From shreya_pathak rediffmail.com Fri Oct 1 12:30:10 2004 From: shreya_pathak rediffmail.com (Shreya Pathak) Date: Fri Oct 1 08:26:51 2004 Subject: [Mp4-tech][audio] Mpeg4 AAC: why to skip frame Message-ID: <20041001112949.15147.qmail@webmail28.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041001/3ba31fb9/attachment.html -------------- next part -------------- ? Hi, Thanks for the reply.I was testing my Mpeg4 AAC Decoder and for PNS test vectors the decoded output length is not same as the reference output. I am skipping the first two frames of decoder output to match it with the reference waveform.For all other test vectors the output matches, but for PNS test vectors the output length doesnt match. Please can you suggest me where i am wrong ? Regards Shreya >On Wed, 29 Sep 2023 DDomazet wrote : > >HI, > > > > >Hi, > > >In Mpeg4 AAC decoder publicly available code from Nero i.e FAAC, > > >in file decoder.c > > >if (hDecoder->frame <= 1) > > >hInfo->samples = 0; > > >They are skipping the first frame. > > >I didn't get why they are skipping the first frame. > > > >BECAUSE YOU NEED TWO ENCODED FRAMES > >IN ORDER TO PRODUCE ONE VALID OUTPUT FRAME. > > > >FOR DIAGRAMS BELOW: > >M=1024 > >t - time samples after inverse MDCT transformation > >f - frequencies after MDCT transformation > > > >DECODING: > > > > > > > >ENCODING: > >M=1024 > > > > > > > > >Regards > > >Shreya > > > >HOPE THIS HELPS, > >DANIEL > > >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From trilh Cybersoft-VN.com Fri Oct 1 14:15:40 2004 From: trilh Cybersoft-VN.com (Tri Le Huu) Date: Fri Oct 1 10:05:15 2004 Subject: [Mp4-tech] [M4IF Technotes] source code for H.264 Encoder Message-ID: <99DF6C0285C2CE4C99F02D7838571966F29EAF@hue.cybersoft-vn.com> Dear Mr. Chen, My name is Tri Le. I am from Vietname. Now I am developing a H.264 encoder. Could you give me the source code. It will help me much. Thank you very much, _______________________________ Tri, Le Huu Software Team Global CyberSoft Vietnam Ltd. 123 Truong Dinh street, District 3, HCMC Tel: (+84) (8) 932 1077 (Ext: 523) Fax: (+84) (8) 932 1073 Mob: 090 387 1828 _______________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041001/f5b81f77/attachment.html From shreya_pathak rediffmail.com Fri Oct 1 15:02:23 2004 From: shreya_pathak rediffmail.com (Shreya Pathak) Date: Sun Oct 3 11:43:08 2004 Subject: [Mp4-tech][audio] Mpeg4 AAC: why to skip frame Message-ID: <20041001113140.13186.qmail@webmail32.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041001/54f13a13/attachment.html -------------- next part -------------- ? Hi, Thanks for the reply.I was testing my Mpeg4 AAC Decoder and for PNS test vectors the decoded output length is not same as the reference output. I am skipping the first two frames of decoder output to match it with the reference waveform.For all other test vectors the output matches, but for PNS test vectors the output length doesnt match. Please can you suggest me where i am wrong ? Regards Shreya >On Wed, 29 Sep 2023 DDomazet wrote : > >HI, > > > > >Hi, > > >In Mpeg4 AAC decoder publicly available code from Nero i.e FAAC, > > >in file decoder.c > > >if (hDecoder->frame <= 1) > > >hInfo->samples = 0; > > >They are skipping the first frame. > > >I didn't get why they are skipping the first frame. > > > >BECAUSE YOU NEED TWO ENCODED FRAMES > >IN ORDER TO PRODUCE ONE VALID OUTPUT FRAME. > > > >FOR DIAGRAMS BELOW: > >M=1024 > >t - time samples after inverse MDCT transformation > >f - frequencies after MDCT transformation > > > >DECODING: > > > > > > > >ENCODING: > >M=1024 > > > > > > > > >Regards > > >Shreya > > > >HOPE THIS HELPS, > >DANIEL > > >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From dsn2603 rediffmail.com Fri Oct 1 15:48:46 2004 From: dsn2603 rediffmail.com (sakthi narayanan) Date: Sun Oct 3 11:45:17 2004 Subject: [Mp4-tech] Re:MPEG-4 SA Message-ID: <20041001144830.503.qmail@webmail28.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041001/0ef374bb/attachment.html -------------- next part -------------- ? sir, Can u give some brief intro. abt MPEG-4 SA.Already ,i go thru the documents from the given sites. My doubt is, why they are going for MPEG-4 SA.What way it will be efficient compared to MPEG-4 AAC. Whether any enoding process will be take place in MPEG-4 SA. Kindly give me immediate reply for this mail. From ralph.sperschneider iis.fraunhofer.de Fri Oct 1 18:22:14 2004 From: ralph.sperschneider iis.fraunhofer.de (Ralph Sperschneider) Date: Sun Oct 3 11:47:44 2004 Subject: [Mp4-tech] Re: [Audio] Conformance Test ER-AAC-LC profile In-Reply-To: References: Message-ID: <415D7626.6070509@iis.fraunhofer.de> lisa wrote: > Hi all, > > 1)Could you please explain the conformance test proccedure for the > ER-AAC-LC profile? > > 2)For ER_AL_10 --> RMS procedure is mentioned but there is no such > requirement for the other files? > > 3)How does one conduct the conformance test for the following files > a)ER_AL12 > b)ER_AL15 > c)ER_AL18 > d)ER_AL21 > e)ER_AL23 > f)ER_AL26 > > > With Regards > Lisa Jones Hi Lisa, the standard (ISO/IEC 14496-4) says: " If no test is specified, a check of conformance using appropriate measurements, e.g. the LSB criterion (for those sequences that do not utilize PNS) or objective perceptual measurement systems, is not mandatory but highly recommended. This also applies to bitstreams with non-meaningful window sequences. " So in fact no test is required for ER_AL1[258] and ER_AL2[136], but you should run the RMS/LSB test for all sequences that do not use PNS (that is why PNS is only used at either fs1 or fs2). For ER_AL10 the RMS/LSB test is mandatory. Background: The RMS/LSB test is mandatory only for the sine sweep (ER_AL10), but not for the other signals. Best regards, Ralph -- Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 Fraunhofer IIS | Fax: +49 9131 776 398 Am Wolfsmantel 33 | mailto:ralph.sperschneider@iis.fraunhofer.de D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ From ralph.sperschneider iis.fraunhofer.de Fri Oct 1 18:30:27 2004 From: ralph.sperschneider iis.fraunhofer.de (Ralph Sperschneider) Date: Sun Oct 3 11:49:24 2004 Subject: [Mp4-tech] Re: MPEG 2-AAC vs. MPEG-4 AAC In-Reply-To: References: Message-ID: <415D7813.8060305@iis.fraunhofer.de> S. R. Quackenbush wrote: > MPEG-4 AAC is MPEG-2 AAC > -plus the PNS tool > -plus the MPEG-4 Systems structure of > decoderSpecificInfo > accessUnits > -and can be carried in the MPEG-4 File Format - plus + AAC LPT + AAC scalable + ER AAC LC + ER AAC LTP + ER AAC scalable + ER AAC LD Ralph -- Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 Fraunhofer IIS | Fax: +49 9131 776 398 Am Wolfsmantel 33 | mailto:ralph.sperschneider@iis.fraunhofer.de D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ From rlei ati.com Fri Oct 1 12:35:18 2004 From: rlei ati.com (Ryan Lei) Date: Sun Oct 3 11:50:50 2004 Subject: [Mp4-tech] [system][conference] MPEG4 video conference Message-ID: Hi, All I want to know what is the most commonly used transport solution in industry for MPEG4 video conference? Thanks. Ryan Lei, Ph.D Handheld Products Group | ATI Technologies Inc. | 905.882.2600x2172 | www.ati.com -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041001/9231ce1d/attachment.html From ralph.sperschneider iis.fraunhofer.de Fri Oct 1 19:31:22 2004 From: ralph.sperschneider iis.fraunhofer.de (Ralph Sperschneider) Date: Sun Oct 3 11:51:58 2004 Subject: [Mp4-tech] [Audio] Bitrate from ADTS In-Reply-To: <0908EAA859AF3641B7D9BEDAB8DD6A74094822@dbde2k01.itg.ti.com> References: <0908EAA859AF3641B7D9BEDAB8DD6A74094822@dbde2k01.itg.ti.com> Message-ID: <415D865A.3020407@iis.fraunhofer.de> Mody, Mihir wrote: > Hi Ann, > > As AAC ADTS format doesn't have bitrate information, you have to use other means. The framelength of given audio frame will give you instantaneous bit-rate (as frame_length (in byte) * 8 * sampling frequency/1024). You can average out this bit-rate over certain number (say N) of frames to find out average bitrate. > Or one could use the subsequent information on the bit reservoir status to compute the mean frame length (just not bit exact, since the state of the bit reservoir is rounded). Ralph -- Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 Fraunhofer IIS | Fax: +49 9131 776 398 Am Wolfsmantel 33 | mailto:ralph.sperschneider@iis.fraunhofer.de D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ From harish.vasudeva amd.com Fri Oct 1 18:53:42 2004 From: harish.vasudeva amd.com (harish.vasudeva@amd.com) Date: Sun Oct 3 11:52:41 2004 Subject: [Mp4-tech] Need help with display order.. Message-ID: <2C2A71D85648D343BCCCC36784726266B0592D@SAUSEXMB1.amd.com> Hi, I am trying to figure out a way to get the temporal number in the MPEG4 IF decoder. Is it already present in the header? What I am looking for is : Decode Order : I1 P2 B3 B4.... Display Order : I1 B3 B4 P2.... It would be great if this number is attached to every frame that is currently being decoded. Any help is greatly appreciated. Thanks HV From jjaji2003 yahoo.com Sat Oct 2 10:13:40 2004 From: jjaji2003 yahoo.com (ahmad jalal) Date: Sun Oct 3 11:53:56 2004 Subject: [Mp4-tech] H.263 decoder information Message-ID: <20041002081340.69683.qmail@web41802.mail.yahoo.com> Hello to All, I m searching the some source code whose information is as below: i need the linux source.. Environment:- Operating system:- Linux Programming language:- C++ / C Language (and i will use the Qt/E and Qtopia by Trolltech for presenting the video) Topic:- H.263 Decoder (Codec) for video playback and other related information, If I find the source code for H.263 decoder, the source is execute in PDA. (in pda,) i m waiting for some kind reply thanks Ahmad --------------------------------- Yahoo! Messenger - Communicate instantly..."Ping" your friends today! Download Messenger Now -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041002/28cc54d5/attachment.html From kapo acroid.com Sun Oct 3 19:08:56 2004 From: kapo acroid.com (Eno Kapo) Date: Mon Oct 4 03:50:50 2004 Subject: [Mp4-tech] Re:MPEG-4 SA References: <20041001144830.503.qmail@webmail28.rediffmail.com> Message-ID: <00a801c4a963$4ac7a1b0$0100a8c0@COI> Hi everybody, Anyone had experience with sun streaming server 2.1 using live 3gpp streaming or mpg4 to mobile phones over GPRS or UTMS. any tips related to witch encoder you used are welcome. Thanks in advance Eno Kapo Acroid.com ----- Original Message ----- From: sakthi narayanan To: mp4-tech@lists.mpegif.org Sent: Friday, October 01, 2023 4:48 PM Subject: [Mp4-tech] Re:MPEG-4 SA sir, Can u give some brief intro. abt MPEG-4 SA.Already ,i go thru the documents from the given sites. My doubt is, why they are going for MPEG-4 SA.What way it will be efficient compared to MPEG-4 AAC. Whether any enoding process will be take place in MPEG-4 SA. Kindly give me immediate reply for this mail. ------------------------------------------------------------------------------ _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041003/d1707a42/attachment.html From orho leadtek.com.tw Mon Oct 4 13:47:05 2004 From: orho leadtek.com.tw (=?big5?B?T3JobyCvzqxGpbA=?=) Date: Mon Oct 4 03:52:25 2004 Subject: [Mp4-tech][video][AVC] Rate-Control Message-ID: Dear All, Where I can download the relational JVT discussion documents? I want to know the paper about Rate Control be implemented in JM8x. Orho From garysull windows.microsoft.com Mon Oct 4 02:06:32 2004 From: garysull windows.microsoft.com (Gary Sullivan) Date: Mon Oct 4 04:27:26 2004 Subject: [Mp4-tech][video][AVC] Rate-Control Message-ID: <91D7F2CEE3425A4A9D11311D09FCE2460B3947BF@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com> The JVT ftp site is currently at standards.polycom.com. You are probably looking for documents JVT-H014, JVT-H017, JVT-H021 (these in the 2003_05_Geneva directory), and JVT-K049 (in the 2004_03_Munich directory). Best Regards, Gary Sullivan +> -----Original Message----- +> From: mp4-tech-bounces@lists.mpegif.org +> [mailto:mp4-tech-bounces@lists.mpegif.org] On Behalf Of Orho ??? +> Sent: Sunday, October 03, 2023 9:47 PM +> To: mp4-tech@lists.mpegif.org +> Subject: [Mp4-tech][video][AVC] Rate-Control +> +> Dear All, +> +> Where I can download the relational JVT discussion documents? +> I want to know the paper about Rate Control be implemented in JM8x. +> +> Orho +> +> _______________________________________________ +> NOTE: Please use clear subject lines for your posts. Include +> [audio, [video], [systems], [general] or another +> apppropriate identifier to indicate the type of question you have. +> +> Note: Conduct on the mailing list is subject to the +> Antitrust guidelines found at +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant +> itrust.php +> From bharat.soni st.com Mon Oct 4 16:00:56 2004 From: bharat.soni st.com (Bharat P. SONI) Date: Mon Oct 4 06:36:38 2004 Subject: [Mp4-tech] AVC Teststreams Message-ID: <002f01c4a9f4$d9e0cfc0$9308b40a@dlh.st.com> Hi, Can any one tell me from where I can download the AVC test streams with SEI information for buffer analysis? Regards, Bharat From ralph.sperschneider iis.fraunhofer.de Sat Oct 2 16:37:37 2004 From: ralph.sperschneider iis.fraunhofer.de (Ralph Sperschneider) Date: Mon Oct 4 08:58:48 2004 Subject: [Mp4-tech] [Audio] Bitrate from ADTS In-Reply-To: <23047.144.189.40.221.1096655135.squirrel@webmail.io.com> References: <0908EAA859AF3641B7D9BEDAB8DD6A74094822@dbde2k01.itg.ti.com> <415D865A.3020407@iis.fraunhofer.de> <23047.144.189.40.221.1096655135.squirrel@webmail.io.com> Message-ID: <415EAF21.1050101@iis.fraunhofer.de> athorn@io.com wrote: > How would I do that? Thanks! bit_reservoir_state[frame]=bit_reservoir_state[frame-1]+mean_framelength-framelength[frame] Ralph > > >>Mody, Mihir wrote: >> >> >>>Hi Ann, >>> >>>As AAC ADTS format doesn't have bitrate information, you have to use >>>other means. The framelength of given audio frame will give you >>>instantaneous bit-rate (as frame_length (in byte) * 8 * sampling >>>frequency/1024). You can average out this bit-rate over certain number >>>(say N) of frames to find out average bitrate. >>> >> >>Or one could use the subsequent information on the bit reservoir status to >>compute the mean frame length (just not bit exact, since the state of the >>bit >>reservoir is rounded). >> >>Ralph >> >>-- >>Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 >>Fraunhofer IIS | Fax: +49 9131 776 398 >>Am Wolfsmantel 33 | >>mailto:ralph.sperschneider@iis.fraunhofer.de >>D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ >> > > -- Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 Fraunhofer IIS | Fax: +49 9131 776 398 Am Wolfsmantel 33 | mailto:ralph.sperschneider@iis.fraunhofer.de D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ From ralph.sperschneider iis.fraunhofer.de Mon Oct 4 14:19:28 2004 From: ralph.sperschneider iis.fraunhofer.de (Ralph Sperschneider) Date: Mon Oct 4 08:59:48 2004 Subject: [Mp4-tech] Re: HE AAC conformance bitstream In-Reply-To: <20041004035814.13814.qmail@webmail9.rediffmail.com> References: <20041004035814.13814.qmail@webmail9.rediffmail.com> Message-ID: <416131C0.3090206@iis.fraunhofer.de> sanjay shivkumar mishra wrote: > Dear Relph, > > I am a Ph.D student from IIT Powai Bombay,India.I am working on HE AAC Algorithms and i want to know from where i can get HE AAC (AAC+) Conformance bitstream. > > Bunch of thanks in advance. > > With best regards, > Sanjay Kumar > > > ftp://mpaudconf:adif2mp4@ftp.iis.fraunhofer.de/mpeg4audio-conformance/compressedMp4/sbr_new/ -- Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 Fraunhofer IIS | Fax: +49 9131 776 67 344 Am Wolfsmantel 33 | mailto:ralph.sperschneider@iis.fraunhofer.de D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ From nso01r ecs.soton.ac.uk Mon Oct 4 16:50:12 2004 From: nso01r ecs.soton.ac.uk (Noor Othman) Date: Mon Oct 4 11:21:29 2004 Subject: [Mp4-tech] [audio]Help needed and some questions Message-ID: <41616324.1060404@ecs> Hello everyone, Help needed in compiling the version 1 software of MPEG-4 reference software 2001 edition 14496-5:2001. I have tried compiling using this: make VERSION=1 it have list of errors, of some variables haven't been declared, of which those variables are only declared when VERSION2 is used. Has it been tested i.e. the compilation. Or it's expected to have those errors. Or should I change anything in the makefile My question is the VERSION 2 software (for the particular case of CELP-RPE) automatically being set to Error Resilience Mode? Or is it an option that I can choose. Or if I don't want the Error Resilience mode being chosen, do I then need to use the VERSION 1. Thank you. Regards, Noor From ralph.sperschneider iis.fraunhofer.de Mon Oct 4 20:13:10 2004 From: ralph.sperschneider iis.fraunhofer.de (Ralph Sperschneider) Date: Mon Oct 4 16:20:37 2004 Subject: [Mp4-tech] Re: [audio] Mpeg4 AAC: why to skip frame In-Reply-To: <20041001113140.13186.qmail@webmail32.rediffmail.com> References: <20041001113140.13186.qmail@webmail32.rediffmail.com> Message-ID: <416184A6.3070902@iis.fraunhofer.de> Shreya Pathak wrote: > >Hi, > Thanks for the reply.I was testing my Mpeg4 AAC Decoder and >for PNS test vectors the decoded output length is not same as the reference output. >I am skipping the first two frames of decoder output to match it with the reference waveform.For all other test vectors the output matches, but for PNS test vectors the output length doesnt match. >Please can you suggest me where i am wrong ? >Regards >Shreya > > Shreya, the reference waveforms have been generated using different decoders, at different times of the development state of those decoders. Hence it happens that they behave differently with regard to the number of skipped frames. Ralph -- Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 Fraunhofer IIS | Fax: +49 9131 776 67 344 Am Wolfsmantel 33 | mailto:ralph.sperschneider@iis.fraunhofer.de D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ From harish.vasudeva amd.com Mon Oct 4 16:28:39 2004 From: harish.vasudeva amd.com (harish.vasudeva@amd.com) Date: Mon Oct 4 17:55:29 2004 Subject: [Mp4-tech] B-frame detection? Message-ID: <2C2A71D85648D343BCCCC36784726266B0593C@SAUSEXMB1.amd.com> Hi, Is there any way of detecting (in the decoder) if B-frames are present in the sequence? May be by a combination of bits? Best Regards, HARISH V From praseetha soc-soft.com Tue Oct 5 13:17:26 2004 From: praseetha soc-soft.com (praseetha@soc-soft.com) Date: Tue Oct 5 03:51:23 2004 Subject: [Mp4-tech] h.263 reference code Message-ID: <4BF47D56A0DD2346A1B8D622C5C5902C08FFD8@soc-mail.soc-soft.com> Hi, Can somebody help me how to get the Standard reference code for H.263 Regards, Praseetha The information contained in this e-mail message and in any annexure is confidential to the recipient and may contain privileged information. If you are not the intended recipient, please notify the sender and delete the message along with any annexure. You should not disclose, copy or otherwise use the information contained in the message or any annexure. Any views expressed in this e-mail are those of the individual sender except where the sender specifically states them to be the views of SoCrates Software India Pvt Ltd., Bangalore. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041005/605cab3e/attachment.html From ravimpeg4video yahoo.co.in Tue Oct 5 20:38:04 2004 From: ravimpeg4video yahoo.co.in (ravi kumar) Date: Wed Oct 6 04:11:26 2004 Subject: [Mp4-tech] MPEG 4 LC, SBR licence fee In-Reply-To: <200410051606.i95G5IbC013534@lists1.magma.ca> Message-ID: <20041005183804.80645.qmail@web8507.mail.in.yahoo.com> Hi all Does FADD2 code could be used as the base for the MPEG 4 LC, SBR with out Licence ? If so (a) which version we can use with out GPL licence? (b) Does one should pay licence fee for MPEG 4 LC and SBR regards ravi Yahoo! India Matrimony: Find your life partneronline. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041005/d6511173/attachment.html From Wesley.DeNeve ugent.be Tue Oct 5 21:51:19 2004 From: Wesley.DeNeve ugent.be (Wesley De Neve) Date: Wed Oct 6 04:12:38 2004 Subject: [Mp4-tech] B-frame detection? References: <2C2A71D85648D343BCCCC36784726266B0593C@SAUSEXMB1.amd.com> Message-ID: <006e01c4ab0c$517a18d0$0200a8c0@Persephone> Hi, harish.vasudeva@amd.com wrote: > Hi, > > Is there any way of detecting (in the decoder) if B-frames are > present in the sequence? May be by a combination of bits? The decoder should check for the value of the vop_coding_type syntax element in case you're talking about an MPEG-4 Visual Elementary Stream: vop_coding_type coding method 00 intra-coded (I) 01 predictive-coded (P) 10 bidirectionally-predictive-coded (B) 11 sprite (S) Best regards, -Wesley From harish.vasudeva amd.com Tue Oct 5 14:57:46 2004 From: harish.vasudeva amd.com (Vasudeva, Harish) Date: Wed Oct 6 04:13:22 2004 Subject: [Mp4-tech] B-frame detection? Message-ID: <2C2A71D85648D343BCCCC36784726266B05943@SAUSEXMB1.amd.com> oh, sorry that i forgot to mention that i needed this at the VOL level. I found that i could use the low_delay param to detect the presence of B-Frames. But, next I needed to find the consecutive numbers of B-Frames. From the usermanual.doc (in the reference decoder) I found that the number of Bs within Ps should be only 2. Further, I checked out some other streams that DO NOT have P-Frames (just Is & Bs). In these the number of B-Frames between I-Frames was 9 (in all streams). So, I concluded that : 1. If PVOPs are present, then the number of consecutive BVOPS will be 2. 2. If PVOPs are absent (with only IVOPs), then the number of consecutive BVOPS will be 9. Is that a correct assumption? thanx HV -----Original Message----- From: Wesley De Neve [mailto:Wesley.DeNeve@ugent.be] Sent: Tuesday, October 05, 2023 1:51 PM To: Vasudeva, Harish; Mp4-tech@lists.mpegif.org Subject: Re: [Mp4-tech] B-frame detection? Hi, harish.vasudeva@amd.com wrote: > Hi, > > Is there any way of detecting (in the decoder) if B-frames are > present in the sequence? May be by a combination of bits? The decoder should check for the value of the vop_coding_type syntax element in case you're talking about an MPEG-4 Visual Elementary Stream: vop_coding_type coding method 00 intra-coded (I) 01 predictive-coded (P) 10 bidirectionally-predictive-coded (B) 11 sprite (S) Best regards, -Wesley From Wesley.DeNeve ugent.be Tue Oct 5 22:44:52 2004 From: Wesley.DeNeve ugent.be (Wesley De Neve) Date: Wed Oct 6 04:14:29 2004 Subject: [Mp4-tech] B-frame detection? References: <2C2A71D85648D343BCCCC36784726266B05943@SAUSEXMB1.amd.com> Message-ID: <010701c4ab13$c8360bd0$0200a8c0@Persephone> Hi, Vasudeva, Harish wrote: > oh, sorry that i forgot to mention that i needed this at the VOL > level. I found that i could use the low_delay param to detect the > presence of B-Frames. But, next I needed to find the consecutive > numbers of B-Frames. > > From the usermanual.doc (in the reference decoder) I found that the > number of Bs within Ps should be only 2. Further, I checked out some > other streams that DO NOT have P-Frames (just Is & Bs). In these the > number of B-Frames between I-Frames was 9 (in all streams). So, I > concluded that : > > 1. If PVOPs are present, then the number of consecutive BVOPS will be > 2. > 2. If PVOPs are absent (with only IVOPs), then the number of > consecutive BVOPS will be 9. > > Is that a correct assumption? The low_delay parameter is indeed a good approach for detecting the presence of B-VOPs. You could also have a look at the profile@level information in the VOSH in case you are eager to perform an early detection: the Simple Profile in MPEG-4 Visual does not allow the presence of B-VOPs, while the Advanced Simple Profile does allow them (the profiles in question are the ones most used in current implementations of MPEG-4 Visual). With respect to finding the consecutive number of B-VOPs: I think your assumption is pretty dangerous in case you're looking for a generic solution. MPEG-4 Visual does not fix the GOP structure. As quoted from the spec: "When a video object layer contains coded B-VOPs, the number of consecutive coded B-VOP is variable and unbounded." For instance, don't forget to have a look at the following parameters for the reference encoder (microsoft-v2.4-030710-NTU): Motion.PBetweenICount The number of predicted P-VOPs between each I-VOP. If this value is less than zero, then there will be one I-VOP at the start of the sequence and then all successive frames will be predicted. Motion.BBetweenPCount The number of B-VOPs between each P-VOP. This value is 2 for an IBBPBBPBBP sequence. So, as far as *I* know you will have to look after the value of the vop_encoding_type syntax element in order to be sure about the number of consecutive B-VOPs in case they are present (unless you already have some a priori knowledge about the GOP structure used by the encoder...). Hope this helps, -Wesley From mbakker gmail.com Wed Oct 6 13:43:31 2004 From: mbakker gmail.com (Menno Bakker) Date: Wed Oct 6 14:08:42 2004 Subject: [Mp4-tech] MPEG 4 LC, SBR licence fee In-Reply-To: <20041005183804.80645.qmail@web8507.mail.in.yahoo.com> References: <200410051606.i95G5IbC013534@lists1.magma.ca> <20041005183804.80645.qmail@web8507.mail.in.yahoo.com> Message-ID: > Does FADD2 code could be used as the base for the MPEG 4 LC, SBR with > out Licence ? If so No > (a) which version we can use with out GPL licence? The one you have to pay for. For more info: http://www.audiocoding.com/modules/contact_ahead/ > (b) Does one should pay licence fee for MPEG 4 LC and SBR You mean patent licenses? Yes, you have to pay those too. Menno > > > > regards > > ravi > > Yahoo! India Matrimony: Find your life partner online. > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, > [video], [systems], [general] or another apppropriate identifier to indicate > the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust guidelines > found at > http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > > From mbakker gmail.com Wed Oct 6 13:43:31 2004 From: mbakker gmail.com (Menno Bakker) Date: Wed Oct 6 14:09:40 2004 Subject: [Mp4-tech] MPEG 4 LC, SBR licence fee In-Reply-To: <20041005183804.80645.qmail@web8507.mail.in.yahoo.com> References: <200410051606.i95G5IbC013534@lists1.magma.ca> <20041005183804.80645.qmail@web8507.mail.in.yahoo.com> Message-ID: > Does FADD2 code could be used as the base for the MPEG 4 LC, SBR with > out Licence ? If so No > (a) which version we can use with out GPL licence? The one you have to pay for. For more info: http://www.audiocoding.com/modules/contact_ahead/ > (b) Does one should pay licence fee for MPEG 4 LC and SBR You mean patent licenses? Yes, you have to pay those too. Menno > > > > regards > > ravi > > Yahoo! India Matrimony: Find your life partner online. > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, > [video], [systems], [general] or another apppropriate identifier to indicate > the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust guidelines > found at > http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > > From amith_reddyb yahoo.co.in Wed Oct 6 12:44:00 2004 From: amith_reddyb yahoo.co.in (amith reddy) Date: Wed Oct 6 14:10:18 2004 Subject: [Mp4-tech] mode selection Message-ID: <20041006104400.10647.qmail@web8509.mail.in.yahoo.com> hai , in intra prediction on what criteria we decided 4X4 or 16X16, and also which type of predction is to be selected i,e ver, hor , Dc or -------please help me in this . in h.264 encoder . regards amith --------------------------------- ALL-NEW Yahoo! Messenger - all new features - even more fun! -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041006/2015305e/attachment.html From spsatendra gmail.com Wed Oct 6 17:46:45 2004 From: spsatendra gmail.com (Satendra) Date: Wed Oct 6 14:10:59 2004 Subject: [Mp4-tech] ideal rate control algorithm Message-ID: <5942723204100604167fde941e@mail.gmail.com> hi, I am working on rate control for MPEG-2. I want to know what should be the characteristics of an ideal rate control algorithm? and how can we build that? Thanx From srq ieee.org Wed Oct 6 09:53:37 2004 From: srq ieee.org (S. R. Quackenbush) Date: Wed Oct 6 14:11:45 2004 Subject: [Mp4-tech] MPEG 4 LC, SBR licence fee In-Reply-To: <20041005183804.80645.qmail@web8507.mail.in.yahoo.com> Message-ID: There are licences for copyright, which pertains to code, and patents, which pertain to algorithms and idea. No matter what code you use for MPEG-4 AAC-LC and SBR, you need patent licences (see vialicensing.com). The MPEG reference code (see mpeg.audioresearchlabs.com) can be used for a conforming product without any copyright licenses. Best, Schuyler Quackenbush --- Dr. Schuyler Quackenbush, Audio Research Labs 336 Park Ave, Suite 200, Scotch Plains, NJ 07076 office: 908 490 0700 srq@audioresearchlabs.com mobile: 908 612 9423 fax: 908 842 9151 www.audioresearchlabs.com -----Original Message----- From: mp4-tech-bounces@lists.mpegif.org [mailto:mp4-tech-bounces@lists.mpegif.org]On Behalf Of ravi kumar Sent: Tuesday, October 05, 2023 2:38 PM To: mp4-tech@lists.mpegif.org Subject: [Mp4-tech] MPEG 4 LC, SBR licence fee Hi all Does FADD2 code could be used as the base for the MPEG 4 LC, SBR with out Licence ? If so (a) which version we can use with out GPL licence? (b) Does one should pay licence fee for MPEG 4 LC and SBR regards ravi Yahoo! India Matrimony: Find your life partner online. