[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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>
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