July 2005 - ffmpeg-devel - lists.ffmpeg.org

AVCodecContext no frame_rate
by jeffrey.s.behrbaum＠boeing.com 29 Jul '05

29 Jul '05

Correct you are. Please try to forget that I ever posted this message. I had an old version on my system as well as the new one and I had brought up the wrong file in my editor. Nothing like a stupid mistake to keep me guessing for a couple of hours. Sorry about the lame posting. -----Original Message----- From: M?ns Rullg?rd [mailto:mru at inprovide.com] Sent: Friday, July 29, 2005 11:51 AM To: FFMpeg development discussions and patches Subject: Re: [Ffmpeg-devel] AVCodecContext no frame_rate Behrbaum, Jeffrey S said: > Hi all. I'm getting a compile error I can't seem to figure out. The error > keeps telling me that AVCodecContext has no frame_rate member. See below for > the exact error. Also, please note that the code and make file are simplified. > The values I'm setting are junk, I'm just trying to see why I'm getting this > error. Also, I have the CVS code from 26 July 2005. I'm use Msys and MinGW to > compile this on a Win32 platform. All the other variables of the > AVCodecContext seem to be fine. I can see the frame_rate int in the avcodec.h > file but I have no idea why this compile error keeps creeping up! Thank you > again to any and all for your help. It's simple. There is no member called frame_rate. It was replaced by time_base quite some time ago. -- M?ns Rullg?rd mru at inprovide.com _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel at mplayerhq.hu http://mplayerhq.hu/mailman/listinfo/ffmpeg-devel

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Compile error on ppc/idct_altivec.c
by avarab＠gmail.com 29 Jul '05

29 Jul '05

I'm getting a compile error on ppc/idct_altivec.c while compiling MPlayer with --enable-altivec Steps to reproduce: $ cc -w -std=gnu99 -DHAVE_AV_CONFIG_H -I.. -D_GNU_SOURCE -c -o ppc/idct_altivec.o ppc/idct_altivec.cIn file included from ppc/../dsputil.h:31, from ppc/idct_altivec.c:40: ppc/../common.h:209:36: fastmemcpy.h: No such file or directory ppc/idct_altivec.c:160: error: initializer element is not constant ppc/idct_altivec.c:160: error: (near initialization for `constants[0]') ppc/idct_altivec.c:161: error: initializer element is not constant ppc/idct_altivec.c:161: error: (near initialization for `constants[1]') ppc/idct_altivec.c:162: error: initializer element is not constant ppc/idct_altivec.c:162: error: (near initialization for `constants[2]') ppc/idct_altivec.c:163: error: initializer element is not constant ppc/idct_altivec.c:163: error: (near initialization for `constants[3]') ppc/idct_altivec.c:165: error: initializer element is not constant ppc/idct_altivec.c:165: error: (near initialization for `constants[4]')

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AVCodecContext no frame_rate
by jeffrey.s.behrbaum＠boeing.com 29 Jul '05

29 Jul '05

Hi all. I'm getting a compile error I can't seem to figure out. The error keeps telling me that AVCodecContext has no frame_rate member. See below for the exact error. Also, please note that the code and make file are simplified. The values I'm setting are junk, I'm just trying to see why I'm getting this error. Also, I have the CVS code from 26 July 2005. I'm use Msys and MinGW to compile this on a Win32 platform. All the other variables of the AVCodecContext seem to be fine. I can see the frame_rate int in the avcodec.h file but I have no idea why this compile error keeps creeping up! Thank you again to any and all for your help. ./src/Main.cpp:23: error: 'struct AVCodecContext' has no member named 'frame_rate' Below is my make file: vpath %.cpp ./src vpath %.h ./inc vpath %.o ./obj OBJ_FILE = MAIN : ${OBJ_FILE} g++ ./src/Main.cpp -o ./bin/MpegTest.exe \ -I/ffmpeg/libavcodec -I/ffmpeg/libavformat -L/ffmpeg/libavformat -L/ffmpeg/libavcodec \ -lavcodec -lavformat clean: -rm ./obj/*.o -rm ./bin/MpegTest.exe Below is my Main.cpp //Standard C++ includes #include <iostream> #include <sstream> #include <string> //Local includes #include "avcodec.h" #include "avformat.h" using namespace std; int main(int arg, char **argc) { AVCodecContext pAV_EncoderContext; pAV_EncoderContext.bit_rate = 225; pAV_EncoderContext.bit_rate_tolerance = 10; pAV_EncoderContext.flags = 1; pAV_EncoderContext.sub_id = 9; pAV_EncoderContext.me_method = 10; pAV_EncoderContext.extradata = NULL; pAV_EncoderContext.extradata_size = 100; pAV_EncoderContext.frame_rate = 12;//WHY NOT! pAV_EncoderContext.width = 200; pAV_EncoderContext.height = 400; pAV_EncoderContext.gop_size = 400; // pAV_EncoderContext.PixelFormat = 0; pAV_EncoderContext.rate_emu = 400; pAV_EncoderContext.sample_rate = 30; pAV_EncoderContext.channels = 2; pAV_EncoderContext.frame_size = 400; pAV_EncoderContext.frame_number = 400; return EXIT_SUCCESS; }//End main

