Is it possible at all to convert G.711 audio to AAC audio in a stream using ffmpeg? My cam is streaming fine in Chrome (video + audio) but all the other browsers have no audio. I would appreciate any help / input / suggestions / comments at all. Many thanks in advance!
Here is the code:
ffmpeg -rtsp_transport tcp -i rtsp://login:[email protected]:554/11 -map 0 -c:v copy -c:a copy -f
segment -segment_list /usr/local/nginx/html/hls/mystream.m3u8 -segment_list_type hls -
segment_list_size 10 -segment_list_flags +live /usr/local/nginx/html/hls/mystream%01d.ts
I have of course tried putting -c:a aac there before but it does not help.
There are millions of online tutorials and examples on how to convert a file. But it's not a file in my case it's a live stream. Am I missing something here? Do I perhaps somehow have to convert my final output file called mystream.m3u8?
P.S. Currently my cam is streaming in libx264 for video and G.711 for audio, but I can easily change it to stream in libx265 for video and G.726 for audio should that be of any help to play both video + audio. These are the only two options available in my camera and I cannot set it to anything else.
P.S.S. Here's the copy-pasted log, Gyan. I am not sure if that helps though. I do not see anything unusual there. Can you tell what is wrong?
ffmpeg started on 2020-04-08 at 04:54:52
Report written to "ffmpeg-20200408-045452.log"
Command line:
ffmpeg -rtsp_transport tcp -i rtsp://login:[email protected]:1/111 -map 0 -c:v copy -c:a aac -f segment -segment_list /usr/local/nginx/html/hls/mystream.m3u8 -segment_list_type hls -segment_list_size 10 -segment_list_flags +live "/usr/local/nginx/html/hls/mystream%01d.ts" -loglevel verbose -report -t 30
ffmpeg version 4.1.4-1~deb10u1 Copyright (c) 2000-2019 the FFmpeg developers
built with gcc 8 (Debian 8.3.0-6)
configuration: --prefix=/usr --extra-version='1~deb10u1' --toolchain=hardened --libdir=/usr/lib/x86_64-linux-gnu --incdir=/usr/include/x86_64-linux-gnu --arch=amd64 --enable-gpl --disable-stripping --enable-avresample --disable-filter=resample --enable-avisynth --enable-gnutls --enable-ladspa --enable-libaom --enable-libass --enable-libbluray --enable-libbs2b --enable-libcaca --enable-libcdio --enable-libcodec2 --enable-libflite --enable-libfontconfig --enable-libfreetype --enable-libfribidi --enable-libgme --enable-libgsm --enable-libjack --enable-libmp3lame --enable-libmysofa --enable-libopenjpeg --enable-libopenmpt --enable-libopus --enable-libpulse --enable-librsvg --enable-librubberband --enable-libshine --enable-libsnappy --enable-libsoxr --enable-libspeex --enable-libssh --enable-libtheora --enable-libtwolame --enable-libvidstab --enable-libvorbis --enable-libvpx --enable-libwavpack --enable-libwebp --enable-libx265 --enable-libxml2 --enable-libxvid --enable-libzmq --enable-libzvbi --enable-lv2 --en libavutil 56. 22.100 / 56. 22.100
libavcodec 58. 35.100 / 58. 35.100
libavformat 58. 20.100 / 58. 20.100
libavdevice 58. 5.100 / 58. 5.100
libavfilter 7. 40.101 / 7. 40.101
libavresample 4. 0. 0 / 4. 0. 0
libswscale 5. 3.100 / 5. 3.100
libswresample 3. 3.100 / 3. 3.100
libpostproc 55. 3.100 / 55. 3.100
Splitting the commandline.
Reading option '-rtsp_transport' ... matched as AVOption 'rtsp_transport' with argument 'tcp'.
Reading option '-i' ... matched as input url with argument 'rtsp://login:[email protected]:554/111'.
Reading option '-map' ... matched as option 'map' (set input stream mapping) with argument '0'.
