has 
third-party dependency lib that supports AEC?

Thanks!

Regards

Andrew
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[FFmpeg-devel] fail to build ffmpeg with icc 14

[qw]

Hi,


I use the following command to build ffmpeg:


./configure --cc=/opt/intel/bin/icc --enable-version3 --enable-asm 
--enable-avfilter --disable-static --enable-shared --enable-gpl 
--enable-nonfree --prefix=/usr/local/ --extra-cflags='-I/usr/local/include 
--extra-ldflags='-L/usr/local/lib --enable-stripping
make


But I fail to compile ffmpeg, and error message is shown as below:


icc: command line warning #10121: overriding '-std=gnu99' with '-std=c99'
libavutil/cpu.c(48): error: identifier "ATOMIC_VAR_INIT" is undefined
  static atomic_int cpu_flags = ATOMIC_VAR_INIT(-1);
^


libavutil/cpu.c(48): error: function call is not allowed in a constant 
expression
  static atomic_int cpu_flags = ATOMIC_VAR_INIT(-1);
^


compilation aborted for libavutil/cpu.c (code 2)
make: *** [libavutil/cpu.o] Error 2
make: *** Waiting for unfinished jobs
In file included from /opt/intel/include/xmmintrin.h(24),
 from /opt/intel/include/emmintrin.h(39),
 from /opt/intel/include/pmmintrin.h(21),
 from /opt/intel/include/tmmintrin.h(14),
 from /opt/intel/include/smmintrin.h(23),
 from /opt/intel/include/nmmintrin.h(24),
 from /opt/intel/include/wmmintrin.h(23),
 from /opt/intel/include/immintrin.h(14),
 from libavutil/x86/intmath.h(30),
 from libavutil/intmath.h(33),
 from libavutil/common.h(106),
 from libavutil/display.h(25),
 from libavutil/display.c(25):
/opt/intel/include/mmintrin.h(162): warning #2959: function prototype is 
missing the argument types
  extern __m64 __ICL_INTRINCC _mm_setzero_si64();
  ^
Why?


Thanks


Regards


Andrew
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[FFmpeg-devel] fail to seek key video packet

[qw]

Hi,


I use following command to test ffmpeg-2.8.1:


