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• Correct usage of fdkaac for liquidsoap->icecast

  1er février 2024, par user4893295

  Can't seem to get aac working using fdkaac with liquidsoap. It just seems to shutdown.

  


  TBF I have no idea which aot or transmux to use for "normal" icecast stuff. mp3 works fine, and the icecast server works fine with aac from other sources (BUTT etc).

  


  This is my starting point from the liquidsoap cookbook :

  


  %fdkaac(channels=2, samplerate=44100, bandwidth="auto", bitrate=64, afterburner=false, aot="mpeg2_he_aac_v2", transmux="adts", sbr_mode=false)


  


  And this is my output from liquidsoap :

  


  2024/02/01 20:49:05 [clock:3] Using native (high-precision) implementation for latency control
2024/02/01 20:49:07 [main:3] Standard library loaded in 1.64 seconds.
2024/02/01 20:49:07 [lang:2] WARNING: "set" is deprecated and will be removed in future version. Please use `settings.path.to.key := value`
2024/02/01 20:49:07 [lang:2] WARNING: "set" is deprecated and will be removed in future version. Please use `settings.path.to.key := value`
2024/02/01 20:49:07 [frame:3] Using 44100Hz audio, 25Hz video, 44100Hz main.
2024/02/01 20:49:07 [frame:3] Video frame size set to: 1280x720
2024/02/01 20:49:07 [frame:3] Frame size must be a multiple of 1764 ticks = 1764 audio samples = 1 video samples.
2024/02/01 20:49:07 [frame:3] Targeting 'frame.duration': 0.04s = 1764 audio samples = 1764 ticks.
2024/02/01 20:49:07 [frame:3] Frames last 0.04s = 1764 audio samples = 1 video samples = 1764 ticks.
2024/02/01 20:49:07 [sandbox:3] Could not find binary bwrap, disabling sandboxing.
2024/02/01 20:49:07 [startup:3] DSSI plugins registration: 0.00s
2024/02/01 20:49:07 [startup:3] FFmpeg filters registration: 0.02s
2024/02/01 20:49:07 [startup:3] FFmpeg bitstream filters registration: 0.00s
2024/02/01 20:49:07 [startup:3] Lilv plugins registration: 0.00s
2024/02/01 20:49:07 [startup:3] Frei0r plugin registration: 0.00s
2024/02/01 20:49:07 [startup:3] LADSPA plugins registration: 0.00s
2024/02/01 20:49:07 [startup:3] Typechecking: 1.36s
2024/02/01 20:49:07 [startup:3] Evaluation: 0.01s
2024/02/01 20:49:07 [startup:3] Typechecking: 0.02s
2024/02/01 20:49:07 [startup:3] Evaluation: 0.00s
2024/02/01 20:49:07 [startup:3] Typechecking: 0.00s
2024/02/01 20:49:07 [startup:3] Evaluation: 0.00s
2024/02/01 20:49:07 [startup:3] Typechecking: 0.00s
2024/02/01 20:49:07 [startup:3] Evaluation: 0.08s
2024/02/01 20:49:07 [startup:3] Loaded /etc/liquidsoap/index.liq: 0.08s
2024/02/01 20:49:07 [64_aac:3] Content type is {audio=pcm(stereo)}.
2024/02/01 20:49:07 [cross:3] Content type is {audio=pcm(stereo)}.
2024/02/01 20:49:07 [on_track:3] Content type is {audio=pcm(stereo)}.
2024/02/01 20:49:07 [promos.2:3] Content type is {audio=pcm(stereo)}.
2024/02/01 20:49:07 [promos:3] Content type is {}.
2024/02/01 20:49:07 [current:3] Content type is {audio=pcm(stereo)}.
2024/02/01 20:49:07 [music.3:3] Content type is {audio=pcm(stereo)}.
2024/02/01 20:49:07 [music.2:3] Content type is {}.
2024/02/01 20:49:07 [music:3] Content type is {audio=pcm(stereo)}.
2024/02/01 20:49:07 [safe_blank:3] Content type is {audio=pcm(stereo)}.
2024/02/01 20:49:07 [clock.main:3] Streaming loop starts in auto-sync mode
2024/02/01 20:49:07 [clock.main:3] Delegating synchronization to CPU clock
2024/02/01 20:49:07 [video.converter:3] Using preferred video converter: ffmpeg.
2024/02/01 20:49:07 [audio.converter:3] Using samplerate converter: libsamplerate.
2024/02/01 20:49:07 [video.text:3] Using sdl implementation
2024/02/01 20:49:07 [mksafe:3] Switch to safe_blank.
