[FFmpeg-user] ffmpeg amerge and amix filter delay

Paul B Mahol onemda at gmail.com
Tue Nov 3 09:21:37 CET 2015


On 11/2/15, Paul B Mahol <onemda at gmail.com> wrote:
> Dana 2. 11. 2015. 20:26 osoba "Leo GSA" <leogsa at gmail.com> napisala je:
>>
>> ffmpeg amerge and amix filter delay
>>
>> I need to take audio-streams from several IP cameras and merge them into
>> one file, so that they would sound simaltaneousely.
>>
>> I tried filter "amix": (for testing purposes I take audio-stream 2 times
>> from the same camera. yes, I tried 2 cameras - result is the same)
>>
>>
>> ffmpeg -i rtsp://user:pass@172.22.5.202 -i rtsp://user:pass@172.22.5.202
>> -map 0:a -map 1:a  -filter_complex
>> amix=inputs=2:duration=first:dropout_transition=3  -ar 22050 -vn -f flv
>> rtmp://172.22.45.38:1935/live/stream1
>>
>>
>> result: I say "hello". And hear in speakers the first "hello" and in 1
>> second I hear the second "hello". Instead of hearing two "hello"'s
>> simaltaneousely.
>>
>> and tried filter "amerge":
>>
>>
>> ffmpeg -i rtsp://user:pass@172.22.5.202 -i rtsp://user:pass@172.22.5.202
>> -map 0:a -map 1:a  -filter_complex amerge -ar 22050 -vn -f flv rtmp://
>> 172.22.45.38:1935/live/stream1
>>
>>
>> result: the same as in the first example, but now I hear the first
>> "hello"
>> in left speaker and in 1 second I hear the second "hello" in right
> speaker,
>> instead of hearing two "hello"'s in both speakers simaltaneousely.
>>
>> So, the question is: how to make them sound simaltaneousely? May be you
>> know some parameter? or some other command?
>>
>> P.S. Here is ful command-line output for both variants: amix:
>>
>>     ffmpeg -i rtsp://admin:12345@172.22.5.202 -i rtsp://
>> admin:12345 at 172.22.5.202 -map 0:a -map 1:a -filter_complex
>> amix=inputs=2:duration=longest:dropout_transition=0 -vn -ar 22050 -f flv
>> rtmp://172.22.45.38:1935/live/stream1       ffmpeg version
> N-76031-g9099079
>> Copyright (c) 2000-2015 the FFmpeg developers
>>       built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-16)
>>       configuration: --enable-gpl --enable-libx264 --enable-libmp3lame
>> --enable-nonfree --enable-version3
>>       libavutil      55.  4.100 / 55.  4.100
>>       libavcodec     57.  6.100 / 57.  6.100
>>       libavformat    57.  4.100 / 57.  4.100
>>       libavdevice    57.  0.100 / 57.  0.100
>>       libavfilter     6. 11.100 /  6. 11.100
>>       libswscale      4.  0.100 /  4.  0.100
>>       libswresample   2.  0.100 /  2.  0.100
>>       libpostproc    54.  0.100 / 54.  0.100
>>     Input #0, rtsp, from 'rtsp://admin:12345@172.22.5.202':
>>       Metadata:
>>         title           : Media Presentation
>>       Duration: N/A, start: 0.032000, bitrate: N/A
>>         Stream #0:0: Video: h264 (Baseline), yuv420p, 1280x720, 20 fps,
>> 25
>> tbr, 90k tbn, 40 tbc
>>         Stream #0:1: Audio: adpcm_g726, 8000 Hz, mono, s16, 16 kb/s
>>         Stream #0:2: Data: none
>>     Input #1, rtsp, from 'rtsp://admin:12345@172.22.5.202':
>>       Metadata:
>>         title           : Media Presentation
>>       Duration: N/A, start: 0.032000, bitrate: N/A
>>         Stream #1:0: Video: h264 (Baseline), yuv420p, 1280x720, 20 fps,
>> 25
>> tbr, 90k tbn, 40 tbc
>>         Stream #1:1: Audio: adpcm_g726, 8000 Hz, mono, s16, 16 kb/s
>>         Stream #1:2: Data: none
>>     Output #0, flv, to 'rtmp://172.22.45.38:1935/live/stream1':
>>       Metadata:
>>         title           : Media Presentation
>>         encoder         : Lavf57.4.100
>>         Stream #0:0: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002),
> 22050
>> Hz, mono, fltp (default)
>>         Metadata:
>>           encoder         : Lavc57.6.100 libmp3lame
>>     Stream mapping:
>>       Stream #0:1 (g726) -> amix:input0
>>       Stream #1:1 (g726) -> amix:input1
>>       amix -> Stream #0:0 (libmp3lame)
>>     Press [q] to stop, [?] for help
>>     [rtsp @ 0x2689600] Thread message queue blocking; consider raising
>> the
>> thread_queue_size option (current value: 8)
>>     [rtsp @ 0x2727c60] Thread message queue blocking; consider raising
>> the
>> thread_queue_size option (current value: 8)
>>     [rtsp @ 0x2689600] max delay reached. need to consume packet
>>     [NULL @ 0x268c500] RTP: missed 38 packets
>>     [rtsp @ 0x2689600] max delay reached. need to consume packet
>>     [NULL @ 0x268d460] RTP: missed 4 packets
>>     [flv @ 0x2958360] Failed to update header with correct duration.
>>     [flv @ 0x2958360] Failed to update header with correct filesize.