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041006/57854fbe/attachment.html From vpore pace.stpp.soft.net Thu Oct 7 10:57:50 2004 From: vpore pace.stpp.soft.net (Vinayak Pore) Date: Thu Oct 7 06:44:17 2004 Subject: [Mp4-tech] ideal rate control algorithm In-Reply-To: <5942723204100604167fde941e@mail.gmail.com> References: <5942723204100604167fde941e@mail.gmail.com> Message-ID: <4164C5C6.2080008@pace.stpp.soft.net> - Satisfy VBR or CBR as per the need. - Should allocate bits per frame depending on the complexity of the frame. - Should adapt to the local complexity (at slice level , MB level). - Maintain VBV to avoid any overflows or under flows. - should take care of scene changes. - use MVs or SAD or Var or Hist of Diff. or Diff of Hist as a measure of complexity of the picture. - the most important, visual quality after enc and dec should be good and uniform all over the picture (this is subjective). - and at the end seeing is beliving.(this is subjective). hope this helps. regards, Vinayak. Satendra wrote: >hi, > >I am working on rate control for MPEG-2. I want to know what should >be the characteristics of an ideal rate control algorithm? and how >can we build that? > > >Thanx >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > > > From bharat.soni st.com Thu Oct 7 12:47:50 2004 From: bharat.soni st.com (Bharat P. SONI) Date: Thu Oct 7 06:46:08 2004 Subject: [Mp4-tech] ideal rate control algorithm In-Reply-To: <5942723204100604167fde941e@mail.gmail.com> Message-ID: <002701c4ac35$5fdbeff0$9308b40a@dlh.st.com> Hi Satendra, The ideal rate control algorithm, I would say practically does not exist. It is dependent on the requirements. If you define your requirements and if an algorithm meets your requirement then I would call it an ideal algorithm. There are three possible scenarios, 1 constant bitrate: bitrate is constant through out the sequence (the quality may vary). 2 Variable bitrate: bitrate varies over the sequence based on the video complexity, but the quality is maintained at same level. 3 A mix of CBR and VBR: In this case one can allow to vary the bitrate over small duration of sequence but the overall bitrate is controlled and averaged out to be a constant. In this case the quality is better than CBR. Regards, bharat -----Original Message----- From: mp4-tech-bounces@lists.mpegif.org [mailto:mp4-tech-bounces@lists.mpegif.org] On Behalf Of Satendra Sent: Wednesday, October 06, 2023 4:47 PM To: mp4-tech@lists.mpegif.org Subject: [Mp4-tech] ideal rate control algorithm hi, I am working on rate control for MPEG-2. I want to know what should be the characteristics of an ideal rate control algorithm? and how can we build that? Thanx _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From nishihar cpqd.com.br Thu Oct 7 10:41:17 2004 From: nishihar cpqd.com.br (Ricardo Massahiro Nishihara) Date: Fri Oct 8 03:45:33 2004 Subject: RES: [Mp4-tech][video][AVC] Rate-Control Message-ID: <6B9604063D23374295487CA69A85EBA80148544A@MAILSRV1.aquarius.cpqd.com.br> Hi all, I?m new in this forum. Is JVT ftp a public ftp ? How can I get access to JVT ftp ? Tks in advance, Ricardo -----Mensagem original----- De: mp4-tech-bounces@lists.mpegif.org [mailto:mp4-tech-bounces@lists.mpegif.org]Em nome de Gary Sullivan Enviada em: segunda-feira, 4 de outubro de 2004 05:07 Para: Orho ???; mp4-tech@lists.mpegif.org Assunto: RE: [Mp4-tech][video][AVC] Rate-Control The JVT ftp site is currently at standards.polycom.com. You are probably looking for documents JVT-H014, JVT-H017, JVT-H021 (these in the 2003_05_Geneva directory), and JVT-K049 (in the 2004_03_Munich directory). Best Regards, Gary Sullivan +> -----Original Message----- +> From: mp4-tech-bounces@lists.mpegif.org +> [mailto:mp4-tech-bounces@lists.mpegif.org] On Behalf Of Orho ??? +> Sent: Sunday, October 03, 2023 9:47 PM +> To: mp4-tech@lists.mpegif.org +> Subject: [Mp4-tech][video][AVC] Rate-Control +> +> Dear All, +> +> Where I can download the relational JVT discussion documents? +> I want to know the paper about Rate Control be implemented in JM8x. +> +> Orho +> +> _______________________________________________ +> NOTE: Please use clear subject lines for your posts. Include +> [audio, [video], [systems], [general] or another +> apppropriate identifier to indicate the type of question you have. +> +> Note: Conduct on the mailing list is subject to the +> Antitrust guidelines found at +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant +> itrust.php +> _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From kaustubh.patankar vsnl.net Thu Oct 7 18:50:22 2004 From: kaustubh.patankar vsnl.net (kaustubh.patankar@vsnl.net) Date: Fri Oct 8 03:46:57 2004 Subject: [Mp4-tech] ideal rate control algorithm Message-ID: <10c41bd10c92c5.10c92c510c41bd@vsnl.net> Dear Satendar, I have few inputs. Please check the algorithm to refer for interlaced and progressive sequences. Please refer to TM5 rate control model for inplementation of the algorithms. The rate control can be classified as 1. VBR with may be a specified band for variation where the average bit-rate is important. Ultimately the file size can depend on the average bit rate. 2. the purpose of VBR is to allocate maximum bit rate for the complex sequences 3. There are multi pass VBR implementations also avaialbles. 4. regarding the CBR, the purpose can be to achieve constant bit rate with variation in Q parameter over macro block. I think this may be helpful Kaustubh ----- Original Message ----- From: "Bharat P. SONI" Date: Thursday, October 7, 2023 11:47 am Subject: RE: [Mp4-tech] ideal rate control algorithm > Hi Satendra, > > The ideal rate control algorithm, I would say practically does not > exist. It > is dependent on the requirements. If you define your requirements > and if an > algorithm meets your requirement then I would call it an ideal > algorithm.There are three possible scenarios, > 1 constant bitrate: bitrate is constant through out the sequence (the > quality may vary). > 2 Variable bitrate: bitrate varies over the sequence based on the > videocomplexity, but the quality is maintained at same level. > 3 A mix of CBR and VBR: In this case one can allow to vary the > bitrate over > small duration of sequence but the overall bitrate is controlled and > averaged out to be a constant. In this case the quality is better > than CBR. > > Regards, > bharat > > -----Original Message----- > From: mp4-tech-bounces@lists.mpegif.org > [mp4-tech-bounces@lists.mpegif.org] On Behalf Of Satendra > Sent: Wednesday, October 06, 2023 4:47 PM > To: mp4-tech@lists.mpegif.org > Subject: [Mp4-tech] ideal rate control algorithm > > > hi, > > I am working on rate control for MPEG-2. I want to know what > should be the > characteristics of an ideal rate control algorithm? and how can > we build > that? > > > Thanx > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, > [video], [systems], [general] or another apppropriate identifier > to indicate > the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust > guidelinesfound at > http://www.mpegif.org/public/documents/vault/mp-out-30042- > Antitrust.php > > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include > [audio, [video], [systems], [general] or another apppropriate > identifier to indicate the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust > guidelines found at > http://www.mpegif.org/public/documents/vault/mp-out-30042- > Antitrust.php From raosrr rediffmail.com Thu Oct 7 14:08:55 2004 From: raosrr rediffmail.com (soogoor ravinder rao) Date: Fri Oct 8 03:47:52 2004 Subject: [Mp4-tech] MPEG 4 LC, SBR licence fee Message-ID: <20041007130836.31721.qmail@webmail6.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041007/bb5115a1/attachment.html -------------- next part -------------- Hi All, How to calculate bitrate in Hardware i.e while Targetting to FPGA. If you run the JM software, we will get bitrate According to Totalbits coded, frame rate and numberframes. Bitrate depend on which factors(Hardware side). Thanking you all. with regards, Ravi On Wed, 06 Oct 2023 SR.Quackenbush wrote : >There are licences for copyright, which pertains to code, and patents, which >pertain to algorithms and idea. No matter what code you use for MPEG-4 >AAC-LC and SBR, you need patent licences (see vialicensing.com). The MPEG >reference code (see mpeg.audioresearchlabs.com) can be used for a conforming >product without any copyright licenses. > >Best, >Schuyler Quackenbush >--- >Dr. Schuyler Quackenbush, Audio Research Labs >336 Park Ave, Suite 200, Scotch Plains, NJ 07076 >office: 908 490 0700 srq@audioresearchlabs.com >mobile: 908 612 9423 >fax: 908 842 9151 www.audioresearchlabs.com > > > -----Original Message----- > From: mp4-tech-bounces@lists.mpegif.org >[mailto:mp4-tech-bounces@lists.mpegif.org]On Behalf Of ravi kumar > Sent: Tuesday, October 05, 2023 2:38 PM > To: mp4-tech@lists.mpegif.org > Subject: [Mp4-tech] MPEG 4 LC, SBR licence fee > > > Hi all > > Does FADD2 code could be used as the base for the MPEG 4 LC, SBR >with out Licence ? If so > > (a) which version we can use with out GPL licence? > > (b) Does one should pay licence fee for MPEG 4 LC and SBR > > > > regards > > ravi > > Yahoo! India Matrimony: Find your life partner online. > >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From raosrr rediffmail.com Thu Oct 7 14:09:35 2004 From: raosrr rediffmail.com (soogoor ravinder rao) Date: Fri Oct 8 03:49:03 2004 Subject: [Mp4-tech] bitrate Message-ID: <20041007130918.32531.qmail@webmail6.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041007/002b5ae4/attachment.html -------------- next part -------------- Hi All, How to calculate bitrate in Hardware i.e while Targetting to FPGA. If you run the JM software, we will get bitrate According to Totalbits coded, frame rate and numberframes. Bitrate depend on which factors(Hardware side). Thanking you all. with regards, Ravi On Wed, 06 Oct 2023 SR.Quackenbush wrote : >There are licences for copyright, which pertains to code, and patents, which >pertain to algorithms and idea. No matter what code you use for MPEG-4 >AAC-LC and SBR, you need patent licences (see vialicensing.com). The MPEG >reference code (see mpeg.audioresearchlabs.com) can be used for a conforming >product without any copyright licenses. > >Best, >Schuyler Quackenbush >--- >Dr. Schuyler Quackenbush, Audio Research Labs >336 Park Ave, Suite 200, Scotch Plains, NJ 07076 >office: 908 490 0700 srq@audioresearchlabs.com >mobile: 908 612 9423 >fax: 908 842 9151 www.audioresearchlabs.com > > > -----Original Message----- > From: mp4-tech-bounces@lists.mpegif.org >[mailto:mp4-tech-bounces@lists.mpegif.org]On Behalf Of ravi kumar > Sent: Tuesday, October 05, 2023 2:38 PM > To: mp4-tech@lists.mpegif.org > Subject: [Mp4-tech] MPEG 4 LC, SBR licence fee > > > Hi all > > Does FADD2 code could be used as the base for the MPEG 4 LC, SBR >with out Licence ? If so > > (a) which version we can use with out GPL licence? > > (b) Does one should pay licence fee for MPEG 4 LC and SBR > > > > regards > > ravi > > Yahoo! India Matrimony: Find your life partner online. > >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From gripened gmail.com Fri Oct 8 11:45:26 2004 From: gripened gmail.com (Jayant Chauhan) Date: Fri Oct 8 03:50:26 2004 Subject: [Mp4-tech] MPEG-4 reference code Message-ID: <5b996acb04100722152727e21e@mail.gmail.com> I am trying to get a media player working which supports MP4 format. Have it running now. I am using the reference MP4 parser provided by ISO. I have to implement the Fast forward now. The problem is how do I make my TrackReader jump to the particular Sample in a track. I can get the sample number in the track using the MP4MediaTimeToSampleNUm. Currently, to play the video, we just do a MP4TrackReaderGetNextAccessUnit, as in, traversing a link list. But can we directly jump to a sample number and then traverse from there ??? From veni soc-soft.com Fri Oct 8 11:52:23 2004 From: veni soc-soft.com (veni@soc-soft.com) Date: Fri Oct 8 03:51:42 2004 Subject: [Mp4-tech] MPEG4 Systems.. VisualSampleEntry Message-ID: <4BF47D56A0DD2346A1B8D622C5C5902C09E68E@soc-mail.soc-soft.com> Folks, Can someone tell me about the mystery of VisualSampleEntry. In the specs all fields are reserved with some values(useful one like height width etc like quick time file format). Can anyone give me the exact defination of VisualSampleEntry for mpeg4 video. veni enJOY life The information contained in this e-mail message and in any annexure is confidential to the recipient and may contain privileged information. If you are not the intended recipient, please notify the sender and delete the message along with any annexure. You should not disclose, copy or otherwise use the information contained in the message or any annexure. Any views expressed in this e-mail are those of the individual sender except where the sender specifically states them to be the views of SoCrates Software India Pvt Ltd., Bangalore. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041008/b384c9a1/attachment.html From zhenzhongc hotmail.com Sat Oct 9 02:21:14 2004 From: zhenzhongc hotmail.com (Zhenzhong CHEN) Date: Fri Oct 8 14:37:20 2004 Subject: [Mp4-tech] ideal rate control algorithm References: <10c41bd10c92c5.10c92c510c41bd@vsnl.net> Message-ID: Hi Satendra, My two cents: The fundamental problem in rate control can be stated as follows: min D, subject to: R < R_max This objective can be further specified as either minimizing average distortion (MINAVE) or minimizing maximum distortion (MINMAX) of the video sequence. However, ideal (or optimal) rate control depends on your requirements. >From the bit rate point of view, we can classify the video coding applications into two categories: constant bit rate (CBR) and variable bit rate (VBR). As Bharat have described, the possible scenarios for rate control are: constant bitrate based, variable bitrate based, a mix of CBR and VBR based (sometimes involved in the first scenario). >From the constraint point of view, for end-to-end real-time video communication systems, there is delay constraint to avoid delay jitter and jerky motion. For constant bit-rate applications, buffer constraint is introduced. For storage video applications, the budget constraint should be considered since the storage space is fixed. Different application has different computational complexity requirement. So how can we build the ideal rate control that depends on above requiements. Regards, Zhenzhong CHEN ----- Original Message ----- From: To: "Bharat P. SONI" Cc: Sent: Thursday, October 07, 2023 8:50 PM Subject: Re: RE: [Mp4-tech] ideal rate control algorithm > Dear Satendar, > > I have few inputs. > > Please check the algorithm to refer for interlaced and progressive sequences. Please refer to TM5 rate control model for inplementation of the algorithms. > The rate control can be classified as > 1. VBR with may be a specified band for variation where the average bit-rate is important. Ultimately the file size can depend on the average bit rate. > 2. the purpose of VBR is to allocate maximum bit rate for the complex sequences > 3. There are multi pass VBR implementations also avaialbles. > 4. regarding the CBR, the purpose can be to achieve constant bit rate with variation in Q parameter over macro block. > > I think this may be helpful > > Kaustubh > > ----- Original Message ----- > From: "Bharat P. SONI" > Date: Thursday, October 7, 2023 11:47 am > Subject: RE: [Mp4-tech] ideal rate control algorithm > > > Hi Satendra, > > > > The ideal rate control algorithm, I would say practically does not > > exist. It > > is dependent on the requirements. If you define your requirements > > and if an > > algorithm meets your requirement then I would call it an ideal > > algorithm.There are three possible scenarios, > > 1 constant bitrate: bitrate is constant through out the sequence (the > > quality may vary). > > 2 Variable bitrate: bitrate varies over the sequence based on the > > videocomplexity, but the quality is maintained at same level. > > 3 A mix of CBR and VBR: In this case one can allow to vary the > > bitrate over > > small duration of sequence but the overall bitrate is controlled and > > averaged out to be a constant. In this case the quality is better > > than CBR. > > > > Regards, > > bharat > > > > -----Original Message----- > > From: mp4-tech-bounces@lists.mpegif.org > > [mp4-tech-bounces@lists.mpegif.org] On Behalf Of Satendra > > Sent: Wednesday, October 06, 2023 4:47 PM > > To: mp4-tech@lists.mpegif.org > > Subject: [Mp4-tech] ideal rate control algorithm > > > > > > hi, > > > > I am working on rate control for MPEG-2. I want to know what > > should be the > > characteristics of an ideal rate control algorithm? and how can > > we build > > that? > > > > > > Thanx > > _______________________________________________ > > NOTE: Please use clear subject lines for your posts. Include [audio, > > [video], [systems], [general] or another apppropriate identifier > > to indicate > > the type of question you have. > > > > Note: Conduct on the mailing list is subject to the Antitrust > > guidelinesfound at > > http://www.mpegif.org/public/documents/vault/mp-out-30042- > > Antitrust.php > > > > _______________________________________________ > > NOTE: Please use clear subject lines for your posts. Include > > [audio, [video], [systems], [general] or another apppropriate > > identifier to indicate the type of question you have. > > > > Note: Conduct on the mailing list is subject to the Antitrust > > guidelines found at > > http://www.mpegif.org/public/documents/vault/mp-out-30042- > > Antitrust.php > > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > From cngr rediffmail.com Fri Oct 8 19:04:08 2004 From: cngr rediffmail.com (cng r) Date: Fri Oct 8 14:39:17 2004 Subject: [Mp4-tech] MPEG-4 Variable encoding rate Message-ID: <20041008180349.28081.qmail@webmail28.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041008/b71e8dec/attachment.html -------------- next part -------------- Dear friends, I am doing my B.Tech project and in that I am aiming to do the following. Improving the performance of multimeadia streaming over the internet by estimating the n/w parameters(such as available bandwidth,packet loss..etc). I am planning to vary encoding rate at the sender depending upon the n/w status. And my n/w testbed consists of 2 end hosts and 2 routers(all are Linux configured machines) with NIST Net emulator loded in both routers. My doubts are 1. is is possible to develope dynamic rate encoding at the sender? 2.for doing this, I need to develope a program which will interact with n/w status estimation tools and MPEG4 encoder. So for that, whether I need MPEG4 source code? 3. If it is needed how can I get it? 4. which programming language will be better suited for developing interface module(such as C++, Perl,..etc). please help me. any suggestions regarding project idea will be greatly helpful for me. thanking you all in advance, regards cngr From bir resderm.com Fri Oct 8 15:23:59 2004 From: bir resderm.com (Barry Resnik) Date: Fri Oct 8 16:39:16 2004 Subject: [Mp4-tech] mpg1 to mp4 Message-ID: <011301c4ad63$fbbea780$9865fea9@OBEAR> can anyone give me some help on converting an mpeg1 file to an mpeg4 file for use in iMovie in OS X 10.2.8? Sincerely, Barry -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041008/34ce4cbc/attachment.html From singer apple.com Fri Oct 8 12:27:34 2004 From: singer apple.com (Dave Singer) Date: Fri Oct 8 16:41:12 2004 Subject: [Mp4-tech] MPEG4 Systems.. VisualSampleEntry In-Reply-To: <4BF47D56A0DD2346A1B8D622C5C5902C09E68E@soc-mail.soc-soft.com> References: <4BF47D56A0DD2346A1B8D622C5C5902C09E68E@soc-mail.soc-soft.com> Message-ID: At 10:52 AM +0530 10/8/04, veni@soc-soft.com wrote: >Content-Type: multipart/alternative; > boundary="----_=_NextPart_001_01C4ACF6.CA87CF5B" >content-class: urn:content-classes:message > >Folks, > >Can someone tell me about the mystery of VisualSampleEntry. In the >specs all fields are reserved with some values(useful one like >height width etc like quick time file format). Can anyone give me >the exact defination of VisualSampleEntry for mpeg4 video. > >veni >enJOY life class VisualSampleEntry(codingname) extends SampleEntry (codingname){ unsigned int(16) pre_defined = 0; const unsigned int(16) reserved = 0; unsigned int(32)[3] pre_defined = 0; unsigned int(16) width; unsigned int(16) height; template unsigned int(32) horizresolution = 0x00480000; // 72 dpi template unsigned int(32) vertresolution = 0x00480000; // 72 dpi const unsigned int(32) reserved = 0; template unsigned int(16) frame_count = 1; string[32] compressorname; template unsigned int(16) depth = 0x0018; int(16) pre_defined = -1; } class MP4VisualSampleEntry() extends VisualSampleEntry ('mp4v'){ ESDBox ES; } The fields width and height in the VisualSampleEntry and in the Track Header Box shall be set to the pixel dimensions of the visual stream. The esds box contains an ES Descriptor and so on; the decoder config descriptor is the VOL header etc. for the stream, without any trailing end code sequence. Hope this helps. -- David Singer Apple Computer/QuickTime -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041008/90b94d67/attachment.html From veni soc-soft.com Sat Oct 9 20:19:11 2004 From: veni soc-soft.com (veni@soc-soft.com) Date: Sat Oct 9 17:52:25 2004 Subject: [Mp4-tech] MPEG4 Systems.. VisualSampleEntry Message-ID: <4BF47D56A0DD2346A1B8D622C5C5902C0C5339@soc-mail.soc-soft.com> it helped :) veni enJOY life -----Original Message----- From: Dave Singer [mailto:singer@apple.com] Sent: Friday, October 08, 2023 11:58 PM To: Veni Madhav Soni; mp4-tech@lists.mpegif.org Subject: Re: [Mp4-tech] MPEG4 Systems.. VisualSampleEntry At 10:52 AM +0530 10/8/04, veni@soc-soft.com wrote: Content-Type: multipart/alternative; boundary="----_=_NextPart_001_01C4ACF6.CA87CF5B" content-class: urn:content-classes:message Folks, Can someone tell me about the mystery of VisualSampleEntry. In the specs all fields are reserved with some values(useful one like height width etc like quick time file format). Can anyone give me the exact defination of VisualSampleEntry for mpeg4 video. veni enJOY life class VisualSampleEntry(codingname) extends SampleEntry (codingname){ unsigned int(16) pre_defined = 0; const unsigned int(16) reserved = 0; unsigned int(32)[3] pre_defined = 0; unsigned int(16) width; unsigned int(16) height; template unsigned int(32) horizresolution = 0x00480000; // 72 dpi template unsigned int(32) vertresolution = 0x00480000; // 72 dpi const unsigned int(32) reserved = 0; template unsigned int(16) frame_count = 1; string[32] compressorname; template unsigned int(16) depth = 0x0018; int(16) pre_defined = -1; } class MP4VisualSampleEntry() extends VisualSampleEntry ('mp4v'){ ESDBox ES; } The fields width and height in the VisualSampleEntry and in the Track Header Box shall be set to the pixel dimensions of the visual stream. The esds box contains an ES Descriptor and so on; the decoder config descriptor is the VOL header etc. for the stream, without any trailing end code sequence. Hope this helps. -- David Singer Apple Computer/QuickTime The information contained in this e-mail message and in any annexure is confidential to the recipient and may contain privileged information. If you are not the intended recipient, please notify the sender and delete the message along with any annexure. You should not disclose, copy or otherwise use the information contained in the message or any annexure. Any views expressed in this e-mail are those of the individual sender except where the sender specifically states them to be the views of SoCrates Software India Pvt Ltd., Bangalore. