Please guide me to achieve the following result in my program (written in C):
I have a stream source as HTTP MPEG TS stream (codecs h264 & aac), It has 1 video and 1 audio substream.
I need to get MPEG ES frames (of same codecs), to send them via RTP to
RTSP clients. It'll be best if libavformat give frames with RTP
header.
MPEG ES is needed, because, as i know, media players on Blackberry
phones do not play TS (i tried it).
Although, i appreciate if anyone point me some another format, easier to get
in this situation, that can hold h264 & aac, and plays well on
blackberry and other phones.
I've already succeed with other task to open the stream and remux to
FLV container.
Tried to open two output format contexts with "rtp" formats, also got
frames. Sent to client. No success.
I've also tried writing frames to "m4v" AVFormatContext, have got
frames, have cut them by NAL, added RTP header before each frame, and sent to client. Client displays 1st frame and hangs, or plays a second of video+audio (faster than needed) each 10 seconds or more.
In VLC player log i have this: http://pastebin.com/NQ3htvFi
I've scaled timestamps to make them start with 0 for simplicity.
I compared it with what VLC (or Wowza, sorry i dont remember) incremented audio TS by 1024, not 1920, so i did additional linear scaling to be similar to other streamers.
Packet dump of playback of bigbuckbunny_450.mp4 is here:
ftp://rtb.org.ua/tmp/output_my_bbb_450.log
BTW in both cases i've hardly copied SDP from Wowza or VLC.
What is the right way to get what i need?
I'm also interested if there's some library similar to
libavformat? Maybe even in embryo state.
Related
I am receiving audio data in RTP stream. The audio can be either in G711 A-law or u-law depending on the source. How to decode the audio byte stream using ffmpeg api's? Can ALSA on Linux directly play the G711 audio format?
Libav for sure supports G.711. The associated codec ID are AV_CODEC_ID_PCM_MULAW and AV_CODEC_ID_PCM_ALAW. I suggest you start from the example program they provide and modify audio_decode_example() in order to use G.711.
avcodec.h: http://libav.org/doxygen/master/avcodec_8h.html
libav example: http://libav.org/doxygen/master/avcodec_8c-example.html
I have downloaded the MPDs "http://dash.edgesuite.net/adobe/hdworld_dash/HDWorld.mpd" and all related .m4s files.
I tried running it on VLC player. But the format is not recognized by VLC player.
I have downloaded this media segment using wget (1 to 14 segments are available)
http://dash.edgesuite.net/adobe/hdworld_dash/hdworld_seg_hdworld_0696kbps_ffmpeg.mp4.video_temp2.m4s.
Can anybody tell me solution how to run .m4s format file on player?
System: Ubuntu 11.10
You need the initialization segment. It is often named "00" or "init" or doesn't have a sequence number like the other files, and often ends in ".mp4" rather than ".m4s". Then you just concatenate the files together. You can start anywhere in the sequence so long as you begin with the initialization segment.
For example
cat init.mp4 *.m4s > output.mp4
Then you have a playable mp4 file with content, assuming there is no encryption (DRM) applied to it.
.m4s file format is ISO Base Media File. i.e. MPEG-4 Part 14. read specs for more info you may get m4s player for windows. As far as I know on Linux platform GPAC will help. You can create your own MPD from any media source using MP4Box a GPAC tool.
You can use MP4Client for playing your DASHed Media from MPD. Actually .m4s's separate segment is not able to play by its own bcoz player should know Codec and mime type to play any media and m4s is not supported by any player, i.e. it has its own header and data (moof & mdat).
For playing MPD which contains many m4s segment (you can make your own MPD or download each audio and video segment separately from any MPD & put it in to a same folder):
install GPAC.
$MP4Client MYWorld.mpd will open Osmo4 player and you can see your video is playing. Enjoy..
FYI, local streaming server can also play this video:
$MP4Client http://localhost/MYWorld.mpd
if not working change segmentAlignment flag, i.e. <AdaptationSet segmentAlignment="true" subsegmentAlignment="true">.
you can play it using GPAC player, installing it with all the third party codecs also -
http://gpac.wp.mines-telecom.fr/player/
some ppl claim that they are able use vlc, i have not tested it.
Try this in the OSX terminal:
open -a Osmo4 example.mpd
It works for me.
I am transcoding a video using FFMPEG API in c code.
I am trying to set the video bit rate using the ffmpeg API as shown below:
ovCodecCtx->bit_rate = 100 * 1000;
The Encoder I am using is libx264.
But this parameter is not taken into effect and the resulting video quality is very bad.
I have even tried setting related parameters like rc_min_rate, rc_max_rate, etc.. but the video quality is still very low as these related parameters are not taken into effect.
Could any expert tell how one can set the bit rate correctly using the FFMPEG API?
Thanks
I have found the solution to my problem. In fact somebody who was facing the same problem has posted the solution in ffmpeg(libav) user forum. This seems to work in my case too. I am posting the answer to my own question so that other users facing similar issue might benefit from this post.
Problem:
Setting the Video Bit Rate programmatically for the H264 Video Codec was not honoured by the libx264 Codec. Even though it was working for MPEG1, 2 and MPEG4 video codecs, this setting was not recognised for H264 Video Codec. And the resulting video quality was very bad.
Solution:
We need to set the pts for the decoded/resized frames before they are fed to encoder.
The person who found the solution has gone through ffmpeg.c source and was able to figure this out. We need to first rescale the AVFrame's pts from the stream's time_base to the codec time_base to get a simple frame number (e.g. 1, 2, 3).
pic->pts = av_rescale_q(pic->pts, ost->time_base, ovCodecCtx->time_base);
avcodec_encode_video2(ovCodecCtx, &newpkt, pic, &got_packet_ptr);
And when we receive back the encoded packet from the libx264 codec, we need to rescale the pts and dts of the encoded video packet to the stream time base
newpkt.pts = av_rescale_q(newpkt.pts, ovCodecCtx->time_base, ost->time_base);
newpkt.dts = av_rescale_q(newpkt.dts, ovCodecCtx->time_base, ost->time_base);
Thanks
I am implementing the operation of encoding video with TI DM365 mpeg4 encoder and containerizing it with ffmpeg mp4 container using a dummy FMP4 codec to produce headers and footers. While the container is proven to be working correctly using similar Intel based mpeg4 encoder, the dm365 gives a mosaic result if P frames are used at all. Using only I frames works, but I would like to minimize amount of data stored.
The example of the result can be viewed here. Settings are 1-Iframe, 9-Pframes
TI developers didn't answer my question regarding this in 2 days, so I am trying to get help here.
This may help, a TI data sheet on the various settings/parameters and their effect. Apologies if it is telling you stuff you already know...
TI Data Sheet spraba9.pdf
world.
I'm trying to grab rtsp mjpeg stream from IP-camera (realtime) as described in http://www.inb.uni-luebeck.de/~boehme/using_libavcodec.html, but ported to newer version.
It works well with mpeg file (loading it full as one AVPacket), but working with stream, avcodec_decode_video2 returns -1 (error). AVPacket in this case is a part of a frame.
How can I fix this?