Reading option '-c:v' ... matched as option 'c' (codec name) with argument 'copy'.
Reading option '-c:a' ... matched as option 'c' (codec name) with argument 'aac'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'segment'.
Reading option '-segment_list' ... matched as AVOption 'segment_list' with argument '/usr/local/nginx/html/hls/mystream.m3u8'.
Reading option '-segment_list_type' ... matched as AVOption 'segment_list_type' with argument 'hls'.
Reading option '-segment_list_size' ... matched as AVOption 'segment_list_size' with argument '10'.
Reading option '-segment_list_flags' ... matched as AVOption 'segment_list_flags' with argument '+live'.
Reading option '/usr/local/nginx/html/hls/mystream%01d.ts' ... matched as output url.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'verbose'.
Reading option '-report' ... matched as option 'report' (generate a report) with argument '1'.
Reading option '-t' ... matched as option 't' (record or transcode "duration" seconds of audio/video) with argument '30'.
Trailing options were found on the commandline.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument verbose.
Applying option report (generate a report) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input url rtsp://login:[email protected]:554/111.
Successfully parsed a group of options.
Opening an input file: rtsp://login:[email protected]:554/111.
[tcp @ 0x564b593fe780] No default whitelist set
[tcp @ 0x564b593fe780] Original list of addresses:
[tcp @ 0x564b593fe780] Address 8.2.1.11 port 554
[tcp @ 0x564b593fe780] Interleaved list of addresses:
[tcp @ 0x564b593fe780] Address 8.2.1.1 port 554
[tcp @ 0x564b593fe780] Starting connection attempt to 8.2.1.1 port 554
[tcp @ 0x564b593fe780] Successfully connected to 8.2.1.1 port 554
[rtsp @ 0x564b593fc540] SDP:
v=0
o=StreamingServer 3331435948 1116907222000 IN IP4 192.168.0.199
s=10
c=IN IP4 0.0.0.0
t=0 0
a=control:*
m=video 0 RTP/AVP 96
a=control:trackID=0
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1;profile-level-id=42002A;sprop-parameter-sets=Z0IAKpY1QPAET8s3AQEBAg==,aM48gA==
a=framesize:96 1920-1080
m=audio 0 RTP/AVP 8
a=control:trackID=1
a=rtpmap:8 PCMA/8000/1
a=ptime:20
a=fmtp:8 packetization-mode=1
[rtsp @ 0x564b593fc540] video codec set to: h264
[rtsp @ 0x564b593fc540] RTP Packetization Mode: 1
[rtsp @ 0x564b593fc540] RTP Profile IDC: 42 Profile IOP: 0 Level: 2a
[rtsp @ 0x564b593fc540] Extradata set to 0x564b594005f0 (size: 28)
[rtsp @ 0x564b593fc540] audio codec set to: pcm_alaw
[rtsp @ 0x564b593fc540] audio samplerate set to: 8000
[rtsp @ 0x564b593fc540] audio channels set to: 1
[rtsp @ 0x564b593fc540] setting jitter buffer size to 0
[rtsp @ 0x564b593fc540] setting jitter buffer size to 0
[rtsp @ 0x564b593fc540] hello state=0
[h264 @ 0x564b59400100] nal_unit_type: 7(SPS), nal_ref_idc: 3
[h264 @ 0x564b59400100] nal_unit_type: 8(PPS), nal_ref_idc: 3
[h264 @ 0x564b59400100] nal_unit_type: 7(SPS), nal_ref_idc: 3
[h264 @ 0x564b59400100] nal_unit_type: 8(PPS), nal_ref_idc: 3
[h264 @ 0x564b59400100] unknown SEI type 229
[h264 @ 0x564b59400100] nal_unit_type: 7(SPS), nal_ref_idc: 3
[h264 @ 0x564b59400100] nal_unit_type: 8(PPS), nal_ref_idc: 3
[h264 @ 0x564b59400100] nal_unit_type: 6(SEI), nal_ref_idc: 0
[h264 @ 0x564b59400100] nal_unit_type: 5(IDR), nal_ref_idc: 3
[h264 @ 0x564b59400100] unknown SEI type 229
[h264 @ 0x564b59400100] Format yuvj420p chosen by get_format().