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041009/8bd1971d/attachment.html From garysull windows.microsoft.com Sat Oct 9 13:47:44 2004 From: garysull windows.microsoft.com (Gary Sullivan) Date: Sat Oct 9 17:54:43 2004 Subject: [Mp4-tech][video][AVC] Rate-Control Message-ID: <91D7F2CEE3425A4A9D11311D09FCE2460B5244A2@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com> Yes. Just use your favorite ftp tool to access the site anonymously. If you have trouble accessing the site using your favorite tool, try using your second-favorite one (we're have a few glitches in the site set-up at the moment). Best Regards, Gary Sullivan +> -----Original Message----- +> From: mp4-tech-bounces@lists.mpegif.org +> [mailto:mp4-tech-bounces@lists.mpegif.org] On Behalf Of +> Ricardo Massahiro Nishihara +> Sent: Thursday, October 07, 2023 5:41 AM +> To: Gary Sullivan; Orho ???; mp4-tech@lists.mpegif.org +> Subject: RES: [Mp4-tech][video][AVC] Rate-Control +> +> Hi all, +> I?m new in this forum. Is JVT ftp a public ftp ? +> How can I get access to JVT ftp ? +> Tks in advance, +> Ricardo +> +> -----Mensagem original----- +> De: mp4-tech-bounces@lists.mpegif.org +> [mailto:mp4-tech-bounces@lists.mpegif.org]Em nome de Gary Sullivan +> Enviada em: segunda-feira, 4 de outubro de 2004 05:07 +> Para: Orho ???; mp4-tech@lists.mpegif.org +> Assunto: RE: [Mp4-tech][video][AVC] Rate-Control +> +> +> +> The JVT ftp site is currently at standards.polycom.com. +> +> You are probably looking for documents JVT-H014, JVT-H017, +> JVT-H021 (these in the 2003_05_Geneva directory), and +> JVT-K049 (in the 2004_03_Munich directory). +> +> Best Regards, +> +> Gary Sullivan +> +> +> -----Original Message----- +> +> From: mp4-tech-bounces@lists.mpegif.org +> +> [mailto:mp4-tech-bounces@lists.mpegif.org] On Behalf Of Orho ??? +> +> Sent: Sunday, October 03, 2023 9:47 PM +> +> To: mp4-tech@lists.mpegif.org +> +> Subject: [Mp4-tech][video][AVC] Rate-Control +> +> +> +> Dear All, +> +> +> +> Where I can download the relational JVT discussion documents? +> +> I want to know the paper about Rate Control be +> implemented in JM8x. +> +> +> +> Orho +> +> +> +> _______________________________________________ +> +> NOTE: Please use clear subject lines for your posts. Include +> +> [audio, [video], [systems], [general] or another +> +> apppropriate identifier to indicate the type of question you have. +> +> +> +> Note: Conduct on the mailing list is subject to the +> +> Antitrust guidelines found at +> +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant +> +> itrust.php +> +> +> +> _______________________________________________ +> NOTE: Please use clear subject lines for your posts. Include +> [audio, [video], [systems], [general] or another +> apppropriate identifier to indicate the type of question you have. +> +> Note: Conduct on the mailing list is subject to the +> Antitrust guidelines found at +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant +> itrust.php +> +> _______________________________________________ +> NOTE: Please use clear subject lines for your posts. Include +> [audio, [video], [systems], [general] or another +> apppropriate identifier to indicate the type of question you have. +> +> Note: Conduct on the mailing list is subject to the +> Antitrust guidelines found at +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant +> itrust.php +> From garysull windows.microsoft.com Sat Oct 9 13:47:40 2004 From: garysull windows.microsoft.com (Gary Sullivan) Date: Sat Oct 9 17:56:25 2004 Subject: [Mp4-tech] ideal rate control algorithm Message-ID: <91D7F2CEE3425A4A9D11311D09FCE2460B5244A0@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com> There is some confusion in these replies regarding the definition of CBR. CBR does not mean that the number of bits used is kept constant for every picture or for every macroblock. In fact, there can be very large variations in bit usage from picture-to-picture in CBR operation, or even for significanly-long sequences of pictures. Instead, CBR operation means that the rate of the flow of bits into the input buffer of the (hypothetical) decoder is constant. When that buffer is large, there can be large variations in bit usage from picture-to-picture while relying on the buffer capacity to smooth over those variations. Best Regards, Gary Sullivan +> -----Original Message----- +> From: mp4-tech-bounces@lists.mpegif.org +> [mailto:mp4-tech-bounces@lists.mpegif.org] On Behalf Of +> kaustubh.patankar@vsnl.net +> Sent: Thursday, October 07, 2023 5:50 AM +> To: Bharat P. SONI +> Cc: mp4-tech@lists.mpegif.org +> Subject: Re: RE: [Mp4-tech] ideal rate control algorithm +> +> Dear Satendar, +> +> I have few inputs. +> +> Please check the algorithm to refer for interlaced and +> progressive sequences. Please refer to TM5 rate control +> model for inplementation of the algorithms. +> The rate control can be classified as +> 1. VBR with may be a specified band for variation where the +> average bit-rate is important. Ultimately the file size can +> depend on the average bit rate. +> 2. the purpose of VBR is to allocate maximum bit rate for +> the complex sequences +> 3. There are multi pass VBR implementations also avaialbles. +> 4. regarding the CBR, the purpose can be to achieve constant +> bit rate with variation in Q parameter over macro block. +> +> I think this may be helpful +> +> Kaustubh +> +> ----- Original Message ----- +> From: "Bharat P. SONI" +> Date: Thursday, October 7, 2023 11:47 am +> Subject: RE: [Mp4-tech] ideal rate control algorithm +> +> > Hi Satendra, +> > +> > The ideal rate control algorithm, I would say practically does not +> > exist. It +> > is dependent on the requirements. If you define your requirements +> > and if an +> > algorithm meets your requirement then I would call it an ideal +> > algorithm.There are three possible scenarios, +> > 1 constant bitrate: bitrate is constant through out the +> sequence (the +> > quality may vary). +> > 2 Variable bitrate: bitrate varies over the sequence based on the +> > videocomplexity, but the quality is maintained at same level. +> > 3 A mix of CBR and VBR: In this case one can allow to vary the +> > bitrate over +> > small duration of sequence but the overall bitrate is +> controlled and +> > averaged out to be a constant. In this case the quality is better +> > than CBR. +> > +> > Regards, +> > bharat +> > +> > -----Original Message----- +> > From: mp4-tech-bounces@lists.mpegif.org +> > [mp4-tech-bounces@lists.mpegif.org] On Behalf Of Satendra +> > Sent: Wednesday, October 06, 2023 4:47 PM +> > To: mp4-tech@lists.mpegif.org +> > Subject: [Mp4-tech] ideal rate control algorithm +> > +> > +> > hi, +> > +> > I am working on rate control for MPEG-2. I want to know what +> > should be the +> > characteristics of an ideal rate control algorithm? and how can +> > we build +> > that? +> > +> > +> > Thanx +> > _______________________________________________ +> > NOTE: Please use clear subject lines for your posts. +> Include [audio, +> > [video], [systems], [general] or another apppropriate identifier +> > to indicate +> > the type of question you have. +> > +> > Note: Conduct on the mailing list is subject to the Antitrust +> > guidelinesfound at +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- +> > Antitrust.php +> > +> > _______________________________________________ +> > NOTE: Please use clear subject lines for your posts. Include +> > [audio, [video], [systems], [general] or another apppropriate +> > identifier to indicate the type of question you have. +> > +> > Note: Conduct on the mailing list is subject to the Antitrust +> > guidelines found at +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- +> > Antitrust.php +> +> _______________________________________________ +> NOTE: Please use clear subject lines for your posts. Include +> [audio, [video], [systems], [general] or another +> apppropriate identifier to indicate the type of question you have. +> +> Note: Conduct on the mailing list is subject to the +> Antitrust guidelines found at +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant +> itrust.php +> From kaustubh.patankar vsnl.net Sun Oct 10 19:20:44 2004 From: kaustubh.patankar vsnl.net (kaustubh.patankar@vsnl.net) Date: Sun Oct 10 17:18:58 2004 Subject: [Mp4-tech] ideal rate control algorithm Message-ID: <12e6d6012e753d.12e753d12e6d60@vsnl.net> Dear Gary, I have one question, The rate to the Hypo. decoder should remain constant over picture or complete seq or over the time. Also my answer to the CBR was the the Q parameter needs to be variaed, if required over macro block / picture. Also in that case what is the precise difference between CBR and VBR with regards Kaustubh ----- Original Message ----- From: Gary Sullivan Date: Sunday, October 10, 2023 1:17 am Subject: RE: RE: [Mp4-tech] ideal rate control algorithm > > There is some confusion in these replies regarding the definition of > CBR. > > CBR does not mean that the number of bits used is kept constant for > every picture or for every macroblock. > In fact, there can be very large variations in bit usage from > picture-to-picture in CBR operation, or even for significanly-long > sequences of pictures. > > Instead, CBR operation means that the rate of the flow of bits > into the > input buffer of the (hypothetical) decoder is constant. When that > buffer is large, there can be large variations in bit usage from > picture-to-picture while relying on the buffer capacity to smooth over > those variations. > > Best Regards, > > Gary Sullivan > > +> -----Original Message----- > +> From: mp4-tech-bounces@lists.mpegif.org > +> [mp4-tech-bounces@lists.mpegif.org] On Behalf Of > +> kaustubh.patankar@vsnl.net > +> Sent: Thursday, October 07, 2023 5:50 AM > +> To: Bharat P. SONI > +> Cc: mp4-tech@lists.mpegif.org > +> Subject: Re: RE: [Mp4-tech] ideal rate control algorithm > +> > +> Dear Satendar, > +> > +> I have few inputs. > +> > +> Please check the algorithm to refer for interlaced and > +> progressive sequences. Please refer to TM5 rate control > +> model for inplementation of the algorithms. > +> The rate control can be classified as > +> 1. VBR with may be a specified band for variation where the > +> average bit-rate is important. Ultimately the file size can > +> depend on the average bit rate. > +> 2. the purpose of VBR is to allocate maximum bit rate for > +> the complex sequences > +> 3. There are multi pass VBR implementations also avaialbles. > +> 4. regarding the CBR, the purpose can be to achieve constant > +> bit rate with variation in Q parameter over macro block. > +> > +> I think this may be helpful > +> > +> Kaustubh > +> > +> ----- Original Message ----- > +> From: "Bharat P. SONI" > +> Date: Thursday, October 7, 2023 11:47 am > +> Subject: RE: [Mp4-tech] ideal rate control algorithm > +> > +> > Hi Satendra, > +> > > +> > The ideal rate control algorithm, I would say practically > does not > +> > exist. It > +> > is dependent on the requirements. If you define your > requirements > +> > and if an > +> > algorithm meets your requirement then I would call it an > ideal > +> > algorithm.There are three possible scenarios, > +> > 1 constant bitrate: bitrate is constant through out the > +> sequence (the > +> > quality may vary). > +> > 2 Variable bitrate: bitrate varies over the sequence based on > the > +> > videocomplexity, but the quality is maintained at same level. > +> > 3 A mix of CBR and VBR: In this case one can allow to vary > the > +> > bitrate over > +> > small duration of sequence but the overall bitrate is > +> controlled and > +> > averaged out to be a constant. In this case the quality is > better > +> > than CBR. > +> > > +> > Regards, > +> > bharat > +> > > +> > -----Original Message----- > +> > From: mp4-tech-bounces@lists.mpegif.org > +> > [mp4-tech-bounces@lists.mpegif.org] On Behalf Of Satendra > +> > Sent: Wednesday, October 06, 2023 4:47 PM > +> > To: mp4-tech@lists.mpegif.org > +> > Subject: [Mp4-tech] ideal rate control algorithm > +> > > +> > > +> > hi, > +> > > +> > I am working on rate control for MPEG-2. I want to know what > +> > should be the > +> > characteristics of an ideal rate control algorithm? and how > can > +> > we build > +> > that? > +> > > +> > > +> > Thanx > +> > _______________________________________________ > +> > NOTE: Please use clear subject lines for your posts. > +> Include [audio, > +> > [video], [systems], [general] or another apppropriate > identifier > +> > to indicate > +> > the type of question you have. > +> > > +> > Note: Conduct on the mailing list is subject to the Antitrust > +> > guidelinesfound at > +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- > +> > Antitrust.php > +> > > +> > _______________________________________________ > +> > NOTE: Please use clear subject lines for your posts. Include > +> > [audio, [video], [systems], [general] or another apppropriate > +> > identifier to indicate the type of question you have. > +> > > +> > Note: Conduct on the mailing list is subject to the Antitrust > +> > guidelines found at > +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- > +> > Antitrust.php > +> > +> _______________________________________________ > +> NOTE: Please use clear subject lines for your posts. Include > +> [audio, [video], [systems], [general] or another > +> apppropriate identifier to indicate the type of question you have. > +> > +> Note: Conduct on the mailing list is subject to the > +> Antitrust guidelines found at > +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant > +> itrust.php > +> > > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include > [audio, [video], [systems], [general] or another apppropriate > identifier to indicate the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust > guidelines found at > http://www.mpegif.org/public/documents/vault/mp-out-30042- > Antitrust.php From garysull windows.microsoft.com Sun Oct 10 10:16:40 2004 From: garysull windows.microsoft.com (Gary Sullivan) Date: Sun Oct 10 17:21:07 2004 Subject: [Mp4-tech] ideal rate control algorithm Message-ID: <91D7F2CEE3425A4A9D11311D09FCE2460B524556@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com> In CBR operation, the rate of bits flowing into the input buffer of the hypothetical decoder is constant for the duration of the coded video sequence. In VBR operation, it is not necessarily constant. Yes, the quantization fidelity probably needs to be adjusted once in a while to maintain CBR operation. However, it doesn't have to be very often if the input buffer is sufficiently large. The number of bits per macroblock or per picture can vary a lot on a local basis while still maintaining CBR operation. The size of the required input buffer is specified in the standard. It can be used to smooth out local variations, which can be considerable in magnitude. Best Regards, Gary Sullivan +> -----Original Message----- +> From: kaustubh.patankar@vsnl.net [mailto:kaustubh.patankar@vsnl.net] +> Sent: Sunday, October 10, 2023 6:21 AM +> To: Gary Sullivan +> Cc: Bharat P. SONI; mp4-tech@lists.mpegif.org +> Subject: Re: RE: RE: [Mp4-tech] ideal rate control algorithm +> +> Dear Gary, +> +> I have one question, +> +> The rate to the Hypo. decoder should remain constant over +> picture or complete seq or over the time. +> +> Also my answer to the CBR was the the Q parameter needs to +> be variaed, if required over macro block / picture. +> +> Also in that case what is the precise difference between CBR and VBR +> +> with regards +> +> Kaustubh +> +> ----- Original Message ----- +> From: Gary Sullivan +> Date: Sunday, October 10, 2023 1:17 am +> Subject: RE: RE: [Mp4-tech] ideal rate control algorithm +> +> > +> > There is some confusion in these replies regarding the +> definition of +> > CBR. +> > +> > CBR does not mean that the number of bits used is kept constant for +> > every picture or for every macroblock. +> > In fact, there can be very large variations in bit usage from +> > picture-to-picture in CBR operation, or even for significanly-long +> > sequences of pictures. +> > +> > Instead, CBR operation means that the rate of the flow of bits +> > into the +> > input buffer of the (hypothetical) decoder is constant. When that +> > buffer is large, there can be large variations in bit usage from +> > picture-to-picture while relying on the buffer capacity to +> smooth over +> > those variations. +> > +> > Best Regards, +> > +> > Gary Sullivan +> > +> > +> -----Original Message----- +> > +> From: mp4-tech-bounces@lists.mpegif.org +> > +> [mp4-tech-bounces@lists.mpegif.org] On Behalf Of +> > +> kaustubh.patankar@vsnl.net +> > +> Sent: Thursday, October 07, 2023 5:50 AM +> > +> To: Bharat P. SONI +> > +> Cc: mp4-tech@lists.mpegif.org +> > +> Subject: Re: RE: [Mp4-tech] ideal rate control algorithm +> > +> +> > +> Dear Satendar, +> > +> +> > +> I have few inputs. +> > +> +> > +> Please check the algorithm to refer for interlaced and +> > +> progressive sequences. Please refer to TM5 rate control +> > +> model for inplementation of the algorithms. +> > +> The rate control can be classified as +> > +> 1. VBR with may be a specified band for variation where the +> > +> average bit-rate is important. Ultimately the file size can +> > +> depend on the average bit rate. +> > +> 2. the purpose of VBR is to allocate maximum bit rate for +> > +> the complex sequences +> > +> 3. There are multi pass VBR implementations also avaialbles. +> > +> 4. regarding the CBR, the purpose can be to achieve constant +> > +> bit rate with variation in Q parameter over macro block. +> > +> +> > +> I think this may be helpful +> > +> +> > +> Kaustubh +> > +> +> > +> ----- Original Message ----- +> > +> From: "Bharat P. SONI" +> > +> Date: Thursday, October 7, 2023 11:47 am +> > +> Subject: RE: [Mp4-tech] ideal rate control algorithm +> > +> +> > +> > Hi Satendra, +> > +> > +> > +> > The ideal rate control algorithm, I would say practically +> > does not +> > +> > exist. It +> > +> > is dependent on the requirements. If you define your +> > requirements +> > +> > and if an +> > +> > algorithm meets your requirement then I would call it an +> > ideal +> > +> > algorithm.There are three possible scenarios, +> > +> > 1 constant bitrate: bitrate is constant through out the +> > +> sequence (the +> > +> > quality may vary). +> > +> > 2 Variable bitrate: bitrate varies over the sequence based on +> > the +> > +> > videocomplexity, but the quality is maintained at same level. +> > +> > 3 A mix of CBR and VBR: In this case one can allow to vary +> > the +> > +> > bitrate over +> > +> > small duration of sequence but the overall bitrate is +> > +> controlled and +> > +> > averaged out to be a constant. In this case the quality is +> > better +> > +> > than CBR. +> > +> > +> > +> > Regards, +> > +> > bharat +> > +> > +> > +> > -----Original Message----- +> > +> > From: mp4-tech-bounces@lists.mpegif.org +> > +> > [mp4-tech-bounces@lists.mpegif.org] On Behalf Of Satendra +> > +> > Sent: Wednesday, October 06, 2023 4:47 PM +> > +> > To: mp4-tech@lists.mpegif.org +> > +> > Subject: [Mp4-tech] ideal rate control algorithm +> > +> > +> > +> > +> > +> > hi, +> > +> > +> > +> > I am working on rate control for MPEG-2. I want to know what +> > +> > should be the +> > +> > characteristics of an ideal rate control algorithm? and how +> > can +> > +> > we build +> > +> > that? +> > +> > +> > +> > +> > +> > Thanx +> > +> > _______________________________________________ +> > +> > NOTE: Please use clear subject lines for your posts. +> > +> Include [audio, +> > +> > [video], [systems], [general] or another apppropriate +> > identifier +> > +> > to indicate +> > +> > the type of question you have. +> > +> > +> > +> > Note: Conduct on the mailing list is subject to the Antitrust +> > +> > guidelinesfound at +> > +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- +> > +> > Antitrust.php +> > +> > +> > +> > _______________________________________________ +> > +> > NOTE: Please use clear subject lines for your posts. Include +> > +> > [audio, [video], [systems], [general] or another apppropriate +> > +> > identifier to indicate the type of question you have. +> > +> > +> > +> > Note: Conduct on the mailing list is subject to the Antitrust +> > +> > guidelines found at +> > +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- +> > +> > Antitrust.php +> > +> +> > +> _______________________________________________ +> > +> NOTE: Please use clear subject lines for your posts. Include +> > +> [audio, [video], [systems], [general] or another +> > +> apppropriate identifier to indicate the type of +> question you have. +> > +> +> > +> Note: Conduct on the mailing list is subject to the +> > +> Antitrust guidelines found at +> > +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant +> > +> itrust.php +> > +> +> > +> > _______________________________________________ +> > NOTE: Please use clear subject lines for your posts. Include +> > [audio, [video], [systems], [general] or another apppropriate +> > identifier to indicate the type of question you have. +> > +> > Note: Conduct on the mailing list is subject to the Antitrust +> > guidelines found at +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- +> > Antitrust.php +> +> From benjamin.rowe studentmail.newcastle.edu.au Mon Oct 11 11:09:39 2004 From: benjamin.rowe studentmail.newcastle.edu.au (BENJAMIN DLUZEWSKA ROWE) Date: Mon Oct 11 18:22:14 2004 Subject: [Mp4-tech] Extracting BIFS Data Message-ID: <143e80147901.147901143e80@studentmail.newcastle.edu.au> Hi, I am attempting to extract BIFS data from an MP4 file. Can someone tell me where in the file the data exists? From what I understand it lives in the trak atom but I'm not entirely sure. Regards, Ben From dattagurubn yahoo.com Sun Oct 10 23:43:38 2004 From: dattagurubn yahoo.com (Dattaguru B.N.) Date: Mon Oct 11 18:23:53 2004 Subject: [Mp4-tech] [Audio] Querries on Integer implementation of SBR Decoder In-Reply-To: <91D7F2CEE3425A4A9D11311D09FCE2460B524556@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com> Message-ID: <20041011054338.51883.qmail@web20103.mail.yahoo.com> Dear All, I want to know some information on integer implementation of SBR decoder. For the QMF analysis filter, input is the output of the windowing ( output of AAC ). Integer implementation of AAC decoder will have an error upto 2 bits (output of windowing) or may be more also!!!. As QMF analysis filter consists of lots of cumulative additions and multiplications, the small difference in the input ( output from windowing) will result in large error at the output of the analysis filter. Hence, there will be a huge difference at the output. Could you please let me know 1. Is there any specific RMS/LSB error values specified for integer implementation of SBR decoder by Coding technologies? 2. Is there any specific implementation for integer? Thanks and regards, Dattaguru _______________________________ Do you Yahoo!? Declare Yourself - Register online to vote today! http://vote.yahoo.com From bharat.soni st.com Mon Oct 11 13:45:09 2004 From: bharat.soni st.com (Bharat P. SONI) Date: Mon Oct 11 18:25:42 2004 Subject: [Mp4-tech] ideal rate control algorithm In-Reply-To: <91D7F2CEE3425A4A9D11311D09FCE2460B524556@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com> Message-ID: <001b01c4af62$0ae25f40$9308b40a@dlh.st.com> Dear Gary, You said "the rate of bits flowing into the input buffer of the hypothetical decoder is constant". I fully agree with this statement. But size of this decoding buffer is mainly useful to control overflow/underflow condition. I think the size of this buffer has little role on controlling the bitrate (in case of CBR)if it is sufficiently high. The bitrate is controlled based on the Target bits available for encoding the remaining pictures in an encoding segment (say a GOP) over which bitrate is being controlled/monitored. Keeping this input buffer quite large (specifically for CBR) may not be a good solution as this buffer may not underflow, but it is possible that quality will fluctuate drastically in case of fast moving sequence. Correct me if I am wrong. Regards, Bharat -----Original Message----- From: Gary Sullivan [mailto:garysull@windows.microsoft.com] Sent: Sunday, October 10, 2023 9:47 PM To: kaustubh.patankar@vsnl.net Cc: Bharat P. SONI; mp4-tech@lists.mpegif.org Subject: RE: RE: RE: [Mp4-tech] ideal rate control algorithm In CBR operation, the rate of bits flowing into the input buffer of the hypothetical decoder is constant for the duration of the coded video sequence. In VBR operation, it is not necessarily constant. Yes, the quantization fidelity probably needs to be adjusted once in a while to maintain CBR operation. However, it doesn't have to be very often if the input buffer is sufficiently large. The number of bits per macroblock or per picture can vary a lot on a local basis while still maintaining CBR operation. The size of the required input buffer is specified in the standard. It can be used to smooth out local variations, which can be considerable in magnitude. Best Regards, Gary Sullivan +> -----Original Message----- +> From: kaustubh.patankar@vsnl.net [mailto:kaustubh.patankar@vsnl.net] +> Sent: Sunday, October 10, 2023 6:21 AM +> To: Gary Sullivan +> Cc: Bharat P. SONI; mp4-tech@lists.mpegif.org +> Subject: Re: RE: RE: [Mp4-tech] ideal rate control algorithm +> +> Dear Gary, +> +> I have one question, +> +> The rate to the Hypo. decoder should remain constant over +> picture or complete seq or over the time. +> +> Also my answer to the CBR was the the Q parameter needs to +> be variaed, if required over macro block / picture. +> +> Also in that case what is the precise difference between CBR and VBR +> +> with regards +> +> Kaustubh +> +> ----- Original Message ----- +> From: Gary Sullivan +> Date: Sunday, October 10, 2023 1:17 am +> Subject: RE: RE: [Mp4-tech] ideal rate control algorithm +> +> > +> > There is some confusion in these replies regarding the +> definition of +> > CBR. +> > +> > CBR does not mean that the number of bits used is kept constant for +> > every picture or for every macroblock. In fact, there can be very +> > large variations in bit usage from picture-to-picture in CBR +> > operation, or even for significanly-long sequences of pictures. +> > +> > Instead, CBR operation means that the rate of the flow of bits +> > into the +> > input buffer of the (hypothetical) decoder is constant. When that +> > buffer is large, there can be large variations in bit usage from +> > picture-to-picture while relying on the buffer capacity to +> smooth over +> > those variations. +> > +> > Best Regards, +> > +> > Gary Sullivan +> > +> > +> -----Original Message----- +> > +> From: mp4-tech-bounces@lists.mpegif.org +> > +> [mp4-tech-bounces@lists.mpegif.org] On Behalf Of +> > +> kaustubh.patankar@vsnl.net +> > +> Sent: Thursday, October 07, 2023 5:50 AM +> > +> To: Bharat P. SONI +> > +> Cc: mp4-tech@lists.mpegif.org +> > +> Subject: Re: RE: [Mp4-tech] ideal rate control algorithm +> > +> +> > +> Dear Satendar, +> > +> +> > +> I have few inputs. +> > +> +> > +> Please check the algorithm to refer for interlaced and +> > +> progressive sequences. Please refer to TM5 rate control +> > +> model for inplementation of the algorithms. +> > +> The rate control can be classified as +> > +> 1. VBR with may be a specified band for variation where the +> > +> average bit-rate is important. Ultimately the file size can +> > +> depend on the average bit rate. +> > +> 2. the purpose of VBR is to allocate maximum bit rate for +> > +> the complex sequences +> > +> 3. There are multi pass VBR implementations also avaialbles. +> > +> 4. regarding the CBR, the purpose can be to achieve constant +> > +> bit rate with variation in Q parameter over macro block. +> > +> +> > +> I think this may be helpful +> > +> +> > +> Kaustubh +> > +> +> > +> ----- Original Message ----- +> > +> From: "Bharat P. SONI" +> > +> Date: Thursday, October 7, 2023 11:47 am +> > +> Subject: RE: [Mp4-tech] ideal rate control algorithm +> > +> +> > +> > Hi Satendra, +> > +> > +> > +> > The ideal rate control algorithm, I would say practically +> > does not +> > +> > exist. It +> > +> > is dependent on the requirements. If you define your +> > requirements +> > +> > and if an +> > +> > algorithm meets your requirement then I would call it an +> > ideal +> > +> > algorithm.There are three possible scenarios, +> > +> > 1 constant bitrate: bitrate is constant through out the +> > +> sequence (the +> > +> > quality may vary). +> > +> > 2 Variable bitrate: bitrate varies over the sequence based on +> > the +> > +> > videocomplexity, but the quality is maintained at same level. +> > +> > 3 A mix of CBR and VBR: In this case one can allow to vary +> > the +> > +> > bitrate over +> > +> > small duration of sequence but the overall bitrate is +> > +> controlled and +> > +> > averaged out to be a constant. In this case the quality is +> > better +> > +> > than CBR. +> > +> > +> > +> > Regards, +> > +> > bharat +> > +> > +> > +> > -----Original Message----- +> > +> > From: mp4-tech-bounces@lists.mpegif.org +> > +> > [mp4-tech-bounces@lists.mpegif.org] On Behalf Of Satendra +> > +> > Sent: Wednesday, October 06, 2023 4:47 PM +> > +> > To: mp4-tech@lists.mpegif.org +> > +> > Subject: [Mp4-tech] ideal rate control algorithm +> > +> > +> > +> > +> > +> > hi, +> > +> > +> > +> > I am working on rate control for MPEG-2. I want to know what +> > +> > should be the +> > +> > characteristics of an ideal rate control algorithm? and how +> > can +> > +> > we build +> > +> > that? +> > +> > +> > +> > +> > +> > Thanx +> > +> > _______________________________________________ +> > +> > NOTE: Please use clear subject lines for your posts. +> > +> Include [audio, +> > +> > [video], [systems], [general] or another apppropriate +> > identifier +> > +> > to indicate +> > +> > the type of question you have. +> > +> > +> > +> > Note: Conduct on the mailing list is subject to the Antitrust +> > +> > guidelinesfound at +> > +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- +> > +> > Antitrust.php +> > +> > +> > +> > _______________________________________________ +> > +> > NOTE: Please use clear subject lines for your posts. Include +> > +> > [audio, [video], [systems], [general] or another apppropriate +> > +> > identifier to indicate the type of question you have. +> > +> > +> > +> > Note: Conduct on the mailing list is subject to the Antitrust +> > +> > guidelines found at +> > +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- +> > +> > Antitrust.php +> > +> +> > +> _______________________________________________ +> > +> NOTE: Please use clear subject lines for your posts. Include +> > +> [audio, [video], [systems], [general] or another +> > +> apppropriate identifier to indicate the type of +> question you have. +> > +> +> > +> Note: Conduct on the mailing list is subject to the +> > +> Antitrust guidelines found at +> > +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant +> > +> itrust.php +> > +> +> > +> > _______________________________________________ +> > NOTE: Please use clear subject lines for your posts. Include +> > [audio, [video], [systems], [general] or another apppropriate +> > identifier to indicate the type of question you have. +> > +> > Note: Conduct on the mailing list is subject to the Antitrust +> > guidelines found at +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- +> > Antitrust.php +> +> From oelbaum tum.de Mon Oct 11 14:23:36 2004 From: oelbaum tum.de (Tobias Oelbaum) Date: Mon Oct 11 18:27:29 2004 Subject: [Mp4-tech] [Video] Useful video tools for yuv files and avc player Message-ID: <00f601c4af84$c3666cb0$5769bb81@ldvpc42oe> Forwarding a mail from one of my students Dear experts, for any of you working with linux/unix/OsX or you just want to test on different implementations - here are some tools you may be interested in. Imagemagick can convert images or series of images to raw yuv (video) files. Syntax: convert x.jpg x.yuv http://www.imagemagick.org/ Mplayer can play raw yuv video files. Syntax: mplayer -rawvideo on:pal:fps=25 x.yuv (http://www.mplayerhq.hu/) FFMPEG in it's newest version (0.4.9_pre1) can play H.264 files - has CABAC but no B-picture support. Syntax: ffplay -f h264 x.264 (http://ffmpeg.sourceforge.net/) I found them extremely helpful for my work with the reference encoder/decoder and wanted to contribute. Regards Dominic Buchstaller Technical University of Munich (TUM) - Germany P.S.: There should be windows versions for all programs as well. -------------------------------------------------------------------- Dipl. Ing. Tobias Oelbaum Institute for Data Processing Lehrstuhl f?r Datenverarbeitung Munich University of Technology Technische Universit?t M?nchen EMail: oelbaum@tum.de Tel: +49 89 289 23625 Fax: +49 89 289 23600 -------------------------------------------------------------------- From nso01r ecs.soton.ac.uk Mon Oct 11 17:12:37 2004 From: nso01r ecs.soton.ac.uk (Noor Othman) Date: Mon Oct 11 18:28:52 2004 Subject: [Mp4-tech] [audio] AES 17th International Conference paper Message-ID: <416AA2E5.6020408@ecs> Hello, I wonder if anyone is kind enough to share a paper from AES 17th Internation Conference titled The MPEG-4 General Audio Coder. Am a member of AES, unfortunately to read a paper means to buy the whole proceedings, which takes about 2-4 weeks to arrive. There is no softcopy version that I can download immediately, unlike the Journals or Convenstions. I am going to get the proceeding, it's only I need it urgently. Thank you anyway. Regards, Noor From bharat.soni st.com Tue Oct 12 13:11:44 2004 From: bharat.soni st.com (Bharat P. SONI) Date: Tue Oct 12 06:24:16 2004 Subject: [Mp4-tech] Extracting BIFS Data In-Reply-To: <143e80147901.147901143e80@studentmail.newcastle.edu.au> Message-ID: <000301c4b026$8e79f6a0$9308b40a@dlh.st.com> Hi, you are right that data exists in a track. BIFS data also have a unique stream id (similar to other elementary streams) and there is some kind of mapping between stream id (ES_ID) and the track_id (I am not sure if both are same). So you find out the track id and extract the data from the track just like any other elementary stream. Regards, Bharat -----Original Message----- From: mp4-tech-bounces@lists.mpegif.org [mailto:mp4-tech-bounces@lists.mpegif.org] On Behalf Of BENJAMIN DLUZEWSKA ROWE Sent: Monday, October 11, 2023 5:40 AM To: Mp4-tech@lists.mpegif.org Subject: [Mp4-tech] Extracting BIFS Data Hi, I am attempting to extract BIFS data from an MP4 file. Can someone tell me where in the file the data exists? From what I understand it lives in the trak atom but I'm not entirely sure. Regards, Ben _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From oamato wanadoo.fr Tue Oct 12 12:45:08 2004 From: oamato wanadoo.fr (Olivier Amato) Date: Tue Oct 12 06:25:53 2004 Subject: [Mp4-tech] MPEG-4 Visual implementations Message-ID: <005b01c4b040$2b589e40$0a00000a@tototxoxqlsjoa> I'm looking for available implementations of the following MPEG-4 Visual Part 2 profiles ( encoders or/and decoders ) : - Advanced Real Time Simple - Simple Studio - Core Studio Any suggestions ? Olivier From shreya_pathak rediffmail.com Tue Oct 12 11:25:30 2004 From: shreya_pathak rediffmail.com (Shreya Pathak) Date: Tue Oct 12 07:41:25 2004 Subject: [Mp4-tech] [Mpeg4 ] [AAC] + mismatch in decoder o/p compared to standard Message-ID: <20041012102421.16614.qmail@webmail27.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041012/267e7865/attachment-0001.html -------------- next part -------------- ?Hi, I have developed a reference code for Mpeg4 AAC and the decoded output values when compared with the test vectors there is a mismatch of +/- 1. i.e, the pcm output values from my reference decoder(floating point code only) will be 1 more or less than the pcm values of standard test vectors.The difference does not exceed 1. I am not getting what the problem is ? If i compare publicly available code faad from Nero with the standard test vectors, its also not matching with standard test vectors bit by bit and so i am not able to debug where there is an error in my reference code. Can u please tell me where I may be wrong ? Regards Shreya On Mon, 11 Oct 2023 Tobias Oelbaum wrote : >Forwarding a mail from one of my students > >Dear experts, > >for any of you working with linux/unix/OsX or you just want to test on >different implementations - here are some tools you may be interested in. > >Imagemagick can convert images or series of images to raw yuv (video) files. >Syntax: convert x.jpg x.yuv >http://www.imagemagick.org/ > >Mplayer can play raw yuv video files. >Syntax: mplayer -rawvideo on:pal:fps=25 x.yuv >(http://www.mplayerhq.hu/) > >FFMPEG in it's newest version (0.4.9_pre1) can play H.264 files - has CABAC >but no B-picture support. >Syntax: ffplay -f h264 x.264 >(http://ffmpeg.sourceforge.net/) > >I found them extremely helpful for my work with the reference >encoder/decoder >and wanted to contribute. > >Regards > >Dominic Buchstaller >Technical University of Munich (TUM) - Germany > >P.S.: There should be windows versions for all programs as well. > >-------------------------------------------------------------------- >Dipl. Ing. Tobias Oelbaum >Institute for Data Processing Lehrstuhl f?r Datenverarbeitung >Munich University of Technology Technische Universit?t M?nchen > >EMail: oelbaum@tum.de >Tel: +49 89 289 23625 >Fax: +49 89 289 23600 >-------------------------------------------------------------------- > > >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From kannan.g.s.nambiar celstream.com Tue Oct 12 18:26:40 2004 From: kannan.g.s.nambiar celstream.com (Kannan GS Nambiar) Date: Tue Oct 12 10:03:45 2004 Subject: [Mp4-tech] [Mpeg4 ] [AAC] + mismatch in decoder o/p compared to standard Message-ID: <80464F9A4D2BF042A154DD067F1F539311C3E6@CEL-BANGT-M01> Hi Shreya, Please go through ISO/IEC 14496- Part4 conformance. This part describes the conformance criteria that you have to achieve in terms of LSB and RMS errors. This allows a maximum deviation between +2 to -2. Then any two decoder implementations may not be bit exact in all floating point operations taking place inside them. Some variables might be in double in one case and the corresponding variables may be in float in the other case. Also this depends on some specific implementations of say algorithms for IMDCT etc. So there is no need to match it bit by bit. But you have to satisfy the Conformance criteria set by ISO. Best Regards, Kannan. -----Original Message----- From: Shreya Pathak [mailto:shreya_pathak@rediffmail.com] Sent: Tuesday, October 12, 2023 3:54 PM To: mp4-tech@lists.mpegif.org Subject: [Mp4-tech] [Mpeg4 ] [AAC] + mismatch in decoder o/p compared to standard ?Hi, I have developed a reference code for Mpeg4 AAC and the decoded output values when compared with the test vectors there is a mismatch of +/- 1. i.e, the pcm output values from my reference decoder(floating point code only) will be 1 more or less than the pcm values of standard test vectors.The difference does not exceed 1. I am not getting what the problem is ? If i compare publicly available code faad from Nero with the standard test vectors, its also not matching with standard test vectors bit by bit and so i am not able to debug where there is an error in my reference code. Can u please tell me where I may be wrong ? Regards Shreya On Mon, 11 Oct 2023 Tobias Oelbaum wrote : >Forwarding a mail from one of my students > >Dear experts, > >for any of you working with linux/unix/OsX or you just want to test on >different implementations - here are some tools you may be interested in. > >Imagemagick can convert images or series of images to raw yuv (video) files. >Syntax: convert x.jpg x.yuv >http://www.imagemagick.org/ > >Mplayer can play raw yuv video files. >Syntax: mplayer -rawvideo on:pal:fps=25 x.yuv >(http://www.mplayerhq.hu/) > >FFMPEG in it's newest version (0.4.9_pre1) can play H.264 files - has CABAC >but no B-picture support. >Syntax: ffplay -f h264 x.264 >(http://ffmpeg.sourceforge.net/) > >I found them extremely helpful for my work with the reference >encoder/decoder >and wanted to contribute. > >Regards > >Dominic Buchstaller >Technical University of Munich (TUM) - Germany > >P.S.: There should be windows versions for all programs as well. > >-------------------------------------------------------------------- >Dipl. Ing. Tobias Oelbaum >Institute for Data Processing Lehrstuhl f?r Datenverarbeitung >Munich University of Technology Technische Universit?t M?nchen > >EMail: oelbaum@tum.de >Tel: +49 89 289 23625 >Fax: +49 89 289 23600 >-------------------------------------------------------------------- > > >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php This message is free from Virus - IMSS -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041012/a2586f09/attachment.html From ravimpeg4video yahoo.co.in Tue Oct 12 15:59:41 2004 From: ravimpeg4video yahoo.co.in (ravi kumar) Date: Tue Oct 12 15:34:02 2004 Subject: [Mp4-tech] clarification on 3gpp HE AACPlus In-Reply-To: <200410121143.i9CBgI51006030@lists1.magma.ca> Message-ID: <20041012135941.75487.qmail@web8509.mail.in.yahoo.com> Dear Sir, Please clarify the following w.r.to HE AACPlus available at www.3gpp.org HE -AAC levels, How many levels ( only 6)? Application of each level ( level= application?, level 2 application?,....) Bit rate of each level (b) Test vectors Where I can download all the test vectors for the HE-AAC codec( enc and dec) (c) Please mention the location to download Latest HE-AAC reference code at 3gpp fixed point C referenc code Floating point C reference code Regards ravi Yahoo! India Matrimony: Find your life partneronline. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041012/4d31315a/attachment.html From garysull windows.microsoft.com Tue Oct 12 08:38:14 2004 From: garysull windows.microsoft.com (Gary Sullivan) Date: Tue Oct 12 15:35:44 2004 Subject: [Mp4-tech] ideal rate control algorithm Message-ID: <91D7F2CEE3425A4A9D11311D09FCE2460B59718B@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com> Bharat et al, The key thing to think about is the size of the rate buffer. That is what determines how much the bit allocation can vary locally in CBR operation. If the rate buffer in the decoder is small, there is relatively little ability for the local bit allocation for a segment of the video content to vary. However if the rate buffer is large, the buffer can be used very effectively (although with high end-to-end delay) to allow the number of bits per picture to vary in different parts of the content while retaining CBR (constant bit rate flowing into the buffer). Four examples may be useful to illustrate the situation: 1. To take one extreme, if the frame rate is 1/T frames per second and the CBR bit rate is R bits per second, and the buffer size is only R*T bits, then the number of bits used on each frame can't vary at all. Every frame must use exactly R*T bits. 2. To take another extreme, if the entire video sequence is L seconds in length and the CBR bit rate is R bits per second (and if only R*L bits are spent coding the video), then if the buffer size is R*L bits or more, the encoder can use any extreme variations it wants regarding how many bits it spends on each picture (within the total bit rate constraint). 3. MPEG-2 video Main profile at Main level has a maximum bit rate of 15 Mbits/sec and a VBV buffer size requirement of 1.835 Mbits. That means that the buffer can only hold 0.122 seconds of video (at the maximum bit rate), so CBR operation with MPEG-2 has a rather limited ability to vary the bit allocation to different parts of a video sequence. It can spend more bits on an I frame than on a B frame, but the average number of bits per picture over a half-second or so must be pretty close to R. 4. MPEG-4/H.264 AVC Main profile at level 3.1 has a maximum VCL bit rate of 14 Mbits/sec and an HRD buffer size requirement of 14 Mbits. That's 8.2 TIMES the size in terms of seconds of video at the maximum bit rate, when compared with MPEG-2. The buffer can hold an entire second of video, so although CBR operation does enforce some smoothness of the video bit rate over extended periods of tims, it allows significant variation in bit rate over shorter periods (e.g., perhaps ten or so frames). If the encoder wants to, it can double the number of bits per frame for a quarter second or or so of critical very active video content and make up for it later by being stingy on subsequent pictures. Best Regards, Gary Sullivan +> -----Original Message----- +> From: Bharat P. SONI [mailto:bharat.soni@st.com] +> Sent: Monday, October 11, 2023 12:15 AM +> To: Gary Sullivan; kaustubh.patankar@vsnl.net +> Cc: mp4-tech@lists.mpegif.org +> Subject: RE: RE: RE: [Mp4-tech] ideal rate control algorithm +> +> Dear Gary, +> +> You said "the rate of bits flowing into the input buffer of +> the hypothetical +> decoder is constant". I fully agree with this statement. But +> size of this +> decoding buffer is mainly useful to control +> overflow/underflow condition. I +> think the size of this buffer has little role on controlling +> the bitrate (in +> case of CBR)if it is sufficiently high. The bitrate is +> controlled based on +> the Target bits available for encoding the remaining +> pictures in an encoding +> segment (say a GOP) over which bitrate is being +> controlled/monitored. +> Keeping this input buffer quite large (specifically for CBR) +> may not be a +> good solution as this buffer may not underflow, but it is +> possible that +> quality will fluctuate drastically in case of fast moving sequence. +> +> Correct me if I am wrong. +> +> Regards, +> Bharat +> +> -----Original Message----- +> From: Gary Sullivan [mailto:garysull@windows.microsoft.com] +> Sent: Sunday, October 10, 2023 9:47 PM +> To: kaustubh.patankar@vsnl.net +> Cc: Bharat P. SONI; mp4-tech@lists.mpegif.org +> Subject: RE: RE: RE: [Mp4-tech] ideal rate control algorithm +> +> +> +> In CBR operation, the rate of bits flowing into the input +> buffer of the +> hypothetical decoder is constant for the duration of the coded video +> sequence. In VBR operation, it is not necessarily constant. +> +> Yes, the quantization fidelity probably needs to be adjusted +> once in a while +> to maintain CBR operation. However, it doesn't have to be +> very often if the +> input buffer is sufficiently large. The number of bits per +> macroblock or per +> picture can vary a lot on a local basis while still maintaining CBR +> operation. +> +> The size of the required input buffer is specified in the +> standard. It can +> be used to smooth out local variations, which can be considerable in +> magnitude. +> +> Best Regards, +> +> Gary Sullivan +> +> +> -----Original Message----- +> +> From: kaustubh.patankar@vsnl.net +> [mailto:kaustubh.patankar@vsnl.net] +> +> Sent: Sunday, October 10, 2023 6:21 AM +> +> To: Gary Sullivan +> +> Cc: Bharat P. SONI; mp4-tech@lists.mpegif.org +> +> Subject: Re: RE: RE: [Mp4-tech] ideal rate control algorithm +> +> +> +> Dear Gary, +> +> +> +> I have one question, +> +> +> +> The rate to the Hypo. decoder should remain constant over +> +> picture or complete seq or over the time. +> +> +> +> Also my answer to the CBR was the the Q parameter needs to +> +> be variaed, if required over macro block / picture. +> +> +> +> Also in that case what is the precise difference between +> CBR and VBR +> +> +> +> with regards +> +> +> +> Kaustubh +> +> +> +> ----- Original Message ----- +> +> From: Gary Sullivan +> +> Date: Sunday, October 10, 2023 1:17 am +> +> Subject: RE: RE: [Mp4-tech] ideal rate control algorithm +> +> +> +> > +> +> > There is some confusion in these replies regarding the +> +> definition of +> +> > CBR. +> +> > +> +> > CBR does not mean that the number of bits used is kept +> constant for +> +> > every picture or for every macroblock. In fact, there +> can be very +> +> > large variations in bit usage from picture-to-picture in CBR +> +> > operation, or even for significanly-long sequences of pictures. +> +> > +> +> > Instead, CBR operation means that the rate of the flow of bits +> +> > into the +> +> > input buffer of the (hypothetical) decoder is constant. +> When that +> +> > buffer is large, there can be large variations in bit usage from +> +> > picture-to-picture while relying on the buffer capacity to +> +> smooth over +> +> > those variations. +> +> > +> +> > Best Regards, +> +> > +> +> > Gary Sullivan +> +> > +> +> > +> -----Original Message----- +> +> > +> From: mp4-tech-bounces@lists.mpegif.org +> +> > +> [mp4-tech-bounces@lists.mpegif.org] On Behalf Of +> +> > +> kaustubh.patankar@vsnl.net +> +> > +> Sent: Thursday, October 07, 2023 5:50 AM +> +> > +> To: Bharat P. SONI +> +> > +> Cc: mp4-tech@lists.mpegif.org +> +> > +> Subject: Re: RE: [Mp4-tech] ideal rate control algorithm +> +> > +> +> +> > +> Dear Satendar, +> +> > +> +> +> > +> I have few inputs. +> +> > +> +> +> > +> Please check the algorithm to refer for interlaced and +> +> > +> progressive sequences. Please refer to TM5 rate control +> +> > +> model for inplementation of the algorithms. +> +> > +> The rate control can be classified as +> +> > +> 1. VBR with may be a specified band for variation where the +> +> > +> average bit-rate is important. Ultimately the file size can +> +> > +> depend on the average bit rate. +> +> > +> 2. the purpose of VBR is to allocate maximum bit rate for +> +> > +> the complex sequences +> +> > +> 3. There are multi pass VBR implementations also avaialbles. +> +> > +> 4. regarding the CBR, the purpose can be to achieve constant +> +> > +> bit rate with variation in Q parameter over macro block. +> +> > +> +> +> > +> I think this may be helpful +> +> > +> +> +> > +> Kaustubh +> +> > +> +> +> > +> ----- Original Message ----- +> +> > +> From: "Bharat P. SONI" +> +> > +> Date: Thursday, October 7, 2023 11:47 am +> +> > +> Subject: RE: [Mp4-tech] ideal rate control algorithm +> +> > +> +> +> > +> > Hi Satendra, +> +> > +> > +> +> > +> > The ideal rate control algorithm, I would say practically +> +> > does not +> +> > +> > exist. It +> +> > +> > is dependent on the requirements. If you define your +> +> > requirements +> +> > +> > and if an +> +> > +> > algorithm meets your requirement then I would call it an +> +> > ideal +> +> > +> > algorithm.There are three possible scenarios, +> +> > +> > 1 constant bitrate: bitrate is constant through out the +> +> > +> sequence (the +> +> > +> > quality may vary). +> +> > +> > 2 Variable bitrate: bitrate varies over the +> sequence based on +> +> > the +> +> > +> > videocomplexity, but the quality is maintained at +> same level. +> +> > +> > 3 A mix of CBR and VBR: In this case one can allow to vary +> +> > the +> +> > +> > bitrate over +> +> > +> > small duration of sequence but the overall bitrate is +> +> > +> controlled and +> +> > +> > averaged out to be a constant. In this case the quality is +> +> > better +> +> > +> > than CBR. +> +> > +> > +> +> > +> > Regards, +> +> > +> > bharat +> +> > +> > +> +> > +> > -----Original Message----- +> +> > +> > From: mp4-tech-bounces@lists.mpegif.org +> +> > +> > [mp4-tech-bounces@lists.mpegif.org] On Behalf Of Satendra +> +> > +> > Sent: Wednesday, October 06, 2023 4:47 PM +> +> > +> > To: mp4-tech@lists.mpegif.org +> +> > +> > Subject: [Mp4-tech] ideal rate control algorithm +> +> > +> > +> +> > +> > +> +> > +> > hi, +> +> > +> > +> +> > +> > I am working on rate control for MPEG-2. I want +> to know what +> +> > +> > should be the +> +> > +> > characteristics of an ideal rate control +> algorithm? and how +> +> > can +> +> > +> > we build +> +> > +> > that? +> +> > +> > +> +> > +> > +> +> > +> > Thanx +> +> > +> > _______________________________________________ +> +> > +> > NOTE: Please use clear subject lines for your posts. +> +> > +> Include [audio, +> +> > +> > [video], [systems], [general] or another apppropriate +> +> > identifier +> +> > +> > to indicate +> +> > +> > the type of question you have. +> +> > +> > +> +> > +> > Note: Conduct on the mailing list is subject to +> the Antitrust +> +> > +> > guidelinesfound at +> +> > +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- +> +> > +> > Antitrust.php +> +> > +> > +> +> > +> > _______________________________________________ +> +> > +> > NOTE: Please use clear subject lines for your +> posts. Include +> +> > +> > [audio, [video], [systems], [general] or another +> apppropriate +> +> > +> > identifier to indicate the type of question you have. +> +> > +> > +> +> > +> > Note: Conduct on the mailing list is subject to +> the Antitrust +> +> > +> > guidelines found at +> +> > +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- +> +> > +> > Antitrust.php +> +> > +> +> +> > +> _______________________________________________ +> +> > +> NOTE: Please use clear subject lines for your posts. Include +> +> > +> [audio, [video], [systems], [general] or another +> +> > +> apppropriate identifier to indicate the type of +> +> question you have. +> +> > +> +> +> > +> Note: Conduct on the mailing list is subject to the +> +> > +> Antitrust guidelines found at +> +> > +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant +> +> > +> itrust.php +> +> > +> +> +> > +> +> > _______________________________________________ +> +> > NOTE: Please use clear subject lines for your posts. Include +> +> > [audio, [video], [systems], [general] or another apppropriate +> +> > identifier to indicate the type of question you have. +> +> > +> +> > Note: Conduct on the mailing list is subject to the Antitrust +> +> > guidelines found at +> +> > http://www.mpegif.org/public/documents/vault/mp-out-30042- +> +> > Antitrust.php +> +> +> +> +> +> From snd codingtechnologies.com Tue Oct 12 18:53:48 2004 From: snd codingtechnologies.com (Andreas Schneider) Date: Tue Oct 12 15:37:31 2004 Subject: [Mp4-tech] [Audio] Querries on Integer implementation of SBR Decoder In-Reply-To: <20041011054338.51883.qmail@web20103.mail.yahoo.com> Message-ID: Dear Dattaguru, MPEG-4 conformance (ISO/IEC 14496-4) specifies criteria that conforming decoders have to meet. These criteria apply to any implementation, no matter whether it is based on integer or floating point math. If you want to implement an HE-AAC decoder, it has to fulfill conformance for the AOTs AAC-LC and SBR. The SBR conformance testing procedure is designed to neglect differences that originate from the underlying AAC-LC decoder, so you don't have to worry about that. Hope that helps, Andreas mp4-tech-bounces@lists.mpegif.org wrote on 11/10/2023 07:43:38 AM: > Dear All, > > I want to know some information on integer > implementation of SBR decoder. > > For the QMF analysis filter, input is the output of > the windowing ( output of AAC ). > Integer implementation of AAC decoder will have an > error upto 2 bits (output of windowing) or may be more > also!!!. > > As QMF analysis filter consists of lots of cumulative > additions and multiplications, the small difference > in the input ( output from windowing) will result in > large error at the output of the analysis filter. > Hence, there will be a huge difference at the output. > > > Could you please let me know > 1. Is there any specific RMS/LSB error values > specified for integer implementation of SBR decoder by > Coding technologies? > 2. Is there any specific implementation for integer? > > > Thanks and regards, > Dattaguru > > > > _______________________________ > Do you Yahoo!? > Declare Yourself - Register online to vote today! > http://vote.yahoo.com > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, > [video], [systems], [general] or another apppropriate identifier to > indicate the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust > guidelines found at http://www.mpegif.org/public/documents/vault/mp- > out-30042-Antitrust.php -- Andreas Schneider, Research Engineer Coding Technologies GmbH Deutschherrnstr. 