what does VLC expect for RTP/MP3 streaming?

About encoding, codec settings, muxers and filter usage
gonzo1948
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what does VLC expect for RTP/MP3 streaming?

Postby gonzo1948 » 11 Apr 2014 06:14

Not sure if this is the correct forum to post this question. It is about what a VLC stream outputs to a VLC client so I think this is the place. If not, please let me know.

I am working on a server that is streaming audio to the VLC client but I cannot get VLC to process my RTP/MP3 stream. It appears to process the SDP correctly because the log says that it retrieved and opened the SDP file successfully and all parameters are correct, MPGA channels:2 samplerate:44100 bitrate:128, etc.

I think I have a problem with my MP3 encoder. I am using LAME 3.99.5. Here is a debug log edited to remove redundant entries:

Code: Select all

access_http debug: http: server='192.168.1.75' port=9000 file='/mp3.sdp' main debug: net: connecting to 192.168.1.75 port 9000 main debug: connection succeeded (socket = 27) access_http debug: protocol 'HTTP' answer code 200 access_http debug: Server: SimpleHTTP/0.6 Python/2.7.3 access_http debug: Content-Type: application/octet-stream access_http debug: this frame size=147 live555 debug: version 2011.12.23 live555 debug: RTP subsession 'audio/MPA' main debug: selecting program id=0 live555 debug: setup start: 0.000000 stop:0.000000 live555 debug: play start: 0.000000 stop:0.000000 main debug: using demux module "live555" main debug: `http://192.168.1.75:9000/mp3.sdp' successfully opened mpeg_audio debug: MPGA channels:2 samplerate:44100 bitrate:128 main debug: Buffering 0% main debug: reusing audio output pulse debug: using stereo channel map pulse debug: changed buffer metrics: maxlength=4194304, tlength=42336, prebuf=0, minreq=14112 pulse debug: connected to sink alsa_output.pci-0000_00_1b.0.analog-stereo main debug: output 'f32l' 44100 Hz Stereo frame=1 samples/8 bytes main debug: looking for audio volume module matching "any": 2 candidates main debug: using audio volume module "float_mixer" main debug: input 'mpga' 44100 Hz Stereo frame=1152 samples/1053 bytes main debug: looking for audio filter module matching "scaletempo": 14 candidates scaletempo debug: format: 44100 rate, 2 nch, 4 bps, fl32 scaletempo debug: params: 30 stride, 0.200 overlap, 14 search scaletempo debug: 1.000 scale, 1323.000 stride_in, 1323 stride_out, 1059 standing, 264 overlap, 617 search, 2204 queue, fl32 mode main debug: using audio filter module "scaletempo" main debug: conversion: 'mpga'->'f32l' 44100 Hz->44100 Hz Stereo->Stereo main debug: looking for audio converter module matching "any": 12 candidates main debug: no audio converter modules matched main debug: looking for audio converter module matching "any": 12 candidates mpgatofixed32 debug: mpga->f32l, bits per sample: 32 main debug: using audio converter module "mpgatofixed32" main debug: conversion pipeline complete main debug: looking for audio resampler module matching "any": 3 candidates main debug: using audio resampler module "samplerate" main debug: End of audio preroll pulse debug: base volume: 65536 pulse debug: changing sink 0: alsa_output.pci-0000_00_1b.0.analog-stereo (Built-in Audio Analog Stereo) mpeg_audio debug: emulated startcode (no startcode on following frame) mpeg_audio debug: free bitrate mode main debug: Buffering 0% main debug: Buffering 0% mpeg_audio debug: frame too big 977 > 976 (emulated startcode ?) mpeg_audio debug: emulated startcode (no startcode on following frame) mpeg_audio debug: emulated startcode mpeg_audio debug: emulated startcode mpeg_audio debug: emulated startcode mpeg_audio debug: emulated startcode (no startcode on following frame) mpeg_audio debug: emulated startcode mpeg_audio debug: emulated startcode mpeg_audio debug: emulated startcode (no startcode on following frame) main debug: Buffering 0% mpeg_audio debug: emulated startcode (no startcode on following frame) mpeg_audio debug: emulated startcode
Obviously VLC does not like my MP3 data so I have a couple of questions:

Q1: Does VLC expect that the RTP payload will start with the fragmentation 4-byte field as specified by RFC 2250?
Q2: What RTP timestamp increment should I use for MP3 for each RTP pkt? The RTP example in the LAME source tree uses 5 but this seems wrong to me.
Q3: why does the mpeg_audio think a frame of 977 is too large?
Q4: What does "emulated startcode (no startcode on following frame)" mean? I have verified that my RTP payload does indeed have proper MP3 frame headers.

Thanks for any help you can provide.

-Andres

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