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I want to resize getted frame from movie file.
by jcm1981＠gmail.com 29 Jul '05

29 Jul '05

I want to watch fixed display size from movie file. (eg. 640x480) I'm trying to use img_resample function. I don't know what i do before calling img_resample_init function. something following line. if(packet.stream_index == m_nVideoStream) { // decoding next frame. bytesDecoded = avcodec_decode_video(m_pCodecCtx, m_pFrame, &frameFinished, packet.data, packet.size); if(frameFinished) { img_convert((AVPicture *)m_pFrameRGB, PIX_FMT_RGB24, (AVPicture*)m_pFrame, m_pCodecCtx->pix_fmt, m_pCodecCtx->width, m_pCodecCtx->height); // m_pFrame has YUV frame ImgReSampleContext *s; s = img_resample_init( 640, 480, width, height); // initialize resample img_resample(s, m_pic, (AVPicture *)m_pFrameRGB); // error occur img_resample_close(s); Jeon,Chang-Min Thanks.

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I want to resize getting frame.
by jcm1981＠gmail.com 29 Jul '05

29 Jul '05

I want to watch fixed display size from movie file. (eg. 640x480) Could you tell me how to resize RGB frame?

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Adaptive Streaming???
by Ravi_Nallan＠infosys.com 29 Jul '05

29 Jul '05

Does that mean that the media files have to be encoded at different rates and stored at the server? Is there any chance of re-encoding on the fly, I mean can I encode each of the packets that I am going to send at a different rate. I know this would be very time taking for the server, but I would like to know if it is possible. There would lot of information loss in encoding packets separately than encoding the whole file, but can it be done? If the server has to just pick up the right rate stream and pass to client, why does the Video On demand systems need high infrastructure. Coz most of this would be offline task of encoding media into different rates. My questions may seem silly, but I am trying to understand what needs to be done for an effective VoD setup. Thanks, -Ravi -----Original Message----- From: ffmpeg-devel-bounces at mplayerhq.hu [mailto:ffmpeg-devel-bounces at mplayerhq.hu] On Behalf Of Fran?ois Revol Sent: Friday, July 29, 2005 2:28 PM To: FFMpeg development discussions and patches Subject: Re: [Ffmpeg-devel] Adaptive Streaming??? > Hi, > I need some knowledge on what comprises of Adaptive Streaming. > I have been hearing this for quite some time, especially in the > context of > network bandwidth adaptation. > I wanted to know if the file would be re-encoded to a different bit > rate for > network adaptation or a portion of it. > coz, if the whole whole is encoded, as for the network congestion, > the next > stream of packets may need to be encoded in a different rate, hence > the > original encoding may not be useful. Starting from the real "surestream" thingy, afaik the basic way of things is: the source is always encoded at multiple bitrate, and the player asks whichever bitrate it wants depending on the congestion it detected recently. The change probably happens only on keyframes though. The source must always be encoded at all bitrates, since not all players might want the same bitrate at the same time anyway. For static files, they just have several video (and maybe audio) streams, and are hinted for the server. Like quicktime but others including asf do this as well. Fran?ois. _______________________________________________ ffmpeg-devel mailing list ffmpeg-devel at mplayerhq.hu http://mplayerhq.hu/mailman/listinfo/ffmpeg-devel **************** CAUTION - Disclaimer ***************** This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify the sender by e-mail and delete the original message. Further, you are not to copy, disclose, or distribute this e-mail or its contents to any other person and any such actions are unlawful. This e-mail may contain viruses. Infosys has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachment. Infosys reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infosys e-mail system. ***INFOSYS******** End of Disclaimer ********INFOSYS***

