Audio transcode problem

Hi,

I am using Wowza Transcoder AddOn and Wowza Media Server 3.5. I have IP Camera that send stream to my Wowza Media Server. This stream have next parameters: video codec - H.264, audio codec - G.771.

I am using transrate.xml template to encode my stream to multiple bitrates - high and low quality.

Problem:

When i am encode only video, without directive, i have very good video quality and framerate. But when I add

CPU not overworked.

Thanks.

wsmaster,

It sounds to me like a problem with the G.711 implementation. I’ve seen such behavior before… some work, some don’t. Try adjusting the audio settings if possible.

You can post the output of ffprobe, and/or describe your stream and device/model in detail so we can identify if a certain implementation exhibits this behavior. Is it a-law or mu-law? G.711.0 or G.711.1? Sample rate and bitrate?

Alternatively, you can transcode the audio portion before the stream gets to Wowza. Some examples here: Frequently requested FFMPEG command examples for Wowza

Hi,

If you are getting SKIP frame then the transcoder does not have enough CPU. Are you using a CUDA card ? as we have seen some perform really poorly when transcoding.

Andrew.

Hi,

I have the same problem with H.264/G.711 transcoding. Audio is pulsate like helicopter.

Here is ffprobe output:

Stream from camera

Input #0, rtsp, from ‘rtsp://root:root@192.230.1.35:554/cam0_0’:

Metadata:

title : RTSP/RTP stream from a VMFD encoder

comment : cam0_0

Duration: N/A, start: 0.000000, bitrate: N/A

Stream #0.0: Video: h264 (High), yuv420p, 640x480, 0.08 tbr, 90k tbn, 180k tbc

Stream #0.1: Audio: pcm_mulaw, 8000 Hz, 1 channels, s16, 64 kb/s

Stream from Wowza(transcoder)

Metadata:

trackinfo:

type video

profile-level-id 4d401f

sprop-parameter-sets Z01AH5ZSAUB7YCoQAAA+kAAOpg4IABEwAACALD8Y4wQACJgAAEAWH4xw7QsXJA==,aOvNSA==

description {H264CodecConfigInfo: codec:H264, profile:Main, level:3.1, frameSize:640x480, displaySize:640x480, frameRate:29.97, PAR:1:1}

type audio

config 1588

description {AACFrame: codec:AAC, channels:1, frequency:8000, samplesPerFrame:1024, objectType:LC}

rtpsessioninfo:

information cam0_0

name RTSP/RTP stream from a VMFD encoder

origin - 1353672041871969 1 IN IP4 192.230.1.35

timing 0 0

protocolversion 0

attributes:

x-qt-text-inf cam0_0

range npt=0-

x-qt-text-nam RTSP/RTP stream from a VMFD encoder

tool LIVE555 Streaming Media v2008.02.08

type broadcast

videocodecid avc1

width 640.00

height 480.00

frameWidth 640.00

frameHeight 480.00

displayWidth 640.00

displayHeight 480.00

framerate 29.97

videodatarate 250.00

audiochannels 1.00

audiosamplerate 8000.00

audiocodecid mp4a

audiodatarate 62.00

transcoder:

audioCodec AAC

audioBitrate 64000.00

audioEncodingParams FALSE

videoCodec H264

videoBitrate 256000.00

videoImplementation DEFAULT

videoProfile MAIN

videoFrameSizeFitMode fit-height

videoFrameSizeWidth 640.00

videoFrameSizeHeight 480.00

videoKeyFrameFollowSourceTRUE

videoEncodingParams FALSE

[flv @ 01C78C60] Estimating duration from bitrate, this may be inaccurate

Input #0, flv, from ‘rtmp://127.0.0.1:1935/rtp-live/definst/rtsp://root:root@192.230.1.35:554/cam0_0_720p’:

Duration: N/A, start: 0.000000, bitrate: N/A

Stream #0.0: Video: h264 (Main), yuv420p, 640x480 [SAR 1:1 DAR 4:3], 30 tbr, 1k tbn, 59.94 tbc

Stream #0.1: Audio: aac, 8000 Hz, mono, s16

wsmaster,

It sounds to me like a problem with the G.711 implementation. I’ve seen such behavior before… some work, some don’t. Try adjusting the audio settings if possible.

You can post the output of ffprobe, and/or describe your stream and device/model in detail so we can identify if a certain implementation exhibits this behavior. Is it a-law or mu-law? G.711.0 or G.711.1? Sample rate and bitrate?

Alternatively, you can transcode the audio portion before the stream gets to Wowza. Some examples here: Frequently requested FFMPEG command examples for Wowza

Thanks for reply.

Ip camera is Samsung SNB-5000. I have live streaming from Wowza Media Server 3.5.

This is ffprobe output from my ip camera stream:

[NULL @ 0xcf76c0]missing picture in access unit
[rtsp @ 0xcf44e0]Estimating duration from bitrate, this may be inaccurate
Input #0, rtsp, from 'rtsp://ipaddress:554/profile/media.smp':
  Metadata:
    title           : Media Presentation
    comment         : samsung
  Duration: N/A, bitrate: N/A
    Stream #0.0: Video: h264, yuv420p, 640x360, 90k tbr, 90k tbn, 90k tbc
    Stream #0.1: Audio: pcm_mulaw, 8000 Hz, 2 channels, s16, 128 kb/s

This is ffprobe output from my Wowza encoded stream:

[NULL @ 0x9820ed0]missing picture in access unit
[rtsp @ 0x981d430]Could not find codec parameters (Audio: aac, mono, s16)
[rtsp @ 0x981d430]Estimating duration from bitrate, this may be inaccurate
Input #0, rtsp, from 'rtsp://ipaddress:1935/live/my.stream_high':
  Metadata:
    title           : my.stream_high
  Duration: N/A, bitrate: N/A
    Stream #0.0: Audio: aac, mono, s16
    Stream #0.1: Video: h264, yuv420p, 640x480, 90k tbr, 90k tbn, 90k tbc

Video dont have lags and have good framerate when audio is turn off, when I add tags to section, my video have lags and skipframe. But audio is ok, good and clear, problem only with video when I turn on audio. May be video and audio live sync and I have skipframe, dont know… Please help me.

CPU load when transcoder is working - 40-50%.

Thanks.