Dana 2. 11. 2015. 20:26 osoba "Leo GSA" <[email protected]> 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:[email protected] -i rtsp://user:[email protected] > -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:[email protected] -i rtsp://user:[email protected] > -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:[email protected] -i rtsp:// > admin:[email protected] -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:[email protected]': > 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:[email protected]': > 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:[email protected] -i rtsp:// > admin:[email protected] -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:[email protected]': > 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:[email protected]': > 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? > _______________________________________________ > ffmpeg-user mailing list > [email protected] > http://ffmpeg.org/mailman/listinfo/ffmpeg-user _______________________________________________ ffmpeg-user mailing list [email protected] http://ffmpeg.org/mailman/listinfo/ffmpeg-user