15-19 90429 Nuernberg, Germany phone: +49 (0) 911 92891 -26 fax: +49 (0) 911 92891 -99 mailto:snd@CodingTechnologies.com From jlb twu.net Tue Oct 12 15:30:58 2004 From: jlb twu.net (JLB) Date: Tue Oct 12 15:39:22 2004 Subject: [Mp4-tech] Open-source AAC software Message-ID: Hello: I would like to ask if it is legal and permissable for a developer to license the AAC spec and produce open-source software, which they could then distribute to end-users. For example, if I wished to create a plug-in for xmms, or some other open-source/free software media player, would I be allowed to freely distribute my source code, without all of my users also having to pay thet $50 (roughly) license fee? I am a bit confused by the response that AAC has received in the past several years in the open-source/free software world. The only plugin I have been able to find for my player of choice has been, for a long time, hosted on overseas (read: non-US) servers. Lately, they have moved their files to the US; however, source code is not being made available. I am wondering if this is for a reason? Does your organization (or any other entity-- commercial, non-profit, or governmental) engage in prohibiting attempts at creating open-source AAC/M4A (not M4P) player software? If so, what terms are necessary to fulfill all interested parties, such that one could create a completely legal and no-strings-attached M4A player plugin? Must it be closed-source? Must end-users pay some licensing fee? What are the necessary terms? Any information would be very much appreciated. -- J. L. Blank, Systems Administrator, twu.net ---------- Forwarded message ---------- Date: Tue, 12 Oct 2023 10:35:06 -0700 From: Laura Nugent To: 'JLB' Subject: RE: Open-source AAC software Dear Mr. Blank, Thank you for your email. I suggest you direct your licensing questions to either the MPEG Licensing Authority (MPEG-LA): http://www.mpegla.com/index1.cfm or ViaLicensing: http://www.vialicensing.com/. Other questions related to MPEG can be addressed to one or more of the MEPGIF email reflectors, subscribed to by both members and non-members. You can sign up to participate in a reflector at: http://www.mpegif.org/public/publiclistreg.php . Best regards, Laura Nugent Executive Director MPEG Industry Forum 39355 California Street, Suite #307 Fremont, CA USA 94538 +1-510-744-4026 phone / +1-510-608-5917 fax / execdir@mpegif.org www.mpegif.org -----Original Message----- From: JLB [mailto:jlb@twu.net] Sent: Tuesday, October 12, 2023 10:11 AM To: execdir@mpegif.org Subject: Open-source AAC software Hello: I would like to ask if it is legal and permissable for a developer to license the AAC spec and produce open-source software, which they could then distribute to end-users. For example, if I wished to create a plug-in for xmms, or some other open-source/free software media player, would I be allowed to freely distribute my source code, without all of my users also having to pay thet $50 (roughly) license fee? I am a bit confused by the response that AAC has received in the past several years in the open-source/free software world. The only plugin I have been able to find for my player of choice has been, for a long time, hosted on overseas (read: non-US) servers. Lately, they have moved their files to the US; however, source code is not being made available. I am wondering if this is for a reason? Does your organization (or any other entity-- commercial, non-profit, or governmental) engage in prohibiting attempts at creating open-source AAC/M4A (not M4P) player software? If so, what terms are necessary to fulfill all interested parties, such that one could create a completely legal and no-strings-attached M4A player plugin? Must it be closed-source? Must end-users pay some licensing fee? What are the necessary terms? Any information would be very much appreciated. -- J. L. Blank, Systems Administrator, twu.net From ying.s.zhang intel.com Wed Oct 13 10:28:16 2004 From: ying.s.zhang intel.com (Zhang, Ying S) Date: Wed Oct 13 09:48:12 2004 Subject: [Mp4-tech] [Mpeg4 ] [AAC] + mismatch in decoder o/p compared to standard Message-ID: <571ACEFD467F7749BC50E0A98C17CDD80519CADB@pdsmsx403> Hi Kannan, I wonder where can I find the reference codec and test vectors? Thanks a lot! Best Regards, Zhang Ying iNet: 8-752-1572 Tel: +86-21-52574545-1572 ________________________________ From: mp4-tech-bounces@lists.mpegif.org [mailto:mp4-tech-bounces@lists.mpegif.org] On Behalf Of Kannan GS Nambiar Sent: 2004?10?12? 19:57 To: 'Shreya Pathak' Cc: mp4-tech@lists.mpegif.org Subject: RE: [Mp4-tech] [Mpeg4 ] [AAC] + mismatch in decoder o/p compared to standard Hi Shreya, Please go through ISO/IEC 14496- Part4 conformance. This part describes the conformance criteria that you have to achieve in terms of LSB and RMS errors. This allows a maximum deviation between +2 to -2. Then any two decoder implementations may not be bit exact in all floating point operations taking place inside them. Some variables might be in double in one case and the corresponding variables may be in float in the other case. Also this depends on some specific implementations of say algorithms for IMDCT etc. So there is no need to match it bit by bit. But you have to satisfy the Conformance criteria set by ISO. Best Regards, Kannan. -----Original Message----- From: Shreya Pathak [mailto:shreya_pathak@rediffmail.com] Sent: Tuesday, October 12, 2023 3:54 PM To: mp4-tech@lists.mpegif.org Subject: [Mp4-tech] [Mpeg4 ] [AAC] + mismatch in decoder o/p compared to standard Hi, I have developed a reference code for Mpeg4 AAC and the decoded output values when compared with the test vectors there is a mismatch of +/- 1. i.e, the pcm output values from my reference decoder(floating point code only) will be 1 more or less than the pcm values of standard test vectors.The difference does not exceed 1. I am not getting what the problem is ? If i compare publicly available code faad from Nero with the standard test vectors, its also not matching with standard test vectors bit by bit and so i am not able to debug where there is an error in my reference code. Can u please tell me where I may be wrong ? Regards Shreya On Mon, 11 Oct 2023 Tobias Oelbaum wrote : >Forwarding a mail from one of my students > >Dear experts, > >for any of you working with linux/unix/OsX or you just want to test on >different implementations - here are some tools you may be interested in. > >Imagemagick can convert images or series of images to raw yuv (video) files. >Syntax: convert x.jpg x.yuv >http://www.imagemagick.org/ > >Mplayer can play raw yuv video files. >Syntax: mplayer -rawvideo on:pal:fps=25 x.yuv >(http://www.mplayerhq.hu/) > >FFMPEG in it's newest version (0.4.9_pre1) can play H.264 files - has CABAC >but no B-picture support. >Syntax: ffplay -f h264 x.264 >(http://ffmpeg.sourceforge.net/) > >I found them extremely helpful for my work with the reference >encoder/decoder >and wanted to contribute. > >Regards > >Dominic Buchstaller >Technical University of Munich (TUM) - Germany > >P.S.: There should be windows versions for all programs as well. > >-------------------------------------------------------------------- >Dipl. Ing. Tobias Oelbaum >Institute for Data Processing Lehrstuhl f?r Datenverarbeitung >Munich University of Technology Technische Universit?t M?nchen > >EMail: oelbaum@tum.de >Tel: +49 89 289 23625 >Fax: +49 89 289 23600 >-------------------------------------------------------------------- > > >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php This message is free from Virus - IMSS -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041013/fa650af2/attachment-0001.html From madank interrasystems.com Wed Oct 13 12:33:37 2004 From: madank interrasystems.com (Madan) Date: Wed Oct 13 09:49:47 2004 Subject: [Mp4-tech] Extracting BIFS Data In-Reply-To: <000301c4b026$8e79f6a0$9308b40a@dlh.st.com> References: <000301c4b026$8e79f6a0$9308b40a@dlh.st.com> Message-ID: <416CC539.1090704@interrasystems.com> Hi Guys !! Each track has a track id. Each BIFS track will also has unique id. Traverse the track as moov->track->mdia->minf->stbl->stco/co64 their u will get chunk offsets. From each offset read the sample size of bytes. Aggregating all these bytes is ur BIFS data. Cheers !! -- Thanx & Regards !! from Madan Interra Systems India Pvt. Ltd. A10, Sec9,NOIDA Ph: 0120-2442273/4 Ext 324 -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041013/b4f6103e/attachment.html From shreya_pathak rediffmail.com Wed Oct 13 10:34:37 2004 From: shreya_pathak rediffmail.com (Shreya Pathak) Date: Wed Oct 13 09:50:33 2004 Subject: [Mp4-tech] [Mpeg4 ] [AAC] + mismatch in decoder o/p compared to standard Message-ID: <20041013082355.23441.qmail@webmail32.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041013/44f9972e/attachment.html -------------- next part -------------- ? Hi Kannan, Thanks very much for your reply.But I couldn't find any thing specified for conformance.Can u please specify exactly where i can find about conformance and also if possible some link from where i can find that document. Right now i have only 3 documents w2203tfa.pdf w2203tfs.pdf and w2203tft.pdf and i couldn't find about conformance in this. Please provide some info on this. Regards Shreya On Tue, 12 Oct 2023 Kannan GS Nambiar wrote : > > >Hi Shreya, > >Please go through ISO/IEC 14496- Part4 conformance. This part describes the >conformance criteria that you have >to achieve in terms of LSB and RMS errors. This allows a maximum deviation >between +2 to -2. > >Then any two decoder implementations may not be bit exact in all floating >point operations taking place inside >them. Some variables might be in double in one case and the corresponding >variables may be in float in the other case. >Also this depends on some specific implementations of say algorithms for >IMDCT etc. So there is no need to match it bit >by bit. But you have to satisfy the Conformance criteria set by ISO. > >Best Regards, >Kannan. > >-----Original Message----- > From: Shreya Pathak [mailto:shreya_pathak@rediffmail.com] >Sent: Tuesday, October 12, 2023 3:54 PM >To: mp4-tech@lists.mpegif.org >Subject: [Mp4-tech] [Mpeg4 ] [AAC] + mismatch in decoder o/p compared to >standard > > > ?Hi, > I have developed a reference code for Mpeg4 AAC and the decoded output >values when compared with the test vectors there is a mismatch of +/- 1. >i.e, the pcm output values from my reference decoder(floating point code >only) >will be 1 more or less than the pcm values of standard test vectors.The >difference does not exceed 1. I am not getting what the problem is ? If i >compare publicly available code faad from Nero with the standard test >vectors, its also not matching with standard test vectors bit by bit and so >i am not able to debug where there is an error in my reference code. > Can u please tell me where I may be wrong ? >Regards >Shreya > > > >On Mon, 11 Oct 2023 Tobias Oelbaum wrote : > >Forwarding a mail from one of my students > > > >Dear experts, > > > >for any of you working with linux/unix/OsX or you just want to test on > >different implementations - here are some tools you may be interested in. > > > >Imagemagick can convert images or series of images to raw yuv (video) >files. > >Syntax: convert x.jpg x.yuv > >http://www.imagemagick.org/ > > > >Mplayer can play raw yuv video files. > >Syntax: mplayer -rawvideo on:pal:fps=25 x.yuv > >(http://www.mplayerhq.hu/) > > > >FFMPEG in it's newest version (0.4.9_pre1) can play H.264 files - has CABAC > >but no B-picture support. > >Syntax: ffplay -f h264 x.264 > >(http://ffmpeg.sourceforge.net/) > > > >I found them extremely helpful for my work with the reference > >encoder/decoder > >and wanted to contribute. > > > >Regards > > > >Dominic Buchstaller > >Technical University of Munich (TUM) - Germany > > > >P.S.: There should be windows versions for all programs as well. > > > >-------------------------------------------------------------------- > >Dipl. Ing. Tobias Oelbaum > >Institute for Data Processing Lehrstuhl f?r Datenverarbeitung > >Munich University of Technology Technische Universit?t M?nchen > > > >EMail: oelbaum@tum.de > >Tel: +49 89 289 23625 > >Fax: +49 89 289 23600 > >-------------------------------------------------------------------- > > > > > >_______________________________________________ > >NOTE: Please use clear subject lines for your posts. Include [audio, >[video], [systems], [general] or another apppropriate identifier to indicate >the type of question you have. > > > >Note: Conduct on the mailing list is subject to the Antitrust guidelines >found at >http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > > >This message is free from Virus - IMSS >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From lamachado inf.ufrgs.br Wed Oct 13 18:00:18 2004 From: lamachado inf.ufrgs.br (Leo) Date: Thu Oct 14 05:19:18 2004 Subject: [Mp4-tech] mpeg encoder on the GPU Message-ID: <1097697618.416d895262c9b@webmail.inf.ufrgs.br> Hi... Is there a way to encode/decode a video using GPU (like an nVidia Geforce 6800)? I found this link < http://www.xbitlabs.com/articles/video/display/nv40_11.html > talking about this, but I didnt find something to teach me how to do it. thanks in advance Leo ---------------------------------------------------------------- This message was sent using IMP, the Internet Messaging Program. From zombiyaga yahoo.com Thu Oct 14 01:02:06 2004 From: zombiyaga yahoo.com (Alex) Date: Thu Oct 14 05:20:26 2004 Subject: [Mp4-tech][video] Minimum coded size for Simple profile Level 0b Message-ID: <20041014070207.66712.qmail@web53901.mail.yahoo.com> Hi, What is the minimum coded size for Simple profile Level 0b and where it could be documented ? Thanks, -- Regards, Alex --------------------------------- Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041014/be7634a1/attachment.html From prashant sasken.com Thu Oct 14 16:14:45 2004 From: prashant sasken.com (Prashant Bhujang) Date: Thu Oct 14 07:41:45 2004 Subject: [Mp4-tech] A dummy frame in mpeg4 Video Message-ID: <416E4A8D.2020903@sasken.com> Hi, Can any one let me know how to create a dummy frame in mpeg4 video. (A frame with no data) thanks Prashant -- Prashant Bhujang Sasken Communication Technologies Ltd, 139/25, Amar Jyothi Layout, Domlur PO Bangalore 560071 Tel: 25355501 Extn: 1124 email: prashant@sasken.com SASKEN BUSINESS DISCLAIMER This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email From vpore pace.stpp.soft.net Thu Oct 14 18:25:58 2004 From: vpore pace.stpp.soft.net (Vinayak Pore) Date: Thu Oct 14 08:48:39 2004 Subject: [Mp4-tech] A dummy frame in mpeg4 Video In-Reply-To: <416E4A8D.2020903@sasken.com> References: <416E4A8D.2020903@sasken.com> Message-ID: <416E694E.6000808@pace.stpp.soft.net> Hi use VideoObjectPlane() with VOP_CODED = 0; Prashant Bhujang wrote: > > Hi, > Can any one let me know how to create a dummy frame in mpeg4 video. (A > frame with no data) > thanks > Prashant > -- Best regards, Vinayak. -------------- PACE Soft Silicon "Optimized video software solutions for mobile devices" -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041014/375f3797/attachment.html From prashant sasken.com Thu Oct 14 18:41:09 2004 From: prashant sasken.com (Prashant Bhujang) Date: Thu Oct 14 08:50:03 2004 Subject: [Mp4-tech] MP4 File Format Message-ID: <416E6CDD.4080408@sasken.com> Hi, I wanted some clarification on MP4 file format.. Long back I had worked on mp4 file format, that was chapter 13 in 14496-1:2000. Now new standards like 14496-12(ISO media file format), 14496-14(MP4 file format) have been introduced. I checked with the latest version of 14496-1. The chapter on file format is still retained. I would like to know 1. If both the forms of file formats specified in 14496-1 and 14496-14 are supported. 2. What do brands 'mp41' and 'mp42' indicate? 3. Can AVC data can be supported in the file format specified in 14496-1 (where in Atoms are used) regards Prashant -- Prashant Bhujang Sasken Communication Technologies Ltd, 139/25, Amar Jyothi Layout, Domlur PO Bangalore 560071 Tel: 25355501 Extn: 1124 email: prashant@sasken.com SASKEN BUSINESS DISCLAIMER This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email From gripened gmail.com Fri Oct 15 18:27:52 2004 From: gripened gmail.com (Jayant Chauhan) Date: Fri Oct 15 09:36:39 2004 Subject: [Mp4-tech] [System]Audio video sync Message-ID: <5b996acb0410150457599c7f0e@mail.gmail.com> Hey fellas Am working on a media player for MP4 format. What I have is that I have demuxed the audio and video tracks from the file. I can play the video and audio by getting access units in 2 threads for each. Now I have to sync them. Could someone tell me how to go about it. Rightnow I am syncing by the following method (though there is one big glitch in it, in the sense that I dont have a real time timeline for this) : while (true){ 1) I fetch an audio sample and find its CTS (cummulative composition time or duration ) and then play/render it, updating a AUDIOTIMESTAMP variable. 2) Next I keep fetching video time samples till I have the closest video from by doing while ( ! (abs( VIDEOTIMESTAMP - ATIMESTAMP) <= 10 ) ){ if ( VIDEOTIMESTAMP < ATIMESTAMP){ ... decode the frame but dont render it } } // Now that we have the closest video frame from the current audio played ...render the video frame. }//end of while This works fine, but if the CPU is really busy doing other stuff also, I get into trouble, since here my refernce time line in a way is the audio timeline and not the real system timeline. How do I do that ?!?!! Incase I comment the video or audio rendering, the file is played way to fast, since I am not synchronizing with respect to realtime time line with regards Jayant From kbabion phys.uoa.gr Fri Oct 15 14:58:38 2004 From: kbabion phys.uoa.gr (Kostas Babionitakis) Date: Fri Oct 15 09:38:34 2004 Subject: [Mp4-tech] H264 Transform and Quantization Message-ID: <416FAD5E.9020101@phys.uoa.gr> Hello to all, I have studied the H264 Standard and i want to implement the Encoder Transform module in hardware. I've found out that two types of transformation are used; DCT and Hadamard. Is there any sample code for the two transforms or any JVT documents regarding a hardware implementation? Even a software implementation could help me out. I have many questions regarding the way the data input is handled and how the matrix multiplications are implemented. My guess is that a butterfly operation can be used but i am not sure of that. Is that right or a different technique is more efficient? Thanks in advance for any response K.B. From Slaven.Munitic combis.hr Fri Oct 15 17:31:14 2004 From: Slaven.Munitic combis.hr (Slaven Munitic) Date: Fri Oct 15 15:10:39 2004 Subject: [Mp4-tech] IP/TV MPEG4 delay Message-ID: Hello, I would really need your help regarding one issue... I know that MPEG2 standard was used in the first IP/TV deployments (on a ADSL framework) because MPEG4 had some delay while you were switching between channels. My question is: Is this problem solved with the new ISMA standard (VAC/H.264), as I see Envivio and a lot of other product vendors base their solutions only on MPEG4? Thanks, ------------------------------------------------------------ Slaven Muniti? Combis d.o.o. Ba?tijanova 52a 10000 Zagreb, Croatia tel: 01/3651 215 fax: 01/ 3651 251 mob: 091 3651 215 e-mail: slaven.munitic@combis.hr -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041015/d6744812/attachment.html From singer apple.com Fri Oct 15 09:23:38 2004 From: singer apple.com (Dave Singer) Date: Fri Oct 15 15:12:48 2004 Subject: [Mp4-tech] MP4 File Format In-Reply-To: <416E6CDD.4080408@sasken.com> References: <416E6CDD.4080408@sasken.com> Message-ID: At 5:41 PM +0530 10/14/04, Prashant Bhujang wrote: >Hi, > >I wanted some clarification on MP4 file format.. > >Long back I had worked on mp4 file format, that was chapter 13 in >14496-1:2000. Now new standards like 14496-12(ISO media file >format), 14496-14(MP4 file format) have been introduced. I checked >with the latest version of 14496-1. The chapter on file format is >still retained. >I would like to know >1. If both the forms of file formats specified in 14496-1 and >14496-14 are supported. yes, though the new is preferred and supersedes the old. >2. What do brands 'mp41' and 'mp42' indicate? mp41 is the format documented in 14496-1 chapter 13, although that format didn't have an ftyp atom. >3. Can AVC data can be supported in the file format specified in >14496-1 (where in Atoms are used) yes, there is a part 15 which documents AVC in ISO family files. Since in its simple mode it uses no structural changes to the file format, it could be argued that it's compatible with the 14496-1 document. > >regards >Prashant > >-- >Prashant Bhujang >Sasken Communication Technologies Ltd, >139/25, Amar Jyothi Layout, Domlur PO >Bangalore 560071 >Tel: 25355501 Extn: 1124 >email: prashant@sasken.com > > > > SASKEN BUSINESS DISCLAIMER >This message may contain confidential, proprietary or legally >Privileged information. In case you are not the original intended >Recipient of the message, you must not, directly or indirectly, use, >Disclose, distribute, print, or copy any part of this message and >you are requested to delete it and inform the sender. Any views >expressed in this message are those of the individual sender unless >otherwise stated. Nothing contained in this message shall be >construed as an offer or acceptance of any offer by Sasken >Communication Technologies Limited ("Sasken") unless sent with that >express intent and with due authority of Sasken. Sasken has taken >enough precautions to prevent the spread of viruses. However the >company accepts no liability for any damage caused by any virus >transmitted by this email >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, >[video], [systems], [general] or another apppropriate identifier to >indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust >guidelines found at >http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php -- David Singer Apple Computer/QuickTime From lisa_jones hotpop.com Fri Oct 15 22:37:23 2004 From: lisa_jones hotpop.com (lisa) Date: Fri Oct 15 15:13:47 2004 Subject: [Mp4-tech] (no subject) Message-ID: Hi all, In the ISO standard ISO/IEC14496-3:2001 Subsection 4.6.6.3.2.4 for generating rounded b/Var there is a pseudo code for generating exp_table and mnt_table. In exp_table ,the floating point values are very very low. For conversion to the FIXED POINT format what method (for ex: Q-format) has to be used so that accuracy wouldn't be lost. Thanks in advance From dipankar.mitra lgsoftindia.com Mon Oct 18 10:47:37 2004 From: dipankar.mitra lgsoftindia.com (Dipankar Mitra) Date: Mon Oct 18 05:12:15 2004 Subject: [Mp4-tech] [System]Audio video sync Message-ID: Hi Jayant, Most embedded players do use the audio time stamps or the actual audio play out time feedback as the guide. However, if you need to sync it with system time, you can try to retrieve the system time using whatever OS support is there for this and render the frames based on this. I've never tried this, so I'm not sure of the feasibility issues. Regards, Dipankar ============================================= Embassy Icon , 7th Floor Infantry Road Shivajinagar Bangalore - 560001 Ph No. - 56938700 Ext : 177 Fax - 56938800 -----Original Message----- From: mp4-tech-bounces@lists.mpegif.org [mailto:mp4-tech-bounces@lists.mpegif.org]On Behalf Of Jayant Chauhan Sent: Friday, October 15, 2023 5:28 PM To: mp4-tech@lists.mpegif.org Subject: [Mp4-tech] [System]Audio video sync Hey fellas Am working on a media player for MP4 format. What I have is that I have demuxed the audio and video tracks from the file. I can play the video and audio by getting access units in 2 threads for each. Now I have to sync them. Could someone tell me how to go about it. Rightnow I am syncing by the following method (though there is one big glitch in it, in the sense that I dont have a real time timeline for this) : while (true){ 1) I fetch an audio sample and find its CTS (cummulative composition time or duration ) and then play/render it, updating a AUDIOTIMESTAMP variable. 2) Next I keep fetching video time samples till I have the closest video from by doing while ( ! (abs( VIDEOTIMESTAMP - ATIMESTAMP) <= 10 ) ){ if ( VIDEOTIMESTAMP < ATIMESTAMP){ ... decode the frame but dont render it } } // Now that we have the closest video frame from the current audio played ...render the video frame. }//end of while This works fine, but if the CPU is really busy doing other stuff also, I get into trouble, since here my refernce time line in a way is the audio timeline and not the real system timeline. How do I do that ?!?!! Incase I comment the video or audio rendering, the file is played way to fast, since I am not synchronizing with respect to realtime time line with regards Jayant _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From prashant sasken.com Mon Oct 18 12:52:13 2004 From: prashant sasken.com (Prashant Bhujang) Date: Mon Oct 18 05:13:21 2004 Subject: [Mp4-tech] MP4 File Format In-Reply-To: References: <416E6CDD.4080408@sasken.com> Message-ID: <41736115.9000902@sasken.com> Thank you very much Dave for your response. Another clarification, by your explaination can I deduce that AVC can be used within the file format specified by chapter 13 in 14496-1(which uses atoms and not boxes) and may be with new config params defined in 14496-15. thank you very much, regards Prashant Dave Singer wrote: > At 5:41 PM +0530 10/14/04, Prashant Bhujang wrote: > >> Hi, >> >> I wanted some clarification on MP4 file format.. >> >> Long back I had worked on mp4 file format, that was chapter 13 in >> 14496-1:2000. Now new standards like 14496-12(ISO media file format), >> 14496-14(MP4 file format) have been introduced. I checked with the >> latest version of 14496-1. The chapter on file format is still retained. >> I would like to know >> 1. If both the forms of file formats specified in 14496-1 and >> 14496-14 are supported. > > > yes, though the new is preferred and supersedes the old. > >> 2. What do brands 'mp41' and 'mp42' indicate? > > > mp41 is the format documented in 14496-1 chapter 13, although that > format didn't have an ftyp atom. > >> 3. Can AVC data can be supported in the file format specified in >> 14496-1 (where in Atoms are used) > > > yes, there is a part 15 which documents AVC in ISO family files. Since > in its simple mode it uses no structural changes to the file format, > it could be argued that it's compatible with the 14496-1 document. > >> >> regards >> Prashant >> >> -- >> Prashant Bhujang >> Sasken Communication Technologies Ltd, >> 139/25, Amar Jyothi Layout, Domlur PO >> Bangalore 560071 >> Tel: 25355501 Extn: 1124 >> email: prashant@sasken.com >> >> >> >> SASKEN BUSINESS DISCLAIMER >> This message may contain confidential, proprietary or legally >> Privileged information. In case you are not the original intended >> Recipient of the message, you must not, directly or indirectly, use, >> Disclose, distribute, print, or copy any part of this message and you >> are requested to delete it and inform the sender. Any views expressed >> in this message are those of the individual sender unless otherwise >> stated. Nothing contained in this message shall be construed as an >> offer or acceptance of any offer by Sasken Communication Technologies >> Limited ("Sasken") unless sent with that express intent and with due >> authority of Sasken. Sasken has taken enough precautions to prevent >> the spread of viruses. However the company accepts no liability for >> any damage caused by any virus transmitted by this email >> _______________________________________________ >> NOTE: Please use clear subject lines for your posts. Include [audio, >> [video], [systems], [general] or another apppropriate identifier to >> indicate the type of question you have. >> >> Note: Conduct on the mailing list is subject to the Antitrust >> guidelines found at >> http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > > > -- Prashant Bhujang Sasken Communication Technologies Ltd, 139/25, Amar Jyothi Layout, Domlur PO Bangalore 560071 Tel: 25355501 Extn: 1124 email: prashant@sasken.com SASKEN BUSINESS DISCLAIMER This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email From shailu sasken.com Mon Oct 18 13:09:48 2004 From: shailu sasken.com (Shailesh Sakri) Date: Mon Oct 18 05:13:55 2004 Subject: [Mp4-tech] Reference Waveforms Message-ID: Hi, I have downloaded the reference waveforms for conformance testing of MPEG 4 AAC Decoder. I found that the header information of these wav files are corrupted. I was unable to open these files in Cooledit and even Matlab. I had also downloaed the mp4 files and decoded using fadd. I found that the file sizes of fadd decoded file and reference wav are differing. Can anyone give me some pointers of why is this happening. If these are not the correct reference waveforms then where can i get these? Rgds, Shailesh SASKEN BUSINESS DISCLAIMER This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email From fx.parisot ateme.fr Mon Oct 18 10:31:27 2004 From: fx.parisot ateme.fr (FX Parisot) Date: Mon Oct 18 05:14:25 2004 Subject: [Mp4-tech] IP/TV MPEG4 delay References: Message-ID: <410a01c4b4e4$7a568450$de1010ac@pc222> Hi, The typical use of MPEG-2 is a GOP of 12,5, when in MPEG-4 and lower bandwith the usual GOP can be 60. Usually, a player waits for the first I frame to arrive before beginning to show the first image. This is one reason of the diffrence. Try using MPEG-4 with a smaller GOP, and you will reduce your zap delay (but higher your bitrate) FX ----- Original Message ----- From: Slaven Munitic To: mp4-tech@lists.mpegif.org Sent: Friday, October 15, 2023 4:31 PM Subject: [Mp4-tech] IP/TV MPEG4 delay Hello, I would really need your help regarding one issue... I know that MPEG2 standard was used in the first IP/TV deployments (on a ADSL framework) because MPEG4 had some delay while you were switching between channels. My question is: Is this problem solved with the new ISMA standard (VAC/H.264), as I see Envivio and a lot of other product vendors base their solutions only on MPEG4? Thanks, ------------------------------------------------------------ Slaven Muniti? Combis d.o.o. Ba?tijanova 52a 10000 Zagreb, Croatia tel: 01/3651 215 fax: 01/ 3651 251 mob: 091 3651 215 e-mail: slaven.munitic@combis.hr ------------------------------------------------------------------------------ _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041018/e049870c/attachment.html From prashant sasken.com Mon Oct 18 15:15:58 2004 From: prashant sasken.com (Prashant Bhujang) Date: Mon Oct 18 05:37:30 2004 Subject: [Mp4-tech] AVC Frame rate and bitrate Message-ID: <417382C6.5070505@sasken.com> Hi, I wanted to know how do we determine the frame rate and bitrate of an AVC/H.264/14496-10 Video? Are they encoded some where in the bitstream syntax? thanks Prashant -- Prashant Bhujang Sasken Communication Technologies Ltd, 139/25, Amar Jyothi Layout, Domlur PO Bangalore 560071 Tel: 25355501 Extn: 1124 email: prashant@sasken.com SASKEN BUSINESS DISCLAIMER This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email From shreya_pathak rediffmail.com Mon Oct 18 10:01:50 2004 From: shreya_pathak rediffmail.com (Shreya Pathak) Date: Mon Oct 18 05:38:05 2004 Subject: [Mp4-tech] [Mpeg4 AAC] regarding PNS block in Mpeg4 AAC decoder Message-ID: <20041018090122.2665.qmail@webmail46.