ffmpeg -debug_ts -ss 10 -t 1 -i hanma.ts -f null /dev/null




ffmpeg version 2.8.1 Copyright (c) 2000-2015 the FFmpeg developers
  built with icc (ICC) 14.0.2 20140120
  configuration: --cc=/opt/intel/bin/icc --enable-version3 --enable-asm 
--enable-yasm --enable-avfilter --disable-static --enable-shared --enable-gpl 
--enable-nonfree --prefix=/usr/local/ --extra-cflags='-I/usr/local/include 
--extra-ldflags=-L/usr/local/lib' --enable-memalign-hack --enable-stripping
  libavutil  54. 31.100 / 54. 31.100
  libavcodec 56. 60.100 / 56. 60.100
  libavformat56. 40.101 / 56. 40.101
  libavdevice56.  4.100 / 56.  4.100
  libavfilter 5. 40.101 /  5. 40.101
  libswscale  3.  1.101 /  3.  1.101
  libswresample   1.  2.101 /  1.  2.101
  libpostproc53.  3.100 / 53.  3.100
Input #0, mpegts, from './sampleClips/jd-testClips/clips03/hanma.ts':
  Duration: 00:02:39.78, start: 1.40, bitrate: 10913 kb/s
  Program 1 
Metadata:
  service_name: Service01
  service_provider: FFmpeg
Stream #0:0[0x100]: Video: h264 (Constrained Baseline) ([27][0][0][0] / 
0x001B), yuv420p, 1280x720 [SAR 3:4 DAR 4:3], 24 fps, 24 tbr, 90k tbn, 48 tbc
Stream #0:1[0x101]: Audio: ac3 ([129][0][0][0] / 0x0081), 48000 Hz, stereo, 
fltp, 192 kb/s
Output #0, null, to '/dev/null':
  Metadata:
encoder : Lavf56.40.101
Stream #0:0: Video: rawvideo (I420 / 0x30323449), yuv420p, 1280x720 [SAR 
3:4 DAR 4:3], q=2-31, 200 kb/s, 24 fps, 24 tbn, 24 tbc
Metadata:
  encoder : Lavc56.60.100 rawvideo
Stream #0:1: Audio: pcm_s16le, 48000 Hz, stereo, s16, 1536 kb/s
Metadata:
  encoder : Lavc56.60.100 pcm_s16le
Stream mapping:
  Stream #0:0 -> #0:0 (h264 (native) -> rawvideo (native))
  Stream #0:1 -> #0:1 (ac3 (native) -> pcm_s16le (native))
Press [q] to stop, [?] for help
demuxer -> ist_index:1 type:audio next_dts:NOPTS next_dts_time:NOPTS 
next_pts:NOPTS next_pts_time:NOPTS pkt_pts:1024560 pkt_pts_time:11.384 
pkt_dts:1024560 pkt_dts_time:11.384 off:-1140 off_time:-11.4
demuxer+ffmpeg -> ist_index:1 type:audio pkt_pts:-1440 pkt_pts_time:-0.016 
pkt_dts:-1440 pkt_dts_time:-0.016 off:-1140 off_time:-11.4
encoder <- type:audio frame_pts:0 frame_pts_time:0 time_base:1/48000
encoder -> type:audio pkt_pts:0 pkt_pts_time:0 pkt_dts:0 pkt_dts_time:0
muxer <- type:audio pkt_pts:0 pkt_pts_time:0 pkt_dts:0 pkt_dts_time:0 size:3072
demuxer -> ist_index:1 type:audio next_dts:16000 next_dts_time:0.016 
next_pts:16000 next_pts_time:0.016 pkt_pts:1027440 pkt_pts_time:11.416 
pkt_dts:1027440 pkt_dts_time:11.416 off:-1140 off_time:-11.4
demuxer+ffmpeg -> ist_index:1 type:audio pkt_pts:1440 pkt_pts_time:0.016 
pkt_dts:1440 pkt_dts_time:0.016 off:-1140 off_time:-11.4
encoder <- type:audio frame_pts:768 frame_pts_time:0.016 time_base:1/48000
encoder -> type:audio pkt_pts:768 pkt_pts_time:0.016 pkt_dts:768 
pkt_dts_time:0.016
muxer <- type:audio pkt_pts:768 pkt_pts_time:0.016 pkt_dts:768 
pkt_dts_time:0.016 size:6144
demuxer -> ist_index:1 type:audio next_dts:48000 next_dts_time:0.048 
next_pts:48000 next_pts_time:0.048 pkt_pts:1030320 pkt_pts_time:11.448 
pkt_dts:1030320 pkt_dts_time:11.448 off:-1140 off_time:-11.4
demuxer+ffmpeg -> ist_index:1 type:audio pkt_pts:4320 pkt_pts_time:0.048 
pkt_dts:4320 pkt_dts_time:0.048 off:-1140 off_time:-11.4
encoder <- type:audio frame_pts:2304 frame_pts_time:0.048 time_base:1/48000
encoder -> type:audio pkt_pts:2304 pkt_pts_time:0.048 pkt_dts:2304 
pkt_dts_time:0.048
muxer <- type:audio pkt_pts:2304 pkt_pts_time:0.048 pkt_dts:2304 
pkt_dts_time:0.048 size:6144
demuxer -> ist_index:0 type:video next_dts:NOPTS next_dts_time:NOPTS 
next_pts:NOPTS next_pts_time:NOPTS pkt_pts:1026000 pkt_pts_time:11.4 
pkt_dts:1026000 pkt_dts_time:11.4 off:-1140 off_time:-11.4
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:0 pkt_pts_time:0 pkt_dts:0 
pkt_dts_time:0 off:-1140 off_time:-11.4
demuxer -> ist_index:0 type:video next_dts:41667 next_dts_time:0.041667 
next_pts:0 next_pts_time:0 pkt_pts:1029750 pkt_pts_time:11.4417 pkt_dts:1029750 
pkt_dts_time:11.4417 off:-1140 off_time:-11.4
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:3750 pkt_pts_time:0.0416667 
pkt_dts:3750 pkt_dts_time:0.0416667 off:-1140 off_time:-11.4
[h264 @ 0x1bdbda0] Missing reference picture, default is 0
[h264 @ 0x1bdbda0] decode_slice_header error
demuxer -> ist_index:0 type:video next_dts:83334 next_dts_time:0.083334 
next_pts:0 next_pts_time:0 pkt_pts:1033500 pkt_pts_time:11.4833 pkt_dts:1033500 
pkt_dts_time:11.4833 off:-1140 off_time:-11.4
demuxer+ffmpeg -> ist_index:0 type:video pkt_pts:7500 pkt_pts_time:0.083 
pkt_dts:7500 pkt_dts_time:0.083 off:-1140 off_time:-11.4
demuxer -> ist_index:1 type:audio next_dts:8 next_dts_time:0.08 
next_pts:8 next_pts_time:0.08 pkt_pts:1033200 pkt_pts_time:11.48 
pkt_dts:1033200 pkt_dts_time:11.48 off:-1140 off_time:-11.4
demuxer+ffm

Re: [FFmpeg-devel] fail to seek key video packet

[qw]

>> > I use following command to test ffmpeg-2.8.1:
>> 
>> This list is for development of ffmpeg, not for user support or bug reports.
>> 
>> Use the ffmpeg-user list for user support, and http://trac.ffmpeg.org/
>> to report bugs.
>
>And please don't expect anyone to help debug issues with ffmpeg-2.8.1,
>or even accept a bug report against it.


I think the latest ffmpeg may have the same bug. FFmpeg-devel can check the 
issue and fix it if it is.



Thanks!


B.R.


Andrew


At 2018-02-03 06:53:26, "Moritz Barsnick"  wrote:
>On Fri, Feb 02, 2018 at 12:02:29 -0300, James Almer wrote:
>> > I use following command to test ffmpeg-2.8.1:
>> 
>> This list is for development of ffmpeg, not for user support or bug reports.
>> 
>> Use the ffmpeg-user list for user support, and http://trac.ffmpeg.org/
>> to report bugs.
>
>And please don't expect anyone to help debug issues with ffmpeg-2.8.1,
>or even accept a bug report against it.
>
>Moritz
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[FFmpeg-devel] how to flush audio delayed frames

[qw]

hi,


avcodec_encode_audio() is used to encode audio frame. How to flush audio 
delayed frames in audio encoder?


Thanks!


B.R.


Andrew
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