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Adaptive Streaming???
by ravi.nallan＠gmail.com 29 Jul '05

29 Jul '05

Hi, I need some knowledge on what comprises of Adaptive Streaming. I have been hearing this for quite some time, especially in the context of network bandwidth adaptation. I wanted to know if the file would be re-encoded to a different bit rate for network adaptation or a portion of it. coz, if the whole whole is encoded, as for the network congestion, the next stream of packets may need to be encoded in a different rate, hence the original encoding may not be useful. Also as one has sent the previous packets ( of the same file) in different rate, and the current packet in different rate, will there be a continuity problem? Also how can one decide which packet to send when the encoding rates have changed? I am confused as how all this fits into place, esp coz all this need to be done in real time and the re-encoding process is quite time taking? I am experimenting with ogg format, and any one has answers specific to this codec? Can one re-encode a signle page instead of re-encoding the complete file?? thanks, -Ravi

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[Patch] Yamaha SMAF file format support
by vidarino＠gmail.com 29 Jul '05

29 Jul '05

Attached is a parser and writer for SMAF files, a format used for ring tones for phones with a Yamaha MA-2/3/5 chip. (http://smaf-yamaha.com/) It's a bit simplistic in that it won't try to parse all the parameters (and many relate to MIDI or soft-synth, which is (and will remain) unhandled). But it should be able to dig its way to the PCM portion (encoded with Yamaha ADPCM). The writing bit should be fairly complete, though, and the output files are almost identical to the ones generated by proprietary converters. Vidar (Patch applies to ffmpeg root, and mmf.c should be dropped under libavformat/)

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How to get the current bitrate?
by lucabe72＠email.it 29 Jul '05

29 Jul '05

Hi all, I need to stream some compressed audio/video over a network with limited bandwidth (there actually is a bandwidth shaper), so I am trying to understand how the rate control in libavcodec works... If I set rc_max_rate and rc_buffer_size, I see that the encoded stream "kind of respects" the maximum rate, but there sometimes are spikes over such limit (I am using MPEG4 video, single-pass encoding). I am trying to understand such spikes and to do this I am currently using tcpstat to monitor the actual bitrate streamed to the network. But I'd like to read the "current" bitrate (not the average one) directly from the codec... To respect the "maximum rate" constraint libavcodec needs some kind of estimation of the current bitrate, right? Is it possible to read it in some way? Also, does anyone know if it's possible to make the rate control more precise/effective by playing with the codec's parameters? (I see rc_buffer_aggressivity, rc_eq, bit_rate_tolerance, ... But I do not know how to use them). Thanks, Luca -- _____________________________________________________________________________ Copy this in your signature, if you think it is important: N O W A R ! ! !

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include_next preprocessor problem
by jeffrey.s.behrbaum＠boeing.com 28 Jul '05

28 Jul '05

Hi all. Sorry to bother you again, but there is something I can't seem to figure out. OK. So I built ffmpeg using MSys and MinGW. Ran configure then make from the parent directory. Compiling a simple program, I include avcodec.h and get the following error: fatal error C1083: Cannot open include file: 'inttypes.h': No such file or directory Then I put the path to the MinGW include directory in my include path so that the inittypes.h required by ffmpeg's common.h is found, and now get the following error: c:\mingw\include\stddef.h(6) : fatal error C1021: invalid preprocessor command 'include_next' I know that the include_next directive is a valid gcc flag, but I was wondering if there is a way around this for a Win32 build? Thanks again to any and all for your time and help. Jeff P.S: I'm building on a Win2K SP2, VC++ 6.0 SP6 //My example code #include <iostream> #include <string> extern "C" { #include "avcodec.h" } using namespace std; int main(int arg, char **argc) { //Setting junk values. Just here for compile testing CodecID eCodecID = CODEC_ID_MPEG2VIDEO_XVMC; CodecType eCodeType = CODEC_TYPE_VIDEO; PixelFormat ePixFormat = PIX_FMT_GRAY8; Motion_Est_ID eMotionEstID = ME_EPZS; RcOverride strctOverRide; strctOverRide.qscale = 0; strctOverRide.end_frame = 0; strctOverRide.start_frame = 0; strctOverRide.quality_factor = 0.0; cout << "FFMEG Practice" << endl; cout << "Now cleanly exiting..." << endl; return EXIT_SUCCESS; }//End main

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