[h264 @ 0x564b59400100] Reinit context to 1920x1088, pix_fmt: yuvj420p
[h264 @ 0x564b59400100] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 3
[h264 @ 0x564b59400100] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 3
[h264 @ 0x564b59400100] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 3
[h264 @ 0x564b59400100] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 3
[h264 @ 0x564b59400100] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 3
[h264 @ 0x564b59400100] nal_unit_type: 1(Coded slice of a non-IDR picture), nal_ref_idc: 3
[rtsp @ 0x564b593fc540] All info found
[rtsp @ 0x564b593fc540] rfps: 29.500000 0.018596
[rtsp @ 0x564b593fc540] rfps: 29.583333 0.012613
[rtsp @ 0x564b593fc540] rfps: 29.666667 0.007793
[rtsp @ 0x564b593fc540] rfps: 29.666667 0.007793
[rtsp @ 0x564b593fc540] rfps: 29.750000 0.004137
[rtsp @ 0x564b593fc540] rfps: 29.750000 0.004137
[rtsp @ 0x564b593fc540] rfps: 29.833333 0.001644
[rtsp @ 0x564b593fc540] rfps: 29.833333 0.001644
[rtsp @ 0x564b593fc540] rfps: 29.916667 0.000314
[rtsp @ 0x564b593fc540] rfps: 29.916667 0.000314
[rtsp @ 0x564b593fc540] rfps: 30.000000 0.000149
[rtsp @ 0x564b593fc540] rfps: 60.000000 0.000595
[rtsp @ 0x564b593fc540] rfps: 120.000000 0.002380
[rtsp @ 0x564b593fc540] rfps: 240.000000 0.009520
[rtsp @ 0x564b593fc540] rfps: 29.970030 0.000074
[rtsp @ 0x564b593fc540] rfps: 59.940060 0.000297
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://login:[email protected]:554/111':
Metadata:
title : 10
Duration: N/A, start: 0.000000, bitrate: N/A
Stream #0:0, 28, 1/90000: Video: h264 (Baseline), 1 reference frame, yuvj420p(pc, bt709, progressive, left), 1920x1080 (1920x1088), 29.97 tbr, 90k tbn, 180k tbc
Stream #0:1, 72, 1/8000: Audio: pcm_alaw, 8000 Hz, mono, s16, 64 kb/s
Successfully opened the file.
Parsing a group of options: output url /usr/local/nginx/html/hls/mystream%01d.ts.
Applying option map (set input stream mapping) with argument 0.
Applying option c:v (codec name) with argument copy.
Applying option c:a (codec name) with argument aac.
Applying option f (force format) with argument segment.
Successfully parsed a group of options.
Opening an output file: /usr/local/nginx/html/hls/mystream%01d.ts.
Successfully opened the file.