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041018/a58cbd33/attachment-0001.html -------------- next part -------------- Hi, In my reference Mpeg4 AAC LC decoder software, i am able to pass all the test vectors except PNS test vectors.The reference for me was the document for ISO. Is there any change in PNS pseudo code given in the document ISO/IEC CD 14496-3 Subpart 4. Regards Shreya ? On Fri, 15 Oct 2023 Dave Singer wrote : >At 5:41 PM +0530 10/14/04, Prashant Bhujang wrote: >>Hi, >> >>I wanted some clarification on MP4 file format.. >> >>Long back I had worked on mp4 file format, that was chapter 13 in 14496-1:2000. Now new standards like 14496-12(ISO media file format), 14496-14(MP4 file format) have been introduced. I checked with the latest version of 14496-1. The chapter on file format is still retained. >>I would like to know >>1. If both the forms of file formats specified in 14496-1 and 14496-14 are supported. > >yes, though the new is preferred and supersedes the old. > >>2. What do brands 'mp41' and 'mp42' indicate? > >mp41 is the format documented in 14496-1 chapter 13, although that format didn't have an ftyp atom. > >>3. Can AVC data can be supported in the file format specified in 14496-1 (where in Atoms are used) > >yes, there is a part 15 which documents AVC in ISO family files. Since in its simple mode it uses no structural changes to the file format, it could be argued that it's compatible with the 14496-1 document. > >> >>regards >>Prashant >> >>-- >>Prashant Bhujang >>Sasken Communication Technologies Ltd, >>139/25, Amar Jyothi Layout, Domlur PO >>Bangalore 560071 >>Tel: 25355501 Extn: 1124 >>email: prashant@sasken.com >> >> >> >> SASKEN BUSINESS DISCLAIMER >>This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email >>_______________________________________________ >>NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. >> >>Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > > >-- David Singer >Apple Computer/QuickTime >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From fx.parisot ateme.fr Mon Oct 18 12:50:09 2004 From: fx.parisot ateme.fr (FX Parisot) Date: Mon Oct 18 06:40:23 2004 Subject: [Mp4-tech] MPEG-4 Visual implementations References: <005b01c4b040$2b589e40$0a00000a@tototxoxqlsjoa> Message-ID: <42a701c4b4f7$dae6ff30$de1010ac@pc222> Hi, The Advanced Real Time Simple profile includes I-VOP, P-VOP / AC/DC prediction / 4- MV GOB Resync. / Data partitionning / RVLC As suggestion, all this is available in the implementation of Advanced simple profile from Ateme, on PC and DSP, for Full D1 real-time, for encoding and decoding. Other profiles may be developped on demand. See http://www.ateme.com/products/mpeg4.php FX Parisot ----- Original Message ----- From: "Olivier Amato" To: Sent: Tuesday, October 12, 2023 11:45 AM Subject: [Mp4-tech] MPEG-4 Visual implementations > I'm looking for available implementations of the following MPEG-4 Visual > Part 2 profiles ( encoders or/and decoders ) : > - Advanced Real Time Simple > - Simple Studio > - Core Studio > > Any suggestions ? > > Olivier > > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, > [video], [systems], [general] or another apppropriate identifier to > indicate the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust guidelines > found at > http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > From dipankar.mitra lgsoftindia.com Mon Oct 18 16:55:55 2004 From: dipankar.mitra lgsoftindia.com (Dipankar Mitra) Date: Mon Oct 18 06:40:58 2004 Subject: [Mp4-tech] [System]Audio video sync Message-ID: Thanks Bharat, Posting it to the mail list too... Regards, Dipankar ============================================= Embassy Icon , 7th Floor Infantry Road Shivajinagar Bangalore - 560001 Ph No. - 56938700 Ext : 177 Fax - 56938800 -----Original Message----- From: Bharat P. SONI [mailto:bharat.soni@st.com] Sent: Monday, October 18, 2023 3:44 PM To: Dipankar Mitra Subject: RE: [Mp4-tech] [System]Audio video sync Hi, I did use the szstem time to synchronize the audio-video. Following are the steps I used to sznchroniye. 1. Make sure that all your timeing calculations, like clock time, CTS, DTS are relative to common reference start time. 2. while decoding, check if the decoding time of the current access unit is less than current clock time. If yes then decode, otherwise discard that access unit. 2. store the decoded audio-video access units in queue along with their CTS. For video, 1. get the next pending picture with lowest CTS, but greater than the current system time. 2. Schedule the timer for that CTS. 3. When timer hits, send the picture to the renderer. For Audio 1. I worked on a system in which call back was being called whenever audio renderer finishes rendering the data in its buffer. So whenever the call back hits, check the CTS of audio access units in the queue and select the access unit having CTS closest to the current system time. 2. Send this access unit to the renderer. 3. remove all the access units having CTS less than current system time. I hope this helps. Regards, Bharat From gripened gmail.com Mon Oct 18 18:55:18 2004 From: gripened gmail.com (Jayant Chauhan) Date: Mon Oct 18 09:08:05 2004 Subject: [Mp4-tech] [System]Audio video sync In-Reply-To: References: Message-ID: <5b996acb041018052555e47630@mail.gmail.com> Dear Dipankar and Bharat 1) Thanks fellas, but had thought of that. The problem I faced with that approach was in implementing the clock, as in, if I use a TIMER, it becomes process specific. So it would be nice if you could tell as to how to implement a timer which runs on system time (if I have a thread which keeps calling GETSYSTEMTIME, it will slow down the whole system), so could you please suggest me the implementation of the clock itself :D 2) Another problem I am facing (due to unavailability of the ISO specs) is that in some of the standard ISO mpeg-4 streams provided, like v10-transition.mp4, there are alot of visual tracks and they open up in different windows when played in VLC player. Now how do I find out as to which audio track is associated with which video ( as in, how do I use the "tref" atom , assuming that is where I can get the info from) and also as to one basic doubt, which follows in the next query numbered 3 :) 3) In one moov, is it possible that that there be 2 video tracks !?? I have some files which show 2 video tracks , with bothhaving the same durations but one has only 1 sample and the other one is the main video track. What is this video track with just 1 sample number ?!! Is it just a JPEG (though not shown as a JPEG). thanking you in advance with regards Jayant On Mon, 18 Oct 2023 15:55:55 +0530, Dipankar Mitra wrote: > Thanks Bharat, > Posting it to the mail list too... > > Regards, > Dipankar > ============================================= > > Embassy Icon , 7th Floor > Infantry Road > Shivajinagar > Bangalore - 560001 > Ph No. - 56938700 Ext : 177 > Fax - 56938800 > > -----Original Message----- > From: Bharat P. SONI [mailto:bharat.soni@st.com] > Sent: Monday, October 18, 2023 3:44 PM > To: Dipankar Mitra > Subject: RE: [Mp4-tech] [System]Audio video sync > > Hi, > > I did use the szstem time to synchronize the audio-video. > Following are the steps I used to sznchroniye. > 1. Make sure that all your timeing calculations, like clock > time, CTS, DTS are relative to common reference start time. > > 2. while decoding, check if the decoding time of the current > access unit is less than current clock time. If yes then > decode, otherwise discard that access unit. > 2. store the decoded audio-video access units in queue along > with their CTS. > > For video, > 1. get the next pending picture with lowest CTS, but greater > than the current system time. > 2. Schedule the timer for that CTS. > 3. When timer hits, send the picture to the renderer. > > For Audio > 1. I worked on a system in which call back was being called > whenever audio renderer finishes rendering the data in its > buffer. So whenever the call back hits, check the CTS of > audio access units in the queue and select the access unit > having CTS closest to the current system time. > 2. Send this access unit to the renderer. > 3. remove all the access units having CTS less than current > system time. > > I hope this helps. > > Regards, > Bharat > > > > > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > From vpore pace.stpp.soft.net Mon Oct 18 20:03:15 2004 From: vpore pace.stpp.soft.net (Vinayak Pore) Date: Mon Oct 18 09:47:52 2004 Subject: [Mp4-tech] [System]Audio video sync In-Reply-To: <5b996acb041018052555e47630@mail.gmail.com> References: <5b996acb041018052555e47630@mail.gmail.com> Message-ID: <4173C61B.2080805@pace.stpp.soft.net> There two possible methods. 1. Keep a timer in the thread with High Priority with what ever best resolution / precision that OS can give. This thread just updates the system time. Other threads (Audio and Video Renderer) can decide when to submit the respective buffers to devices depending on the time stamp of the buffer which you get from file. Just a caution to keep the processing in the timer thread minimum in order not to load processor. In this method it is synchronized to the master clock as implemented by timer. 2. Second method is to synchronize to the audio buffer playout. So fill the audio buffer for corresponding time, and when this is played out the Message / Interrupt can be received. The video time can be decided based on this. This eliminates the precision timer and processor load for time keeping. hope this helps. -- Best regards, Vinayak. -------------- Jayant Chauhan wrote: >Dear Dipankar and Bharat >1) Thanks fellas, but had thought of that. The problem I faced with >that approach was in implementing the clock, as in, if I use a TIMER, >it becomes process specific. So it would be nice if you could tell as >to how to implement a timer which runs on system time (if I have a >thread which keeps calling GETSYSTEMTIME, it will slow down the whole >system), so could you please suggest me the implementation of the >clock itself :D > >2) Another problem I am facing (due to unavailability of the ISO >specs) is that in some of the standard ISO mpeg-4 streams provided, >like v10-transition.mp4, there are alot of visual tracks and they open >up in different windows when played in VLC player. Now how do I find >out as to which audio track is associated with which video ( as in, >how do I use the "tref" atom , assuming that is where I can get the >info from) and also as to one basic doubt, which follows in the next >query numbered 3 :) > >3) In one moov, is it possible that that there be 2 video tracks !?? I >have some files which show 2 video tracks , with bothhaving the same >durations but one has only 1 sample and the other one is the main >video track. What is this video track with just 1 sample number ?!! Is >it just a JPEG (though not shown as a JPEG). > >thanking you in advance > >with regards >Jayant > >On Mon, 18 Oct 2023 15:55:55 +0530, Dipankar Mitra > wrote: > > >>Thanks Bharat, >>Posting it to the mail list too... >> >>Regards, >>Dipankar >>============================================= >> >>Embassy Icon , 7th Floor >>Infantry Road >>Shivajinagar >>Bangalore - 560001 >>Ph No. - 56938700 Ext : 177 >>Fax - 56938800 >> >>-----Original Message----- >>From: Bharat P. SONI [mailto:bharat.soni@st.com] >>Sent: Monday, October 18, 2023 3:44 PM >>To: Dipankar Mitra >>Subject: RE: [Mp4-tech] [System]Audio video sync >> >>Hi, >> >>I did use the szstem time to synchronize the audio-video. >>Following are the steps I used to sznchroniye. >>1. Make sure that all your timeing calculations, like clock >>time, CTS, DTS are relative to common reference start time. >> >>2. while decoding, check if the decoding time of the current >>access unit is less than current clock time. If yes then >>decode, otherwise discard that access unit. >>2. store the decoded audio-video access units in queue along >>with their CTS. >> >>For video, >>1. get the next pending picture with lowest CTS, but greater >>than the current system time. >>2. Schedule the timer for that CTS. >>3. When timer hits, send the picture to the renderer. >> >>For Audio >>1. I worked on a system in which call back was being called >>whenever audio renderer finishes rendering the data in its >>buffer. So whenever the call back hits, check the CTS of >>audio access units in the queue and select the access unit >>having CTS closest to the current system time. >>2. Send this access unit to the renderer. >>3. remove all the access units having CTS less than current >>system time. >> >>I hope this helps. >> >>Regards, >>Bharat >> >> >> >> >>_______________________________________________ >>NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. >> >>Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php >> >> >> >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041018/d52c28e8/attachment.html From bharat.soni st.com Mon Oct 18 22:15:30 2004 From: bharat.soni st.com (Bharat P. SONI) Date: Mon Oct 18 12:24:08 2004 Subject: [Mp4-tech] [System]Audio video sync Message-ID: Hi, For the time being I will answer your query no 3. I also do not have the spec with me right now. I don?t recall whether there is any restriction on the moov. But you can have multiple video in a presentation. These videos are identified by their their ES ids and in turn OD Id. The relation ship among various objects, like audio corresponding to a video may be defined in the BIFS. I am not sure if this relationship is described at the IOD level. Regards, Bharat From amartin tsc.uc3m.es Mon Oct 18 19:17:36 2004 From: amartin tsc.uc3m.es (=?iso-8859-1?Q?=C1ngel_Mart=EDn_Navas?=) Date: Mon Oct 18 12:42:03 2004 Subject: [Mp4-tech] Pair Macroblocks and Slices in H.264 AVC Message-ID: <002401c4b52d$fafda860$659175a3@tsc.uc3m.es> Hi All, I have a question about pair macroblocks and slices in H.264. Can the top macroblock of a pair macroblocks belong to one slice and the bottom macroblock to another slice? In the next figure you can see the question (MbaffFrameFlag = 1). Can the macroblocks 0, 1, 2, 3 and 4 belong to slice 0 and the 5, 6, 7, 8, 9 to slice 1? ------------------------------------------------------------ | MB 0 | MB 2 | MB 4 | MB 6 | MB 8 | ------------------------------------------------------------ | MB 1 | MB 3 | MB 5 | MB 7 | MB 9 | ------------------------------------------------------------ Thanks for your help Angel From bharat.soni st.com Tue Oct 19 14:08:02 2004 From: bharat.soni st.com (Bharat P. SONI) Date: Tue Oct 19 10:46:53 2004 Subject: [Mp4-tech] [System]Audio video sync Message-ID: Hi, Just few inputs!! In the first proposal there is possibility of slight inaccurecy, but it may be acceptable. Second proposal seems to be more deterministic in nature as the interrupt from audio renderer have very high prioroty so you can update the system time here quite accurately. The problem I see is if due to some reason we are not able to send enough data to the audio renderer (may be due to data loss in the network) will this interrupt be called on time? If not then we will have incorrect system time. Another problem (also related to the problem above) is that in this case the system time is dependent on the audio. For presentatin that may not have audio, your system have no other option for clock. Regards, Bharat From bharat.soni st.com Tue Oct 19 14:17:38 2004 From: bharat.soni st.com (Bharat P. SONI) Date: Tue Oct 19 10:49:06 2004 Subject: [Mp4-tech] [System]Audio video sync Message-ID: <1a1179c8.919d8b4d.81c1300@mail2.dlh.st.com> Hi, In continuation with the last mail.... To be uptodate with the actual time, this timer has to be scheduled very frequenty (duration between the two schedule should be minimum to avoide inaccurecy). This is definetly going to increase the (unnacessary)load on the system. If you still want to reduce the load of the extra processing of the timer, I would suggest to use a function similar to GetSzstemTime() wheneever zou want to check the time. Regards, Bharat From singer apple.com Tue Oct 19 16:51:42 2004 From: singer apple.com (Dave Singer) Date: Tue Oct 19 10:50:56 2004 Subject: [Mp4-tech] MP4 File Format In-Reply-To: <41736115.9000902@sasken.com> References: <416E6CDD.4080408@sasken.com> <41736115.9000902@sasken.com> Message-ID: At 11:52 AM +0530 10/18/04, Prashant Bhujang wrote: >Thank you very much Dave for your response. > >Another clarification, by your explaination can I deduce that AVC >can be used within the file format specified by chapter 13 in >14496-1(which uses atoms and not boxes) and may be with new config >params defined in 14496-15. > >thank you very much, > >regards >Prashant the change from atom to box was just naming. the AVC file format is based on the ISO format which requires an FTYP box which was not present in the chapter 13 version. there are no other required elements in the AVC file format; you can do simple streams using the sample entries and formats for AVC from the AVC file format, and the structures you already know. > > >Dave Singer wrote: > >>At 5:41 PM +0530 10/14/04, Prashant Bhujang wrote: >> >>>Hi, >>> >>>I wanted some clarification on MP4 file format.. >>> >>>Long back I had worked on mp4 file format, that was chapter 13 in >>>14496-1:2000. Now new standards like 14496-12(ISO media file >>>format), 14496-14(MP4 file format) have been introduced. I >>>checked with the latest version of 14496-1. The chapter on file >>>format is still retained. >>>I would like to know >>>1. If both the forms of file formats specified in 14496-1 and >>>14496-14 are supported. >> >> >>yes, though the new is preferred and supersedes the old. >> >>>2. What do brands 'mp41' and 'mp42' indicate? >> >> >>mp41 is the format documented in 14496-1 chapter 13, although that >>format didn't have an ftyp atom. >> >>>3. Can AVC data can be supported in the file format specified in >>>14496-1 (where in Atoms are used) >> >> >>yes, there is a part 15 which documents AVC in ISO family files. >>Since in its simple mode it uses no structural changes to the file >>format, it could be argued that it's compatible with the 14496-1 >>document. >> >>> >>>regards >>>Prashant >>> >>>-- >>>Prashant Bhujang >>>Sasken Communication Technologies Ltd, >>>139/25, Amar Jyothi Layout, Domlur PO >>>Bangalore 560071 >>>Tel: 25355501 Extn: 1124 >>>email: prashant@sasken.com >>> >>> >>> >>> SASKEN BUSINESS DISCLAIMER >>>This message may contain confidential, proprietary or legally >>>Privileged information. In case you are not the original intended >>>Recipient of the message, you must not, directly or indirectly, >>>use, Disclose, distribute, print, or copy any part of this message >>>and you are requested to delete it and inform the sender. Any >>>views expressed in this message are those of the individual sender >>>unless otherwise stated. Nothing contained in this message shall >>>be construed as an offer or acceptance of any offer by Sasken >>>Communication Technologies Limited ("Sasken") unless sent with >>>that express intent and with due authority of Sasken. Sasken has >>>taken enough precautions to prevent the spread of viruses. However >>>the company accepts no liability for any damage caused by any >>>virus transmitted by this email >>>_______________________________________________ >>>NOTE: Please use clear subject lines for your posts. Include >>>[audio, [video], [systems], [general] or another apppropriate >>>identifier to indicate the type of question you have. >>> >>>Note: Conduct on the mailing list is subject to the Antitrust >>>guidelines found at >>>http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php >> >> >> > >-- >Prashant Bhujang >Sasken Communication Technologies Ltd, >139/25, Amar Jyothi Layout, Domlur PO >Bangalore 560071 >Tel: 25355501 Extn: 1124 >email: prashant@sasken.com > > > > SASKEN BUSINESS DISCLAIMER >This message may contain confidential, proprietary or legally >Privileged information. In case you are not the original intended >Recipient of the message, you must not, directly or indirectly, use, >Disclose, distribute, print, or copy any part of this message and >you are requested to delete it and inform the sender. Any views >expressed in this message are those of the individual sender unless >otherwise stated. Nothing contained in this message shall be >construed as an offer or acceptance of any offer by Sasken >Communication Technologies Limited ("Sasken") unless sent with that >express intent and with due authority of Sasken. Sasken has taken >enough precautions to prevent the spread of viruses. However the >company accepts no liability for any damage caused by any virus >transmitted by this email -- David Singer Apple Computer/QuickTime From vpore pace.stpp.soft.net Tue Oct 19 15:24:46 2004 From: vpore pace.stpp.soft.net (Vinayak Pore) Date: Tue Oct 19 10:52:43 2004 Subject: [Mp4-tech] [System]Audio video sync In-Reply-To: References: Message-ID: <4174D656.3040405@pace.stpp.soft.net> Hello, Bharat P. SONI wrote: >Hi, > >Just few inputs!! > >In the first proposal there is possibility of slight >inaccurecy, but it may be acceptable. > > This is dependent on the precision of the timer and thread priority. So it is dependent on OS support and fast response of OS. >Second proposal seems to be more deterministic in nature as >the interrupt from audio renderer have very high prioroty so >you can update the system time here quite accurately. The >problem I see is if due to some reason we are not able to >send enough data to the audio renderer (may be due to data >loss in the network) will this interrupt be called on time? >If not then we will have incorrect system time. > > This can be corrected. You will face same problem in every implementation. The solution is to send silence or repeat the part of the previous audio frame. >Another problem (also related to the problem above) is that >in this case the system time is dependent on the audio. For >presentatin that may not have audio, your system have no >other option for clock. > > Again this can be solved by sending silence of appropriate duration to audio codec. So Video Only file also can be played. If you keep the load on timer thread very low then total processor load can be brought down. I can not give you exact figures as the are very much dependent on OS, PRocessor and Clock and Size of video and decoding but processor will get some free time. >Regards, >Bharat > > > -- Best regards, Vinayak. -------------- PACE Soft Silicon Pvt. Ltd. From dipankar.mitra lgsoftindia.com Wed Oct 20 10:35:08 2004 From: dipankar.mitra lgsoftindia.com (Dipankar Mitra) Date: Wed Oct 20 04:45:03 2004 Subject: [Mp4-tech] [System]Audio video sync Message-ID: Bharat P. SONI wrote: >Second proposal seems to be more deterministic in nature as >the interrupt from audio renderer have very high prioroty so >you can update the system time here quite accurately. [Dipankar Mitra] Here, can the interrupt/callback/token of the audio renderer carry a timestamp? If this is possible, then there is no need to start unwieldy timers, or get_system_time() etc... hence your code becomes OS independent! If this is not possible, then what Bharat suggests above is possibly the next best thing to do. From vpore pace.stpp.soft.net Wed Oct 20 11:19:51 2004 From: vpore pace.stpp.soft.net (Vinayak Pore) Date: Wed Oct 20 04:46:34 2004 Subject: [Mp4-tech] [System]Audio video sync In-Reply-To: References: Message-ID: <4175EE6F.6030208@pace.stpp.soft.net> Dipankar Mitra wrote: >Bharat P. SONI wrote: > > > > >>Second proposal seems to be more deterministic in nature as >>the interrupt from audio renderer have very high prioroty so >>you can update the system time here quite accurately. >> >> >[Dipankar Mitra] Here, can the interrupt/callback/token of the audio renderer carry a timestamp? If this is possible, then there is no need to start unwieldy timers, or get_system_time() etc... hence your code becomes OS independent! If this is not possible, then what Bharat suggests above is possibly the next best thing to do. > > Timestamps can be derived. The just a word that there is nothing independent of OS! You may have to change the code to suit the OS methods best. All you can do is to add abstraction layer to hide OS dependency. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041020/01c0d9dc/attachment.html From sweet_sober_smart yahoo.co.in Wed Oct 20 08:32:49 2004 From: sweet_sober_smart yahoo.co.in (ghvf jyg) Date: Wed Oct 20 04:47:41 2004 Subject: [Mp4-tech] Mp4 File Format Message-ID: <20041020063249.13367.qmail@web8503.mail.in.yahoo.com> Dear Experts, I am working on Mp4 File Format, with Reference to ISO standard (MPEG-4 3rd Edition).Could you please clear my following doubts related to it. 1) In OD descriptors we have Es, Oci and Ipmp decriptors. Do these come in order or they can be present randomly. i.e is it necessary that we will have Oci Descriptor after Es Descr and Ipmp Descr after Oci Descr. 2) Where is ES_Id_Ref descriptor (tag = (0x0F)) present? i.e. what is its parent Descriptor ? 3) Profile Level Indication Index Descriptor (tag = 0x14) is present in DecoderConfig Descriptor, but where is Extended Profile Level Descriptor (Tag = 0x13) present? Thanks and Regards Rahul Yahoo! India Matrimony: Find your life partneronline. