[FFmpeg-user] ffmpeg amerge and amix filter delay
Paul B Mahol
onemda at gmail.com
Mon Nov 2 21:03:57 CET 2015
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?
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