[FFmpeg-user] ffmpeg amerge and amix filter delay
Leo GSA
leogsa at gmail.com
Mon Nov 2 09:52:12 CET 2015
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?
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