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041020/5cd934c7/attachment.html From ralph.sperschneider iis.fraunhofer.de Wed Oct 20 00:38:39 2004 From: ralph.sperschneider iis.fraunhofer.de (Ralph Sperschneider) Date: Wed Oct 20 04:49:03 2004 Subject: [Mp4-tech] Re: Reference Waveforms In-Reply-To: References: Message-ID: <4175895F.8080403@iis.fraunhofer.de> Shailesh Sakri wrote: >Hi, >I have downloaded the reference waveforms for conformance testing of MPEG 4 >AAC Decoder. >I found that the header information of these wav files are corrupted. I was >unable to open these files in Cooledit and even Matlab. > > The header is not corrupted, but your Cooledit does not support the wav extensible format (try the Windows media player, it will play the wav files). >I had also downloaed the mp4 files and decoded using fadd. I found that the >file sizes of fadd decoded file and reference wav are differing. > > This might be due to frames skipped by the decoder at the beginning or the end, or due to different wav header sizes, or due to the fact that faad wrote samples with 16 bit accuracy, while the reference waveforms provide 24 bit per sample. >Can anyone give me some pointers of why is this happening. If these are not >the correct reference waveforms then where can i get these? > > An answer to the latter question would require that you disclose the site you have downloaded the data. > SASKEN BUSINESS DISCLAIMER >This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email > > Please do not use those disclaimers while sending e-mails to open reflectors (unless you don't expect an answer, since noone will be sure that she/he is the original intended recipient). -- Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 Fraunhofer IIS | Fax: +49 9131 776 67 344 Am Wolfsmantel 33 | mailto:ralph.sperschneider@iis.fraunhofer.de D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ From ralph.sperschneider iis.fraunhofer.de Wed Oct 20 00:40:00 2004 From: ralph.sperschneider iis.fraunhofer.de (Ralph Sperschneider) Date: Wed Oct 20 04:50:16 2004 Subject: [Mp4-tech] [Mpeg4 AAC] regarding PNS block in Mpeg4 AAC decoder In-Reply-To: <20041018090122.2665.qmail@webmail46.rediffmail.com> References: <20041018090122.2665.qmail@webmail46.rediffmail.com> Message-ID: <417589B0.80401@iis.fraunhofer.de> Shreya Pathak wrote: >Hi, >In my reference Mpeg4 AAC LC decoder software, i am able to pass all the test vectors except PNS test vectors. > What exactly is your problem? -- Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 Fraunhofer IIS | Fax: +49 9131 776 67 344 Am Wolfsmantel 33 | mailto:ralph.sperschneider@iis.fraunhofer.de D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ From singer apple.com Wed Oct 20 18:17:14 2004 From: singer apple.com (Dave Singer) Date: Wed Oct 20 08:59:51 2004 Subject: [Mp4-tech] Mp4 File Format In-Reply-To: <20041020063249.13367.qmail@web8503.mail.in.yahoo.com> References: <20041020063249.13367.qmail@web8503.mail.in.yahoo.com> Message-ID: At 7:32 AM +0100 10/20/04, ghvf jyg wrote: >Dear Experts, > >I am working on Mp4 File Format, with Reference to ISO standard >(MPEG-4 3rd Edition).Could you please clear my following doubts >related to it. > >1) In OD descriptors we have Es, Oci and Ipmp decriptors. Do >these come in order or they can be present randomly. i.e is it >necessary that we will have Oci Descriptor after Es Descr and Ipmp >Descr after Oci Descr. > >2) Where is ES_Id_Ref descriptor (tag = (0x0F)) present? >i.e. what is its parent Descriptor ? This is used in the file format in OD streams where an ESDescriptor would normally be. It contains a track reference index, and the actual descriptor is in the sampleentry of the referenced track. > >3) Profile Level Indication Index Descriptor (tag = 0x14) is >present in DecoderConfig Descriptor, but where is Extended Profile >Level Descriptor (Tag = 0x13) present? > >Thanks and Regards > >Rahul > > > >Yahoo! >India Matrimony: Find your life partner >online. > >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, >[video], [systems], [general] or another apppropriate identifier to >indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust >guidelines found at >http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php -- David Singer Apple Computer/QuickTime -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041020/3cc57fc7/attachment.html From shreya_pathak rediffmail.com Wed Oct 20 10:31:31 2004 From: shreya_pathak rediffmail.com (Shreya Pathak) Date: Wed Oct 20 09:13:10 2004 Subject: [Mp4-tech] [Mpeg4 AAC] regarding PNS block in Mpeg4 AAC decoder Message-ID: <20041020093105.26985.qmail@webmail18.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041020/0e8910d6/attachment-0001.html -------------- next part -------------- Hi Ralph, ? I have developed reference software for Mpeg4 AAC LC Decoder and i have downloaded the test vectors from this site. http://www.tnt.uni-hannover.de/project/mpeg/audio/ and here i am able to pass all the test vectors except for PNS. al18_08.mp4 and al19_08.mp4 are not passing.I get different values as compared to the reference output given in al18_08.wav and al19_08.wav When I decoded the same test vectors using the publicly available source code faad, there also its giving a difference between values. So I am not able to compare also where am I wrong ? Regards Shreya On Wed, 20 Oct 2023 Ralph Sperschneider wrote : >Shreya Pathak wrote: > >>Hi, >>In my reference Mpeg4 AAC LC decoder software, i am able to pass all the test vectors except PNS test vectors. >> >What exactly is your problem? > >-- Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 >Fraunhofer IIS | Fax: +49 9131 776 67 344 >Am Wolfsmantel 33 | mailto:ralph.sperschneider@iis.fraunhofer.de >D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ > > >_______________________________________________ >NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > >Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From raosrr rediffmail.com Wed Oct 20 14:25:33 2004 From: raosrr rediffmail.com (soogoor ravinder rao) Date: Wed Oct 20 11:37:25 2004 Subject: [Mp4-tech] Hadamard transform Message-ID: <20041020132509.24235.qmail@webmail6.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041020/9b33497f/attachment.html -------------- next part -------------- HELLO SIR, IN INTRAFRAME, IF PRED TYPE IS 16X16, WE WILL PERFORM HADAMARD TRANSFORM ON DC BLOCK. IS THERE ANY REASON BEHIND THIS OPERATION. THANKING YOU SIR, RAVINDER.S From sweet_sober_smart yahoo.co.in Wed Oct 20 17:41:24 2004 From: sweet_sober_smart yahoo.co.in (ghvf jyg) Date: Wed Oct 20 12:57:18 2004 Subject: [Mp4-tech] Mp4 File Format Message-ID: <20041020154124.99466.qmail@web8501.mail.in.yahoo.com> Dear Experts, I am working on Mp4 File Format, with Reference to ISO standard (MPEG-4 3rd Edition).Could you please clear my following doubts related to it. Profile Level Indication Index Descriptor (tag = 0x14) is present in DecoderConfig Descriptor, but where is Extended Profile Level Descriptor (Tag = 0x13) present? Thanks and Regards Rahul Yahoo! India Matrimony: Find your life partneronline. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041020/45b05c61/attachment.html From sweet_sober_smart yahoo.co.in Wed Oct 20 17:39:15 2004 From: sweet_sober_smart yahoo.co.in (ghvf jyg) Date: Wed Oct 20 17:08:56 2004 Subject: [Mp4-tech] mp4 file format Message-ID: <20041020153915.54518.qmail@web8505.mail.in.yahoo.com> Dear Experts, I am working on Mp4 File Format, with Reference to ISO standard (MPEG-4 3rd Edition).Could you please clear my following doubts related to it. In OD descriptors we have Es, Oci and Ipmp decriptors. Do these come in order or they can be present randomly. i.e is it necessary that we will have Oci Descriptor after Es Descr and Ipmp Descr after Oci Descr. Thanks and Regards Rahul Yahoo! India Matrimony: Find your life partneronline. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041020/248a54f9/attachment.html From Ilan.Daniel Adimos.com Wed Oct 20 20:03:17 2004 From: Ilan.Daniel Adimos.com (Ilan Daniel) Date: Wed Oct 20 17:09:52 2004 Subject: [Mp4-tech] Hadamard transform Message-ID: <2A01AFD68E343242BA2C748A5E1A733D294DA2@jerry.Adimos.com> Dear RAVINDER, As I understand, the Hadamard Transform gives another level of compression efficiency by using the correlation that exist between the 16 DC components. This correlation exist only at 16x16 Intra prediction. in all other cases the prediction can be done separately for each 4x4, 8x8 smaller blocks and that destroy the correlation between the 4x4 blocks. ilan. -----Original Message----- From: mp4-tech-bounces@lists.mpegif.org [mailto:mp4-tech-bounces@lists.mpegif.org]On Behalf Of soogoor ravinder rao Sent: Wednesday, October 20, 2023 3:25 PM To: mp4-tech@lists.mpegif.org Subject: [Mp4-tech] Hadamard transform HELLO SIR, IN INTRAFRAME, IF PRED TYPE IS 16X16, WE WILL PERFORM HADAMARD TRANSFORM ON DC BLOCK. IS THERE ANY REASON BEHIND THIS OPERATION. THANKING YOU SIR, RAVINDER.S -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041020/951f50d3/attachment.html From amylu cmu.edu Wed Oct 20 15:47:49 2004 From: amylu cmu.edu (Amy, Mei-Hsuan Lu) Date: Wed Oct 20 17:10:24 2004 Subject: [Mp4-tech] How to fiter out I/P/B frame from a encoded video stream? Message-ID: <1193.128.237.245.73.1098298069.squirrel@128.237.245.73> Hi everyone, I am currently using Microsoft MPEG-4 Visual Reference Software to do some expriment. What i want to do is to "fiter out" I/P/B frame from the encoded video stream generated by the MS FGS encoder (so that I can play with different type of frame separately). I know I can locate the start of a I/B/P frame by using vop_start_code and vop_coding_type. But how can I know the frame length so that I know where is the end of frame? Thanks Amy Lu amylu@cmu.edu From bach noida.interrasystems.com Thu Oct 21 12:36:24 2004 From: bach noida.interrasystems.com (bach@noida.interrasystems.com) Date: Thu Oct 21 03:23:50 2004 Subject: [Mp4-tech] How to fiter out I/P/B frame from a encoded video stream? In-Reply-To: <1193.128.237.245.73.1098298069.squirrel@128.237.245.73> References: <1193.128.237.245.73.1098298069.squirrel@128.237.245.73> Message-ID: <2617.192.168.4.201.1098338784.squirrel@192.168.4.201> Hi Amy, I think there is no certain way to know frame lengths, as frame lengths vary due to a number of conditions. Hence, encoded stream will not contain anything to identify the end of vop. However, in certain cases some heuristics can be helpful. But caution must be taken for cases for erroneous streams, like a start code is corrupted. Thanks, Biswajit > Hi everyone, > > I am currently using Microsoft MPEG-4 Visual Reference Software to do some > expriment. What i want to do is to "fiter out" I/P/B frame from the > encoded video stream generated by the MS FGS encoder (so that I can play > with different type of frame separately). I know I can locate the start of > a I/B/P frame by using vop_start_code and vop_coding_type. But how can I > know the frame length so that I know where is the end of frame? > > Thanks > Amy Lu > amylu@cmu.edu > > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, > [video], [systems], [general] or another apppropriate identifier to > indicate the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust guidelines > found at > http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > From dipankar.mitra lgsoftindia.com Thu Oct 21 13:26:08 2004 From: dipankar.mitra lgsoftindia.com (Dipankar Mitra) Date: Thu Oct 21 04:07:44 2004 Subject: [Mp4-tech] Re: Reference Waveforms Message-ID: > The header is not corrupted, but your Cooledit does not support the wav extensible format (try the Windows media player, it will play the wav files). [Dipankar Mitra] One of these files (al00_08.wav, sent by Sailesh) has the compression code missing. BTW, win media player does play the other file, but it's just noise. >I had also downloaed the mp4 files and decoded using fadd. I found that the >file sizes of fadd decoded file and reference wav are differing. > > This might be due to frames skipped by the decoder at the beginning or the end, or due to different wav header sizes, or due to the fact that faad wrote samples with 16 bit accuracy, while the reference waveforms provide 24 bit per sample. >Can anyone give me some pointers of why is this happening. If these are not >the correct reference waveforms then where can i get these? > > An answer to the latter question would require that you disclose the site you have downloaded the data. > SASKEN BUSINESS DISCLAIMER >This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email > > Please do not use those disclaimers while sending e-mails to open reflectors (unless you don't expect an answer, since noone will be sure that she/he is the original intended recipient). -- Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 Fraunhofer IIS | Fax: +49 9131 776 67 344 Am Wolfsmantel 33 | mailto:ralph.sperschneider@iis.fraunhofer.de D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From rkoenen intertrust.com Thu Oct 21 05:22:07 2004 From: rkoenen intertrust.com (Rob Koenen) Date: Thu Oct 21 08:14:12 2004 Subject: FW: [Mp4-tech] How to fiter out I/P/B frame from a encoded videostream? Message-ID: Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: holi1.gif Type: image/gif Size: 1297 bytes Desc: holi1.gif Url : /pipermail/mp4-tech/attachments/20041021/237de9a1/holi1-0001.gif From amylu cmu.edu Thu Oct 21 13:32:04 2004 From: amylu cmu.edu (Amy, Meihsuan Lu) Date: Thu Oct 21 14:00:10 2004 Subject: [Mp4-tech] How to fiter out I/P/B frame from a encoded video stream? In-Reply-To: <417767C2.5080505@interrasystems.com> Message-ID: <200410211631.i9LGVj4c024726@smtp.andrew.cmu.edu> Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 1297 bytes Desc: not available Url : /pipermail/mp4-tech/attachments/20041021/66af0bf8/attachment.gif From chris.dunn scalatech.co.uk Thu Oct 21 22:10:08 2004 From: chris.dunn scalatech.co.uk (chris dunn) Date: Thu Oct 21 16:46:53 2004 Subject: [Mp4-tech] BSAC Audio Reference Code Bugs Message-ID: <001201c4b7a9$f70480c0$c42f989e@narcissus> Hi All I've been running a win32 build of the BSAC codec contained within the MPEG 4 Reference Code, 2001 Edition, available at: http://www.iso.ch/iso/en/ittf/PubliclyAvailableStandards/ISO_IEC_14496-5_2001_Software_Reference/audio.zip I'd be grateful if anyone can suggest solutions to the following problems: 1. The codec works ok for mono sources, but does not appear to operate for stereo signals. 2. The BSAC encoder appears to operate permanently in VBR mode, with the -vr switch having no effect, hence the decoder -r switch cannot be used to scale encoded bitrate. Is does not seem possible to operate the encoder in fixed bitrate mode. 3. Window switching enabled mode (-wsm 10) crashes the BSAC encoder, hence can only operate in long block mode (-wsm 0) with resulting prenoise artifacts for transient material. 4. The decoder rewrite will not decode BSAC bitstreams that successfully decode with the original decoder. Alternatively does anyone know of a more recent (and publicly available) version of the reference software ? Regards, Chris Dunn _________________________________ Scala Technology Ltd http://www.scalatech.co.uk +44 20 7267 1604 (direct) +44 87 0052 4491 (fax) From sparikh sarnoff.com Thu Oct 21 17:55:01 2004 From: sparikh sarnoff.com (Sandip Parikh) Date: Thu Oct 21 18:34:41 2004 Subject: [Mp4-tech][video] modulo time question Message-ID: <41782225.F6BC1AF2@sarnoff.com> Hello: I recently came across a MPEG-4 (SP L1) stream that did not always have modulo_time_base set to non-zero when the vop_time_increment wrapped around: vop_time_increment modulo_time 340 0 407 0 474 0 541 0 0 0 // modulo_time_base remains 0 67 0 134 0 201 0 Is this valid ? If so, how should a decoder interpret the presentation times ? Thank you, Regards, Sandip From ying.s.zhang intel.com Fri Oct 22 12:28:41 2004 From: ying.s.zhang intel.com (Zhang, Ying S) Date: Fri Oct 22 01:39:37 2004 Subject: [Mp4-tech] Question about mismatch between faad output and reference wave. Message-ID: <571ACEFD467F7749BC50E0A98C17CDD80528D06E@pdsmsx403> I use "faad.exe -b 2 -o outfilename infilename" to generate some 24bit pcm wave file. The output wave files seem quite different with reference stream. The maximum absolute value is much higher than 2-14 relative to full scale. I don't know what's wrong with that. The audio signal sounds alike. I calculate the each sample doing: " Sample = ((*(mmioinfoRef.pchNext+2))<<16)&0xFF0000 + (((*(mmioinfoRef.pchNext+1))<<8)&0xFF00)+(*mmioinfoRef.pchNext&0xFF); if(Sample>>23 ==1) Sample |= 0xFF000000; " Do I use the wrong configuration in faad.exe or do I calculate the sample value wrong? Or the difference is reasonable? For example, I use al_sbr_s_32_1.mp4 as input and the maximum absolute value is 2492181 while RMS is about 41.5 which is acceptable. Can any one give some hint? Thanks! Best Regards, Zhang Ying iNet: 8-752-1572 Tel: +86-21-52574545-1572 -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041022/00d16388/attachment.html From ralph.sperschneider iis.fraunhofer.de Thu Oct 21 17:04:35 2004 From: ralph.sperschneider iis.fraunhofer.de (Ralph Sperschneider) Date: Fri Oct 22 05:19:02 2004 Subject: [Mp4-tech] [Mpeg4 AAC] regarding PNS block in Mpeg4 AAC decoder In-Reply-To: <20041020093105.26985.qmail@webmail18.rediffmail.com> References: <20041020093105.26985.qmail@webmail18.rediffmail.com> Message-ID: <4177C1F3.1020303@iis.fraunhofer.de> Shreya Pathak wrote: >Hi Ralph, > I have developed reference software for Mpeg4 AAC LC Decoder and i have downloaded the test vectors from this site. >http://www.tnt.uni-hannover.de/project/mpeg/audio/ >and here i am able to pass all the test vectors except for PNS. > > You should download the sequences from here: ftp://mpaudconf:adif2mp4@ftp.iis.fraunhofer.de/ >al18_08.mp4 and al19_08.mp4 are not passing.I get different values as compared to the reference output given in al18_08.wav and al19_08.wav > When I decoded the same test vectors using the publicly available source code faad, there also its giving a difference between values. >So I am not able to compare also where am I wrong ? > > You should follow the conformance test procedure as outlined in ISO/IEC 14496-4. Note that special tests are defined for PNS testing. It should be obvious that a sample by sample comparison is meaningless if random generators are used on decoder site. Ralph -- Dipl.-Ing. Ralph Sperschneider | Phone: +49 9131 776 344 Fraunhofer IIS | Fax: +49 9131 776 67 344 Am Wolfsmantel 33 | mailto:ralph.sperschneider@iis.fraunhofer.de D 91058 Erlangen | http://www.iis.fraunhofer.de/amm/ From wujianseven hotmail.com Fri Oct 22 15:51:11 2004 From: wujianseven hotmail.com (=?gb2312?B?vOEgzuI=?=) Date: Fri Oct 22 05:20:28 2004 Subject: [Mp4-tech] about 'Inverse scanning processes and derivation processes for neighbours' Message-ID: hi, In JVT-G050,about 'Inverse scanning processes and derivation processes for neighbours',I don't understand .Why do not line-by-line scanning processes is done? and the word 'Inverse' is exact?why? thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041022/84102e89/attachment-0001.html From mbakker gmail.com Fri Oct 22 10:11:43 2004 From: mbakker gmail.com (Menno Bakker) Date: Fri Oct 22 05:21:29 2004 Subject: [Mp4-tech] Question about mismatch between faad output and reference wave. In-Reply-To: <571ACEFD467F7749BC50E0A98C17CDD80528D06E@pdsmsx403> References: <571ACEFD467F7749BC50E0A98C17CDD80528D06E@pdsmsx403> Message-ID: Hi, Yes, you're doing something wrong. You either forgot to compensate for the different decoder delay (2048 samples) or your code to get "sample" is wrong. Menno On Fri, 22 Oct 2023 11:28:41 +0800, Zhang, Ying S wrote: > > > > I use "faad.exe ?b 2 ?o outfilename infilename" to generate some 24bit pcm > wave file. > > The output wave files seem quite different with reference stream. > > The maximum absolute value is much higher than 2-14 relative to full scale. > > I don't know what's wrong with that. The audio signal sounds alike. > > I calculate the each sample doing: > > " > > Sample = ((*(mmioinfoRef.pchNext+2))<<16)&0xFF0000 + > (((*(mmioinfoRef.pchNext+1))<<8)&0xFF00)+(*mmioinfoRef.pchNext&0xFF); > > if(Sample>>23 ==1) Sample |= 0xFF000000; > > " > > Do I use the wrong configuration in faad.exe or do I calculate the sample > value wrong? > > Or the difference is reasonable? > > For example, I use al_sbr_s_32_1.mp4 as input and the maximum absolute value > is 2492181 while RMS is about 41.5 which is acceptable. > > > > Can any one give some hint? > > > > Thanks! > > > > Best Regards, > > > > > Zhang Ying > > iNet: 8-752-1572 > > Tel: +86-21-52574545-1572 > > > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, > [video], [systems], [general] or another apppropriate identifier to indicate > the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust guidelines > found at > http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > > From tma iis.fhg.de Fri Oct 22 10:27:29 2004 From: tma iis.fhg.de (Herbert Thoma) Date: Fri Oct 22 05:22:46 2004 Subject: [Mp4-tech][video] modulo time question In-Reply-To: <41782225.F6BC1AF2@sarnoff.com> References: <41782225.F6BC1AF2@sarnoff.com> Message-ID: <4178B661.30107@iis.fhg.de> Hi! Sandip Parikh schrieb: > Hello: > > I recently came across a MPEG-4 (SP L1) stream that did not always have > modulo_time_base set to non-zero when the vop_time_increment wrapped > around: > > vop_time_increment modulo_time > 340 0 > 407 0 > 474 0 > 541 0 > 0 0 // modulo_time_base > remains 0 > 67 0 > 134 0 > 201 0 > > Is this valid ? If so, how should a decoder interpret the presentation > times ? I think the only case where this would be valid is when a GOV header is in the bitstream directly before the VOP where the modulo_time_base remains 0. A GOV header does "reset" the modulo_time_base. Regards, Herbert. > Thank you, > > Regards, > > Sandip > > > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > -- Herbert Thoma Group Manager Video Multimedia Realtime Systems Department Fraunhofer IIS Am Wolfsmantel 33, 91058 Erlangen, Germany Phone: +49-9131-776-323 Fax: +49-9131-776-399 email: tma@iis.fhg.de www: http://www.iis.fhg.de/ From tsoligas teihal.gr Fri Oct 22 15:59:18 2004 From: tsoligas teihal.gr (nick) Date: Fri Oct 22 08:49:23 2004 Subject: [Mp4-tech] change detection mask Message-ID: <4178F616.1030109@teihal.gr> Dear all Has anybody developed in matlab a motion seqmentaion algorithm? (Moving Region extraction) I need a change detection mask for successive frames (images). Best regards Nick From anupkc01 yahoo.com Mon Oct 25 04:30:31 2004 From: anupkc01 yahoo.com (anup kc) Date: Mon Oct 25 07:16:58 2004 Subject: [Mp4-tech] Enhanced HE-AAC baseline version In-Reply-To: <4175895F.8080403@iis.fraunhofer.de> Message-ID: <20041025103031.87718.qmail@web53007.mail.yahoo.com> Hi all I have a doubt about baseline version of enhanced HE-AAC ( enhanced aac Plus). What is the maximum number of envelopes allowed in the baseline version. MPEG spec says it is 4 in the case of full version, and the 3GPP spec says that for the baseline version the maximum number of envelopes allowed is 1,but the MPEG standard does not mention anything about this in the explanation of the baseline version Could anyone of you please clarify With Regards -Anup --------------------------------- Do you Yahoo!? vote.yahoo.com - Register online to vote today! -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041025/15400205/attachment.html From jakelee graduate.shu.edu.cn Mon Oct 25 23:06:39 2004 From: jakelee graduate.shu.edu.cn (Jake Lee) Date: Mon Oct 25 15:24:38 2004 Subject: [Mp4-tech] Could you help me combine H.264-based FGS and bitstream switching? Message-ID: Dear Sir/Madam/Miss: I am very interested in the H.264-based FGS(Fine-Granular-Scalability), which came from MPEG-4 video codec standard, because it gives a flexible enhance-layer-quality increase over a variable channel bandwidth. And I am also very interested in bitstream switching derived from SP/SI frame, an important feature of H.264/AVC video codec standard to identically reconstruct the predictive frame even when different reference frames are used. Please give me some advice on them. If you have some c-coding reference software on the combination of H.264-based FGS and bitstream switching, that will be great. I appreciate that you will send me some source code to help me understand this outstanding concept, in comparison with the results of FGS only or that of bitstream switching only. I have download H.264/AVC reference software from http://iphome.hhi.de/suehring/tml/download/ , but there is no guide for the incorporation of FGS into JM(Joint Model) reference software. Thank you in advance. Best Regards, Yours sincerely, Jake Lee From garysull windows.microsoft.com Mon Oct 25 17:15:56 2004 From: garysull windows.microsoft.com (Gary Sullivan) Date: Tue Oct 26 02:00:12 2004 Subject: [Mp4-tech] Could you help me combine H.264-based FGS and bitstreamswitching? Message-ID: <91D7F2CEE3425A4A9D11311D09FCE2460B8F3B53@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com> Jake, Please note that no standard has been developed that includes combining FGS and H.264/MPEG-4 AVC. That is why you will not find software for doing it on the JVT reference software web site. (However, something along those lines is under consideration as a topic for future work in MPEG and VCEG.) Best Regards, Gary Sullivan +> -----Original Message----- +> From: mp4-tech-bounces@lists.mpegif.org +> [mailto:mp4-tech-bounces@lists.mpegif.org] On Behalf Of Jake Lee +> Sent: Monday, October 25, 2023 7:07 AM +> To: mp4-tech@lists.mpegif.org +> Subject: [Mp4-tech] Could you help me combine H.264-based +> FGS and bitstreamswitching? +> +> Dear Sir/Madam/Miss: +> +> I am very interested in the H.264-based +> FGS(Fine-Granular-Scalability), which came from MPEG-4 video +> codec standard, because it gives a flexible +> enhance-layer-quality increase over a variable channel +> bandwidth. And I am also very interested in bitstream +> switching derived from SP/SI frame, an important feature of +> H.264/AVC video codec standard to identically reconstruct +> the predictive frame even when different reference frames are used. +> +> Please give me some advice on them. +> +> If you have some c-coding reference software on the +> combination of H.264-based FGS and bitstream switching, that +> will be great. +> I appreciate that you will send me some source code to help +> me understand this outstanding concept, in comparison with +> the results of FGS only or that of bitstream switching only. +> +> I have download H.264/AVC reference software from +> http://iphome.hhi.de/suehring/tml/download/ +> , but there is no guide for the incorporation of FGS into +> JM(Joint Model) reference software. +> +> +> +> Thank you in advance. +> +> Best Regards, +> +> Yours sincerely, +> +> Jake Lee +> +> _______________________________________________ +> NOTE: Please use clear subject lines for your posts. Include +> [audio, [video], [systems], [general] or another +> apppropriate identifier to indicate the type of question you have. +> +> Note: Conduct on the mailing list is subject to the +> Antitrust guidelines found at +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant +> itrust.php +> From amylu cmu.edu Mon Oct 25 20:49:43 2004 From: amylu cmu.edu (Amy, Meihsuan Lu) Date: Tue Oct 26 02:03:34 2004 Subject: [Mp4-tech] Random drop data in MPEG4 encoded bitstream to simulate wireless media straming In-Reply-To: <200410251600.i9PG0pe7007184@lists1.magma.ca> Message-ID: <200410252349.i9PNnMEN030228@smtp.andrew.cmu.edu> Hi All, I am trying to do some wireless video straming experiment. I use a MPEG4 encoded bitstream created by Microsoft MPEG-4 Visual Reference. What I did is to randomly drop some parts of data in the bitstream (to simulate packet error in wireless). However, after I did so, the modified bitstream became NOT decodable (It caused MS FGS decoder crashed!!!). So did I miss any point here? Is this because MS FGS decoder not well written? Thanks a lot Amy Lu From prashant sasken.com Tue Oct 26 12:45:08 2004 From: prashant sasken.com (Prashant Bhujang) Date: Tue Oct 26 21:21:21 2004 Subject: [Mp4-tech] H.264 Queries Message-ID: <417DEB6C.6000906@sasken.com> Hi, I am trying to generate a fast forward file using a normal H.264 file. I am doing the following steps: 1. Pick IDR pictures at an interval such a way that playing those IDR pics gives the required fast forward feeling. 