2024/02/01 20:49:07 [clock.main:2] Source 64_aac failed while streaming: Lang.Runtime_error { kind: "encoder", msg: "Invalid configuration: aot", pos: [at /etc/liquidsoap/index.liq, line 52, char 2-144] }!
2024/02/01 20:49:07 [clock.main:2] Raised at Liquidsoap_lang__Runtime_error.raise in file "src/lang/runtime_error.ml", line 48, characters 14-21
2024/02/01 20:49:07 [clock.main:2] Called from Stdlib__List.iter in file "list.ml", line 110, characters 12-15
2024/02/01 20:49:07 [clock.main:2] Called from Fdkaac_encoder.create_encoder in file "src/core/encoder/encoders/fdkaac_encoder.ml", line 69, characters 2-22
2024/02/01 20:49:07 [clock.main:2] Called from Fdkaac_encoder.encoder in file "src/core/encoder/encoders/fdkaac_encoder.ml", line 73, characters 12-35
2024/02/01 20:49:07 [clock.main:2] Called from Encoder.get_factory.(fun) in file "src/core/encoder/encoder.ml", line 344, characters 8-32
2024/02/01 20:49:07 [clock.main:2] Called from Icecast2.output#icecast_start in file "src/core/outputs/icecast2.ml", line 555, characters 16-67
2024/02/01 20:49:07 [clock.main:2] Called from Icecast2.output#send in file "src/core/outputs/icecast2.ml", line 544, characters 12-30
2024/02/01 20:49:07 [clock.main:2] Called from Output.encoded#send_frame.output_chunks.f in file "src/core/outputs/output.ml", line 256, characters 10-24
2024/02/01 20:49:07 [clock.main:2] Called from Output.output#output in file "src/core/outputs/output.ml", line 192, characters 45-70
2024/02/01 20:49:07 [clock.main:2] Called from Clock.MkClock.clock#end_tick.(fun) in file "src/core/clock.ml", line 318, characters 16-24
Incorrect BOM value
Error reading lyrics, skipped
2024/02/01 20:49:07 [clock.main:2]
[mp3 @ 0x7fd5ad428000] Estimating duration from bitrate, this may be inaccurate
2024/02/01 20:49:07 [decoder.ffmpeg:3] Requested content-type for "/data/dropbox/promos/current/24-7 promo 1 remix.mp3": {audio=pcm(stereo)}
2024/02/01 20:49:07 [decoder.ffmpeg:3] FFmpeg recognizes "/data/dropbox/promos/current/24-7 promo 1 remix.mp3" as audio: {codec: mp3, 44100Hz, 2 channel(s)}
2024/02/01 20:49:07 [decoder.ffmpeg:3] Decoded content-type for "/data/dropbox/promos/current/24-7 promo 1 remix.mp3": {audio=pcm(stereo)}
2024/02/01 20:49:07 [decoder.ffmpeg:3] Requested content-type for "/data/music/Bruno Mars - Locked out of Heaven.mp3": {audio=pcm(stereo)}
2024/02/01 20:49:07 [decoder.ffmpeg:3] FFmpeg recognizes "/data/music/Bruno Mars - Locked out of Heaven.mp3" as audio: {codec: mp3, 44100Hz, 2 channel(s)}, video: {codec: mjpeg, 640x640, yuvj444p}, video_2: {codec: mjpeg, 640x640, yuvj444p}
2024/02/01 20:49:07 [decoder.ffmpeg:3] Decoded content-type for "/data/music/Bruno Mars - Locked out of Heaven.mp3": {audio=pcm(stereo)}
Incorrect BOM value
Error reading lyrics, skipped
[mp3 @ 0x7fd5ad428000] Estimating duration from bitrate, this may be inaccurate
2024/02/01 20:49:07 [clock.main:3] Streaming loop stopped.
2024/02/01 20:49:07 [threads:3] Main loop exited
2024/02/01 20:49:07 [main:3] Shutdown started!
2024/02/01 20:49:07 [threads:3] Waiting for main threads to terminate...
2024/02/01 20:49:07 [threads:3] Main threads terminated.
2024/02/01 20:49:07 [threads:3] Shutting down scheduler...
2024/02/01 20:49:07 [threads:3] Scheduler shut down.
2024/02/01 20:49:07 [main:3] Cleaning downloaded files...
2024/02/01 20:49:07 [main:3] Freeing memory...
2024/02/01 20:49:07 >>> LOG END


  


  EDIT it does seem to be using video, which wouldn't help ? Looking at this bit :

  


  2024/02/01 20:49:07 [frame:3] Using 44100Hz audio, 25Hz video, 44100Hz main.
2024/02/01 20:49:07 [frame:3] Video frame size set to: 1280x720


  


  But no idea why or how to stop that...