>>     size=      28kB time=00:00:06.18 bitrate=  36.7kbits/s
>>     video:0kB audio:24kB subtitle:0kB other streams:0kB global
>> headers:0kB
>> muxing overhead: 16.331224%
>>
>>
>> and amerge:
>>
>> # ffmpeg -i rtsp://admin:12345@172.22.5.202 -i rtsp://
>> admin:12345 at 172.22.5.202 -map 0:a -map 1:a -filter_complex amerge -vn -ar
>> 22050 -f flv rtmp://172.22.45.38:1935/live/stream1
>>     ffmpeg version N-76031-g9099079 Copyright (c) 2000-2015 the FFmpeg
>> developers
>>       built with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-16)
>>       configuration: --enable-gpl --enable-libx264 --enable-libmp3lame
>> --enable-nonfree --enable-version3
>>       libavutil      55.  4.100 / 55.  4.100
>>       libavcodec     57.  6.100 / 57.  6.100
>>       libavformat    57.  4.100 / 57.  4.100
>>       libavdevice    57.  0.100 / 57.  0.100
>>       libavfilter     6. 11.100 /  6. 11.100
>>       libswscale      4.  0.100 /  4.  0.100
>>       libswresample   2.  0.100 /  2.  0.100
>>       libpostproc    54.  0.100 / 54.  0.100
>>     Input #0, rtsp, from 'rtsp://admin:12345@172.22.5.202':
>>       Metadata:
>>         title           : Media Presentation
>>       Duration: N/A, start: 0.064000, bitrate: N/A
>>         Stream #0:0: Video: h264 (Baseline), yuv420p, 1280x720, 20 fps,
>> 25
>> tbr, 90k tbn, 40 tbc
>>         Stream #0:1: Audio: adpcm_g726, 8000 Hz, mono, s16, 16 kb/s
>>         Stream #0:2: Data: none
>>     Input #1, rtsp, from 'rtsp://admin:12345@172.22.5.202':
>>       Metadata:
>>         title           : Media Presentation
>>       Duration: N/A, start: 0.032000, bitrate: N/A
>>         Stream #1:0: Video: h264 (Baseline), yuv420p, 1280x720, 20 fps,
>> 25
>> tbr, 90k tbn, 40 tbc
>>         Stream #1:1: Audio: adpcm_g726, 8000 Hz, mono, s16, 16 kb/s
>>         Stream #1:2: Data: none
>>     [Parsed_amerge_0 @ 0x3069cc0] No channel layout for input 1
>>     [Parsed_amerge_0 @ 0x3069cc0] Input channel layouts overlap: output
>> layout will be determined by the number of distinct input channels
>>     Output #0, flv, to 'rtmp://172.22.45.38:1935/live/stream1':
>>       Metadata:
>>         title           : Media Presentation
>>         encoder         : Lavf57.4.100
>>         Stream #0:0: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002),
> 22050
>> Hz, stereo, s16p (default)
>>         Metadata:
>>           encoder         : Lavc57.6.100 libmp3lame
>>     Stream mapping:
>>       Stream #0:1 (g726) -> amerge:in0
>>       Stream #1:1 (g726) -> amerge:in1
>>       amerge -> Stream #0:0 (libmp3lame)
>>     Press [q] to stop, [?] for help
>>     [rtsp @ 0x2f71640] Thread message queue blocking; consider raising
>> the
>> thread_queue_size option (current value: 8)
>>     [rtsp @ 0x300fb40] Thread message queue blocking; consider raising
>> the
>> thread_queue_size option (current value: 8)
>>     [rtsp @ 0x2f71640] max delay reached. need to consume packet
>>     [NULL @ 0x2f744a0] RTP: missed 18 packets
>>     [flv @ 0x3058b00] Failed to update header with correct duration.
>>     [flv @ 0x3058b00] Failed to update header with correct filesize.
>>     size=      39kB time=00:00:04.54 bitrate=  70.2kbits/s
>>     video:0kB audio:36kB subtitle:0kB other streams:0kB global
>> headers:0kB
>> muxing overhead: 8.330614%
>>
>>
>>
>> Thanx.
>>
>> UPDATE 30 oct 2015: I found interesting detail when connecting 2 cameras
>> (they have different microphones and I hear the difference between them):
>> the order of "Hello"'s from different cams depends on the ORDER OF
>> INPUTS.
>> with command
>>
>>
>> ffmpeg -i rtsp://cam2 -i rtsp://cam1 -map 0:a -map 1:a -filter_complex
>> amix=inputs=2:duration=longest:dropout_transition=0 -vn -ar 22050 -f flv
>> rtmp://172.22.45.38:1935/live/stream1
>>
>>
>> I hear "hello" from 1st cam and then in 1 second "hello" from 2nd cam.
>>
>> with command
>>
>> ffmpeg -i rtsp://cam1 -i rtsp://cam2 -map 0:a -map 1:a -filter_complex
>> amix=inputs=2:duration=longest:dropout_transition=0 -vn -ar 22050 -f flv
>> rtmp://172.22.45.38:1935/live/stream1
>>
>> I hear "hello" from 2nd cam and then in 1 second "hello" from 1st cam.
>>
>> So, As I understand - ffmpeg takes inputs not simaltaneousely, but in the
>> order of inputs given.
>> Question: how to tell ffmpeg to read inputs simaltaneousely?
>
> Looks like rtmp issue, do you get same issue using files?


What if you increase -thread_queue_size to 64?

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