Advertisement
Not a member of Pastebin yet?
Sign Up,
it unlocks many cool features!
- 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@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@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?
Advertisement
Add Comment
Please, Sign In to add comment
Advertisement