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #0:1 -> #0:1 (pcm_alaw (native) -> aac (native))
Press [q] to stop, [?] for help
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
detected 2 logical cores
[graph_0_in_0_1 @ 0x564b59439cc0] Setting 'time_base' to value '1/8000'
[graph_0_in_0_1 @ 0x564b59439cc0] Setting 'sample_rate' to value '8000'
[graph_0_in_0_1 @ 0x564b59439cc0] Setting 'sample_fmt' to value 's16'
[graph_0_in_0_1 @ 0x564b59439cc0] Setting 'channel_layout' to value '0x4'
[graph_0_in_0_1 @ 0x564b59439cc0] tb:1/8000 samplefmt:s16 samplerate:8000 chlayout:0x4
[format_out_0_1 @ 0x564b5943b040] Setting 'sample_fmts' to value 'fltp'
[format_out_0_1 @ 0x564b5943b040] Setting 'sample_rates' to value '96000|88200|64000|48000|44100|32000|24000|22050|16000|12000|11025|8000|7350'
[format_out_0_1 @ 0x564b5943b040] auto-inserting filter 'auto_resampler_0' between the filter 'Parsed_anull_0' and the filter 'format_out_0_1'
[AVFilterGraph @ 0x564b5941e880] query_formats: 4 queried, 6 merged, 3 already done, 0 delayed
[auto_resampler_0 @ 0x564b5957eb40] [SWR @ 0x564b59507740] Using s16p internally between filters
[auto_resampler_0 @ 0x564b5957eb40] ch:1 chl:mono fmt:s16 r:8000Hz -> ch:1 chl:mono fmt:fltp r:8000Hz
[aac @ 0x564b594200c0] Too many bits 8832.000000 > 6144 per frame requested, clamping to max
[segment @ 0x564b59403b80] Selected stream id:0 type:video
[segment @ 0x564b59403b80] Opening '/usr/local/nginx/html/hls/mystream0.ts' for writing
[file @ 0x564b59580b40] Setting default whitelist 'file,crypto'
[mpegts @ 0x564b59580480] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts
Output #0, segment, to '/usr/local/nginx/html/hls/mystream%01d.ts':
Metadata:
title : 10
encoder : Lavf58.20.100
Stream #0:0, 0, 1/90000: Video: h264 (Baseline), 1 reference frame, yuvj420p(pc, bt709, progressive, left), 1920x1080 (0x0), q=2-31, 29.97 tbr, 90k tbn, 90k tbc
Stream #0:1, 0, 1/90000: Audio: aac (LC), 8000 Hz, mono, fltp, delay 1024, 48 kb/s
Metadata:
encoder : Lavc58.35.100 aac
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[segment @ 0x564b59403b80] Timestamps are unset in a packet for stream 0. This is deprecated and will stop working in the future. Fix your code to set the timestamps properly)
cur_dts is invalid (this is harmless if it occurs once at the start per stream)
[segment @ 0x564b59403b80] segment:'/usr/local/nginx/html/hls/mystream0.ts' starts with packet stream:1 pts:0 pts_time:0 frame:0
[segment @ 0x564b59403b80] stream:1 start_pts_time:0 pts:0 pts_time:0 dts:0 dts_time:0 -> pts:0 pts_time:0 dts:0 dts_time:0
[segment @ 0x564b59403b80] segment:'/usr/local/nginx/html/hls/mystream0.ts' starts with packet stream:0 pts:11520 pts_time:0.128 frame:0
[segment @ 0x564b59403b80] stream:0 start_pts_time:0 pts:11520 pts_time:0.128 dts:11520 dts_time:0.128 -> pts:11520 pts_time:0.128 dts:11520 dts_time:0.128
[segment @ 0x564b59403b80] Opening '/usr/local/nginx/html/hls/mystream.m3u8.tmp' for writing
[file @ 0x564b59530900] Setting default whitelist 'file,crypto'
[segment @ 0x564b59403b80] EXT-X-MEDIA-SEQUENCE:0
[AVIOContext @ 0x564b5959f440] Statistics: 0 seeks, 1 writeouts
[segment @ 0x564b59403b80] segment:'/usr/local/nginx/html/hls/mystream0.ts' count:0 ended
[AVIOContext @ 0x564b59591140] Statistics: 0 seeks, 2 writeouts
[segment @ 0x564b59403b80] Opening '/usr/local/nginx/html/hls/mystream1.ts' for writing
[file @ 0x564b5959c900] Setting default whitelist 'file,crypto'
[mpegts @ 0x564b595a1280] muxrate VBR, pcr every 9000 pkts, sdt every 200, pat/pmt every 40 pkts
[segment @ 0x564b59403b80] segment:'/usr/local/nginx/html/hls/mystream1.ts' starts with packet stream:0 pts:191700 pts_time:2.13 frame:40
-c:a aacand share full log.-t 30to your command. It will be a short process whose log you can paste in your Q.