  


• How can I publish an RTSP stream to a MediaMTX instance with TLS encryption configured ? [closed]

  15 février 2024, par cmd

  I have an instance of (MediaMTX) running on my laptop. I have an Amcrest IP camera. I can publish the camera's RTSP stream to the server so that the stream can be accessed from the MediaMTX instance.

  


  However, I want to encrypt the stream. On the Github page for MediaMTX, they detail how TLS encryption can be enabled for incoming and outgoing RTSP streams by generating a server.key and server.crt and editing lines in the .yml configuration file. Streams can then be published and read with RTSPS and port 8322.

  


  I have followed these steps, and set up the following path for my IP camera in the .yml :

  


  ###############################################
# Path settings

# Settings in "paths" are applied to specific paths, and the map key
# is the name of the path.
# Any setting in "pathDefaults" can be overridden here.
# It's possible to use regular expressions by using a tilde as prefix,
# for example "~^(test1|test2)$" will match both "test1" and "test2",
# for example "~^prefix" will match all paths that start with "prefix".
paths:

  cam1:
    source: rtsp://user:password@192.168.68.142:554
    runOnInit: ffmpeg -i rtsp://user:password@192.168.68.142:554 -c:v copy -c:a copy -f rtsp rtsps://localhost:8322/stream/cam1


  


  But I am getting the following output from my MediaMTX instance when it runs :

  


  2024/02/15 18:20:59 INF [path cam1] [RTSP source] ready: 2 tracks (H264, MPEG-4 Audio)
Input #0, rtsp, from 'rtsp://user:password@192.168.68.142:554':
  Metadata:
    title           : Media Server
  Duration: N/A, start: 0.030000, bitrate: N/A
  Stream #0:0: Video: h264 (Main), yuv420p(progressive), 1920x1080, 100 tbr, 90k tbn
  Stream #0:1: Audio: aac (LC), 16000 Hz, mono, fltp
Stream mapping:
  Stream #0:0 -> #0:0 (copy)
  Stream #0:1 -> #0:1 (copy)
2024/02/15 18:21:00 INF [RTSPS] [conn [::1]:60147] opened
2024/02/15 18:21:00 INF [RTSPS] [session 07a8f23e] created by [::1]:60147
2024/02/15 18:21:00 INF [RTSPS] [conn [::1]:60147] closed: path 'stream/cam1' is not configured
2024/02/15 18:21:00 INF [RTSPS] [session 07a8f23e] destroyed: not in use
[out#0/rtsp @ 00000225c75fe9c0] Could not write header (incorrect codec parameters ?): Server returned 400 Bad Request
Conversion failed!
2024/02/15 18:21:00 INF [path cam1] runOnInit command exited: command exited with code 3486501640
2024/02/15 18:21:29 WAR [path cam1] [RTSP source] 10 RTP packets lost
2024/02/15 18:21:33 WAR [path cam1] [RTSP source] 2 RTP packets lost
2024/02/15 18:21:39 WAR [path cam1] [RTSP source] 11 RTP packets lost


  


  I have no errors and was able to publish the stream with the same path configuration (using rtsp ://, not rtsps ://) when using no encryption. What possible solutions can I try ?

  


• aacenc_pred : rework the way prediction is done

  29 août 2015, par Rostislav Pehlivanov

  aacenc_pred : rework the way prediction is done
  This commit completely alters the algorithm of prediction.
  
  The original commit which introduced prediction was completely
  
  incorrect to even remotely care about what the actual coefficients
  
  contain or whether any options were enabled. Not my actual fault.
  This commit treats prediction the way the decoder does and expects
  
  to do : like lossy encryption. Everything related to prediction now
  
  happens at the very end but just before quantization and encoding
  
  of coefficients. On the decoder side, prediction happens before
  
  anything has had a chance to even access the coefficients.
  Also the original implementation had problems because it actually
  
  touched the band_type of special bands which already had their
  
  scalefactor indices marked and it’s a wonder the asserion wasn’t
  
  triggered when transmitting those.
  Overall, this now drastically increases audio quality and you should
  
  think about enabling it if you don’t plan on playing anything encoded
  
  on really old low power ultra-embedded devices since they might not
  
  support decoding of prediction or AAC-Main. Though the specifications
  
  were written ages ago and as times change so do the FLOPS.
  Signed-off-by : Rostislav Pehlivanov <atomnuker@gmail.com>
  

  • [DH] libavcodec/aac.h
  • [DH] libavcodec/aaccoder.c
  • [DH] libavcodec/aacenc.c
  • [DH] libavcodec/aacenc.h
  • [DH] libavcodec/aacenc_pred.c
  • [DH] libavcodec/aacenc_pred.h