2. Make sure that for these selected picturesHRD analysis is OK. My question is, what all RBSP syntax elements change due to above process. As far as I think, we have to change the following: 1. Sequence Parameter sets to reflect new values of HRD parameters 2. Timing SEI messages. My understanding is also that Nothing in the picture parameter sets, picture/slice data needs to be changed. Please let me know if my understanding is correct. thanks Prashant SASKEN BUSINESS DISCLAIMER This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email From Danijel.Domazet zg.htnet.hr Tue Oct 26 13:11:45 2004 From: Danijel.Domazet zg.htnet.hr (D.Domazet) Date: Tue Oct 26 21:25:06 2004 Subject: [Mp4-tech][audio] High sampling rates in AAC - why? References: <41782225.F6BC1AF2@sarnoff.com> <4178B661.30107@iis.fhg.de> Message-ID: <004f01c4bb44$32edec00$0100a8c0@FREZA> Hi all, As we know, AAC supports high sampling rates, for example 88.2kHz and 96kHz. What is the purpose of this? If I was to test the encoder for quality at this sampling rates, how would I do that? Is down-sampling a must since humans can't hear this high freqs? Or AAC is used by bats and whales... Daniel From bpathak brookes.ac.uk Tue Oct 26 20:13:14 2004 From: bpathak brookes.ac.uk (Bhumin Pathak) Date: Tue Oct 26 21:27:19 2004 Subject: [Mp4-tech] Random drop data in MPEG4 encoded bitstream to simulate wireless media streaming In-Reply-To: <200410261605.i9QG0mWK011444@lists1.magma.ca> Message-ID: Hi there, MS FGS decoder is not made to recover from such errors which you are trying to introduce. In few other decoders frames with such errors and all frames dependent on it are dropped without being decoded and decoding starts again from next Intra-refresh frame. MS FGS decoder is not made to do so and it just crashes !! Bhumin Bhumin H. Pathak ============================== Communications Research Group. Oxford Brookes University, Gipsy Lane, Oxford OX3 0BP - U.K. -----Original Message----- From: mp4-tech-bounces@lists.mpegif.org [mailto:mp4-tech-bounces@lists.mpegif.org]On Behalf Of mp4-tech-request@lists.mpegif.org Sent: 26 October 2023 17:05 To: mp4-tech@lists.mpegif.org Subject: Mp4-tech Digest, Vol 15, Issue 27 Send Mp4-tech mailing list submissions to mp4-tech@lists.mpegif.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.mpegif.org/mailman/listinfo/mp4-tech or, via email, send a message with subject or body 'help' to mp4-tech-request@lists.mpegif.org You can reach the person managing the list at mp4-tech-owner@lists.mpegif.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Mp4-tech digest..." Today's Topics: 1. Could you help me combine H.264-based FGS and bitstream switching? (Jake Lee) 2. RE: Could you help me combine H.264-based FGS and bitstreamswitching? (Gary Sullivan) 3. Random drop data in MPEG4 encoded bitstream to simulate wireless media straming (Amy, Meihsuan Lu) ---------------------------------------------------------------------- Message: 1 Date: Mon, 25 Oct 2023 22:06:39 +0800 From: "Jake Lee" Subject: [Mp4-tech] Could you help me combine H.264-based FGS and bitstream switching? To: Message-ID: Content-Type: text/plain; charset="gb2312" Dear Sir/Madam/Miss: I am very interested in the H.264-based FGS(Fine-Granular-Scalability), which came from MPEG-4 video codec standard, because it gives a flexible enhance-layer-quality increase over a variable channel bandwidth. And I am also very interested in bitstream switching derived from SP/SI frame, an important feature of H.264/AVC video codec standard to identically reconstruct the predictive frame even when different reference frames are used. Please give me some advice on them. If you have some c-coding reference software on the combination of H.264-based FGS and bitstream switching, that will be great. I appreciate that you will send me some source code to help me understand this outstanding concept, in comparison with the results of FGS only or that of bitstream switching only. I have download H.264/AVC reference software from http://iphome.hhi.de/suehring/tml/download/ , but there is no guide for the incorporation of FGS into JM(Joint Model) reference software. Thank you in advance. Best Regards, Yours sincerely, Jake Lee ------------------------------ Message: 2 Date: Mon, 25 Oct 2023 16:15:56 -0700 From: "Gary Sullivan" Subject: RE: [Mp4-tech] Could you help me combine H.264-based FGS and bitstreamswitching? To: "Jake Lee" , Message-ID: <91D7F2CEE3425A4A9D11311D09FCE2460B8F3B53@WIN-MSG-10.wingroup.windeploy.ntde v.microsoft.com> Content-Type: text/plain; charset="us-ascii" Jake, Please note that no standard has been developed that includes combining FGS and H.264/MPEG-4 AVC. That is why you will not find software for doing it on the JVT reference software web site. (However, something along those lines is under consideration as a topic for future work in MPEG and VCEG.) Best Regards, Gary Sullivan +> -----Original Message----- +> From: mp4-tech-bounces@lists.mpegif.org +> [mailto:mp4-tech-bounces@lists.mpegif.org] On Behalf Of Jake Lee +> Sent: Monday, October 25, 2023 7:07 AM +> To: mp4-tech@lists.mpegif.org +> Subject: [Mp4-tech] Could you help me combine H.264-based +> FGS and bitstreamswitching? +> +> Dear Sir/Madam/Miss: +> +> I am very interested in the H.264-based +> FGS(Fine-Granular-Scalability), which came from MPEG-4 video +> codec standard, because it gives a flexible +> enhance-layer-quality increase over a variable channel +> bandwidth. And I am also very interested in bitstream +> switching derived from SP/SI frame, an important feature of +> H.264/AVC video codec standard to identically reconstruct +> the predictive frame even when different reference frames are used. +> +> Please give me some advice on them. +> +> If you have some c-coding reference software on the +> combination of H.264-based FGS and bitstream switching, that +> will be great. +> I appreciate that you will send me some source code to help +> me understand this outstanding concept, in comparison with +> the results of FGS only or that of bitstream switching only. +> +> I have download H.264/AVC reference software from +> http://iphome.hhi.de/suehring/tml/download/ +> , but there is no guide for the incorporation of FGS into +> JM(Joint Model) reference software. +> +> +> +> Thank you in advance. +> +> Best Regards, +> +> Yours sincerely, +> +> Jake Lee +> +> _______________________________________________ +> NOTE: Please use clear subject lines for your posts. Include +> [audio, [video], [systems], [general] or another +> apppropriate identifier to indicate the type of question you have. +> +> Note: Conduct on the mailing list is subject to the +> Antitrust guidelines found at +> http://www.mpegif.org/public/documents/vault/mp-out-30042-Ant +> itrust.php +> ------------------------------ Message: 3 Date: Mon, 25 Oct 2023 19:49:43 -0400 From: "Amy, Meihsuan Lu" Subject: [Mp4-tech] Random drop data in MPEG4 encoded bitstream to simulate wireless media straming To: Message-ID: <200410252349.i9PNnMEN030228@smtp.andrew.cmu.edu> Content-Type: text/plain; charset="us-ascii" Hi All, I am trying to do some wireless video straming experiment. I use a MPEG4 encoded bitstream created by Microsoft MPEG-4 Visual Reference. What I did is to randomly drop some parts of data in the bitstream (to simulate packet error in wireless). However, after I did so, the modified bitstream became NOT decodable (It caused MS FGS decoder crashed!!!). So did I miss any point here? Is this because MS FGS decoder not well written? Thanks a lot Amy Lu ------------------------------ _______________________________________________ Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php End of Mp4-tech Digest, Vol 15, Issue 27 **************************************** From snd codingtechnologies.com Tue Oct 26 22:57:11 2004 From: snd codingtechnologies.com (Andreas Schneider) Date: Tue Oct 26 21:29:32 2004 Subject: [Mp4-tech] Enhanced HE-AAC baseline version In-Reply-To: <20041025103031.87718.qmail@web53007.mail.yahoo.com> Message-ID: Hello Anup, mp4-tech-bounces@lists.mpegif.org wrote on 25/10/2023 12:30:31 PM: > Hi all > I have a doubt about baseline version of enhanced HE-AAC ( enhanced > aac Plus). What is the maximum number of envelopes allowed in the > baseline version. MPEG spec says it is 4 in the case of full > version, and the 3GPP spec says that for the baseline version the > maximum number of envelopes allowed is 1,but the MPEG standard does > not mention anything about this in the explanation of the baseline version Could you please point out to which part of the 3GPP specification you are referring? There is no difference in the number of envelopes between baseline and unrestricted parametric stereo. For both versions the maximum number of envelopes is 4. Any statement to the opposite is wrong and should be fixed. Best regards, Andreas Schneider > Could anyone of you please clarify > With Regards > -Anup > > Do you Yahoo!? > vote.yahoo.com - Register online to vote today! > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, > [video], [systems], [general] or another apppropriate identifier to > indicate the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust > guidelines found at http://www.mpegif.org/public/documents/vault/mp- > out-30042-Antitrust.php -- Andreas Schneider, Research Engineer Coding Technologies GmbH Deutschherrnstr. 15-19 90429 Nuernberg, Germany phone: +49 (0) 911 92891 -26 fax: +49 (0) 911 92891 -99 mailto:snd@CodingTechnologies.com From stream_guy yahoo.com Tue Oct 26 15:57:48 2004 From: stream_guy yahoo.com (video guy) Date: Tue Oct 26 21:31:05 2004 Subject: [Mp4-tech] Help with H.264 start code prefix Message-ID: <20041026215748.27941.qmail@web51304.mail.yahoo.com> Hello, I'm trying to use the start code prefix in H.264 but not sure how to go ahead with it. Here is my problem, I want to send the h.264 encoded byte stream (which is stored as test.264 file) from one system to another using a client server program in C, and, simulate a packet loss environment. At the reciever, some random data will be lost. But, i'm not able to run the decoder due to the lost data. Is there anyway I could make the decoder work ? I have no idea how to use the start code prefix. Any help in this regard would be greatly appreciated. Thank you aar. --------------------------------- Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041026/22527160/attachment.html From srq ieee.org Tue Oct 26 23:02:47 2004 From: srq ieee.org (S. R. Quackenbush) Date: Wed Oct 27 00:58:59 2004 Subject: [Mp4-tech][audio] High sampling rates in AAC - why? In-Reply-To: <004f01c4bb44$32edec00$0100a8c0@FREZA> Message-ID: Hi, One very good reason for running AAC at 96 kHz is that it interoperates with 96 kHz D/A converter, and these converters permit an analog anti-imaging filter with much lower slope (e.g. passband at 20 kHz, stopband at 48 kHz) and hence less phase distortion in the conversion. Schuyler Quackenbush --- Dr. Schuyler Quackenbush, Audio Research Labs 336 Park Ave, Suite 200, Scotch Plains, NJ 07076 office: 908 490 0700 srq@audioresearchlabs.com mobile: 908 612 9423 fax: 908 842 9151 www.audioresearchlabs.com > -----Original Message----- > From: mp4-tech-bounces@lists.mpegif.org > [mailto:mp4-tech-bounces@lists.mpegif.org]On Behalf Of D.Domazet > Sent: Tuesday, October 26, 2023 6:12 AM > To: mp4-tech@lists.mpegif.org > Subject: [Mp4-tech][audio] High sampling rates in AAC - why? > > > Hi all, > As we know, AAC supports high sampling rates, for example 88.2kHz > and 96kHz. > What is the purpose of this? If I was to test the encoder for quality at > this sampling rates, how would I do that? Is down-sampling a must since > humans can't hear this high freqs? Or AAC is used by bats and whales... > > Daniel > > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include > [audio, [video], [systems], [general] or another apppropriate > identifier to indicate the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust > guidelines found at > http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > From amylu cmu.edu Wed Oct 27 00:20:44 2004 From: amylu cmu.edu (Amy, Mei-Hsuan Lu) Date: Wed Oct 27 01:01:57 2004 Subject: [Mp4-tech] Re: IPB frames, maybe someone can explain a bit more in detail? Message-ID: <1330.128.237.231.134.1098847244.squirrel@128.237.231.134> Hi Stefan, I am working on a similar problem: MPEG4 streaming over lossy networks. Could you share wiht me the solutions of problmes you've posted? I would like to know: 1. How did you exactly deal with I/P/B frame parsing? Actually I am dealing with a raw encoded stream withour RTP encapsulated. 2. When you are doing this experiment, didn't you consider the case that header info (other than VoPs) get lost? 3. What you did is just filter out errored frame even thought only one or two packets data get lost? Does that cause a high frame error rate? any better idea to improve PSNR? I really appreciate your comments. Look forward to your response. Regards, Amy Lu PhD Student Carnegie Mellon University From dipankar.mitra lgsoftindia.com Wed Oct 27 10:59:55 2004 From: dipankar.mitra lgsoftindia.com (Dipankar Mitra) Date: Wed Oct 27 01:03:51 2004 Subject: [Mp4-tech][audio] High sampling rates in AAC - why? Message-ID: Daniel, while it is true that most humans can hear roughly up to 20KHz (some only 15KHz), however, it is not correct to say that these sampling rates are of no use. DVD audio uses 96KHz sampling rate, and has been found to be better appreciated by audio enthusiasts than normal CD quality (although this has been debated). I guess even the super audio CD format also uses higher than 44.1KHz. However, irrespective of the fact whether humans can hear the difference, these higher rates are very much in use, which is why AAC supports them. Some basic DSP web articles related to oversampling can explain why having a higher sampling rate is better when it comes to filtering, reconstruction etc. Regards, Dipankar ============================================= Embassy Icon , 7th Floor Infantry Road Shivajinagar Bangalore - 560001 Ph No. - 56938700 Ext : 177 Fax - 56938800 -----Original Message----- From: mp4-tech-bounces@lists.mpegif.org [mailto:mp4-tech-bounces@lists.mpegif.org]On Behalf Of D.Domazet Sent: Tuesday, October 26, 2023 3:42 PM To: mp4-tech@lists.mpegif.org Subject: [Mp4-tech][audio] High sampling rates in AAC - why? Hi all, As we know, AAC supports high sampling rates, for example 88.2kHz and 96kHz. What is the purpose of this? If I was to test the encoder for quality at this sampling rates, how would I do that? Is down-sampling a must since humans can't hear this high freqs? Or AAC is used by bats and whales... Daniel _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php From rkoenen intertrust.com Tue Oct 26 22:43:04 2004 From: rkoenen intertrust.com (Rob Koenen) Date: Wed Oct 27 11:14:50 2004 Subject: FW: [Mp4-tech] Help with H.264 start code prefix Message-ID: Forwarding bounced message. ________________________________ From: Madan [mailto:madank@interrasystems.com] Sent: Tuesday, 26 October 2023 21:16 To: mp4-tech@lists.mpegif.org Cc: video guy Subject: Re: [Mp4-tech] Help with H.264 start code prefix Hi Buddy !! At the decoder end, keep a buffer that accumulates the packets from the server. From this buffer extract the NAL packets. From the bitstream at the reciving end. Use this algo to extract the NAL units for(;;) { if next 24 bits are 0x000001 { startCodeFound = true break; } else { flush 8 bits } }// for(;;) if(true == startCodeFound) { //startcode found // Flush the start code found flush 24 bits //Now navigate up to next start code and put the in between stuff // in the nal structure. for(;;) { get next 24 bits & check if it equals to 0x000001 if(false == (next 24 bits == 000001)) { // search for pattern 0x000000 check if next 24 bits are 0x000000 if(false == result) { // copy the byte into the buffer copy one byte to the Nal unit } else { break; } } else { break; } }//for(;;) once u got the nal , decode it ... Hope this helps u :-) video guy wrote: Hello, I'm trying to use the start code prefix in H.264 but not sure how to go ahead with it. Here is my problem, I want to send the h.264 encoded byte stream (which is stored as test.264 file) from one system to another using a client server program in C, and, simulate a packet loss environment. At the reciever, some random data will be lost. But, i'm not able to run the decoder due to the lost data. Is there anyway I could make the decoder work ? I have no idea how to use the start code prefix. Any help in this regard would be greatly appreciated. Thank you aar. ________________________________ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. ________________________________ _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php -- Thanx & Regards !! from Madan Interra Systems India Pvt. Ltd. A10, Sec9,NOIDA Ph: 0120-2442273/4 Ext 324 -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041026/2c8dd0cf/attachment-0001.html From magnus.hoem popwire.com Wed Oct 27 08:49:57 2004 From: magnus.hoem popwire.com (Magnus Hoem) Date: Wed Oct 27 11:14:56 2004 Subject: [Mp4-tech][audio] High sampling rates in AAC - why? In-Reply-To: <004f01c4bb44$32edec00$0100a8c0@FREZA> References: <41782225.F6BC1AF2@sarnoff.com> <4178B661.30107@iis.fhg.de> <004f01c4bb44$32edec00$0100a8c0@FREZA> Message-ID: <08B98615-27DC-11D9-9D21-000A95C4E67A@popwire.com> Hi! I think you have confused two different values. It is true that humans can't hear much above 20 kHz, but that is not directly related to sampling frequency. The sampling frequency really just states how many times per second you sample the analog soundwave. This means that for 96 kHz sampling frequency you sample 96000 times a second and you can code audio with frequencies up to half of that (Nyqvist frequency or whatever it is called) without distortion. However audio at more "normal" frequencies can also be represented, and this with better quality since it is closer to the original soundwave. Best regards, Magnus On Oct 26, 2004, at 12:11, D.Domazet wrote: > Hi all, > As we know, AAC supports high sampling rates, for example 88.2kHz and > 96kHz. > What is the purpose of this? If I was to test the encoder for quality > at > this sampling rates, how would I do that? Is down-sampling a must since > humans can't hear this high freqs? Or AAC is used by bats and whales... > > Daniel > > _______________________________________________ > NOTE: Please use clear subject lines for your posts. Include [audio, > [video], [systems], [general] or another apppropriate identifier to > indicate the type of question you have. > > Note: Conduct on the mailing list is subject to the Antitrust > guidelines found at > http://www.mpegif.org/public/documents/vault/mp-out-30042- > Antitrust.php > > ********************************************** Popwire Technology Magnus Hoem Research Engineer magnus.hoem@popwire.com ?rsta?ngsv?gen 19 B SE-117 94 Stockholm, Sweden Phone: +46 8 579 116 00 Direct: +46 8 579 112 12 Mobile: +46 733 25 44 34 http://www.popwire.com *********************************************** This message, including any attachments may contain confidential and privileged material; it is intended only for the person to whom it is addressed. Its contents do not constitute a commitment by Popwire except where provided for in a written and undersigned agreement. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 2080 bytes Desc: not available Url : /pipermail/mp4-tech/attachments/20041027/77b5ce32/attachment.bin From dattagurubn yahoo.com Wed Oct 27 00:54:32 2004 From: dattagurubn yahoo.com (Dattaguru B.N.) Date: Wed Oct 27 11:15:03 2004 Subject: [Mp4-tech][audio] High sampling rates in AAC - why? In-Reply-To: Message-ID: <20041027065432.65405.qmail@web20102.mail.yahoo.com> Hi, I feel, high sampling rates would help us in better band widths. Higher the sampling rate, better will be the band width. If I am wrong, please correct. Thanks Datta --- "S. R. Quackenbush" wrote: > Hi, > > One very good reason for running AAC at 96 kHz is > that it interoperates with > 96 kHz D/A converter, and these converters permit an > analog anti-imaging > filter with much lower slope (e.g. passband at 20 > kHz, stopband at 48 kHz) > and hence less phase distortion in the conversion. > > Schuyler Quackenbush > --- > Dr. Schuyler Quackenbush, Audio Research Labs > 336 Park Ave, Suite 200, Scotch Plains, NJ 07076 > office: 908 490 0700 srq@audioresearchlabs.com > mobile: 908 612 9423 > fax: 908 842 9151 www.audioresearchlabs.com > > > > > -----Original Message----- > > From: mp4-tech-bounces@lists.mpegif.org > > [mailto:mp4-tech-bounces@lists.mpegif.org]On > Behalf Of D.Domazet > > Sent: Tuesday, October 26, 2023 6:12 AM > > To: mp4-tech@lists.mpegif.org > > Subject: [Mp4-tech][audio] High sampling rates in > AAC - why? > > > > > > Hi all, > > As we know, AAC supports high sampling rates, for > example 88.2kHz > > and 96kHz. > > What is the purpose of this? If I was to test the > encoder for quality at > > this sampling rates, how would I do that? Is > down-sampling a must since > > humans can't hear this high freqs? Or AAC is used > by bats and whales... > > > > Daniel > > > > _______________________________________________ > > NOTE: Please use clear subject lines for your > posts. Include > > [audio, [video], [systems], [general] or another > apppropriate > > identifier to indicate the type of question you > have. > > > > Note: Conduct on the mailing list is subject to > the Antitrust > > guidelines found at > > > http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > > > > > _______________________________________________ > NOTE: Please use clear subject lines for your posts. > Include [audio, [video], [systems], [general] or > another apppropriate identifier to indicate the type > of question you have. > > Note: Conduct on the mailing list is subject to the > Antitrust guidelines found at > http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > __________________________________ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail From garysull windows.microsoft.com Wed Oct 27 05:28:00 2004 From: garysull windows.microsoft.com (Gary Sullivan) Date: Wed Oct 27 11:15:08 2004 Subject: [Mp4-tech] H.264 Queries Message-ID: <91D7F2CEE3425A4A9D11311D09FCE24608D0629E@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com> You may also need to consider buffering period SEI messages. I think that would be everything -- just VUI, PT SEI, and BP SEI. -Gary ________________________________ From: mp4-tech-bounces@lists.mpegif.org on behalf of Prashant Bhujang Sent: Mon 10/25/2004 11:15 PM To: mp4-tech@lists.mpegif.org Subject: [Mp4-tech] H.264 Queries Hi, I am trying to generate a fast forward file using a normal H.264 file. I am doing the following steps: 1. Pick IDR pictures at an interval such a way that playing those IDR pics gives the required fast forward feeling. 2. Make sure that for these selected picturesHRD analysis is OK. My question is, what all RBSP syntax elements change due to above process. As far as I think, we have to change the following: 1. Sequence Parameter sets to reflect new values of HRD parameters 2. Timing SEI messages. My understanding is also that Nothing in the picture parameter sets, picture/slice data needs to be changed. Please let me know if my understanding is correct. thanks Prashant SASKEN BUSINESS DISCLAIMER This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041027/2a0495b6/attachment.html From stefan.goor ucd.ie Wed Oct 27 15:14:23 2004 From: stefan.goor ucd.ie (Stefan Goor) Date: Wed Oct 27 11:15:13 2004 Subject: [Mp4-tech] Re: IPB frames, maybe someone can explain a bit more in detail? Message-ID: <7900266.1098882863783.JavaMail.stefan.goor@ucd.ie> Hi Amy, Below are some answers / comments to your questions: >1. How did you exactly deal with I/P/B frame parsing? Actually I am dealing with a raw encoded stream withour RTP encapsulated. I,P and B VOPs can be identified by the occurence of a byte alligned startcode '0x01B6' in the MPEG-4 bitstream. Other useful VOP codes are '0x01B0' for the start of a visual object sequence and '0x01B3' for the start of a GOV (I think, I can't remember for certain). When you encounter the VOP startcode the subsequent 2 bits indicate if the VOP is I, P or B. 00 -> I-VOP 01 -> P-VOP 10 -> B-VOP 11 -> S-VOP (used for sprite coding but I don't think this relevant for your question, I just included it for completeness). >2. When you are doing this experiment, didn't you consider the case that header info (other than VoPs) get lost? In my experiements, I did not consider the case when the header was lost. In such cases the stream was cancelled and restarted. However, there are techniques available to avoid the loss of headers such as using FEC for header packets, duplicating the header within the stream or transmitting the header in out of band means such as RTSP (which is over TCP and so is reliable). >3. What you did is just filter out errored frame even thought only one or two packets data get lost? Does that cause a high frame error rate? any better idea to improve PSNR? The experiments I conducted did not use the error resilience (ER) features of MPEG-4 because their performance is subject to proprietary implementation. Without ER, the reference software decoder would frequently crash when incomplete frames were passed. Therefore only frames where all it's constituent packets had been received were passed to the decoder. If even a single packet was lost, then all the data for that frame was discarded and the next full frame was passed to the decoder. This did not have a significant impact on the number of frames lost / damaged because the bit rate constraints we used i.e. >65kbps, meant that few frames were larger than the MTU we used i.e. 1500 for a LAN. Typically only I-VOPs were segemented in multiple packets. What I have mentioned above applies to only the simple packetisation approach discussed in RFC 3016, the other Multi-SL and Multiplexed approaches I studied were slightly different. If you want further information about these approaches, let me know. As for improving the PSNR, one approach might be to examine the effects of ER, but you may have to consider a number of implementations because competing ER approaches may exhibit contrasting results. Hope this helps, Best Regards, Stefan From srq ieee.org Wed Oct 27 11:06:44 2004 From: srq ieee.org (S. R. Quackenbush) Date: Wed Oct 27 11:15:18 2004 Subject: [Mp4-tech][audio] High sampling rates in AAC - why? In-Reply-To: <20041027065432.65405.qmail@web20102.mail.yahoo.com> Message-ID: Re: wider bandwidth Theoretically correct, but in practice most systems will bandlimit output at 20 kHz. Schuyler --- Dr. Schuyler Quackenbush, Audio Research Labs 336 Park Ave, Suite 200, Scotch Plains, NJ 07076 office: 908 490 0700 srq@audioresearchlabs.com mobile: 908 612 9423 fax: 908 842 9151 www.audioresearchlabs.com > -----Original Message----- > From: Dattaguru B.N. [mailto:dattagurubn@yahoo.com] > Sent: Wednesday, October 27, 2023 2:55 AM > To: srq@ieee.org; D.Domazet; mp4-tech@lists.mpegif.org > Subject: RE: [Mp4-tech][audio] High sampling rates in AAC - why? > > > Hi, > > I feel, high sampling rates would help us in better > band widths. Higher the sampling rate, better will be > the band width. > If I am wrong, please correct. > > Thanks > Datta > > --- "S. R. Quackenbush" wrote: > > > Hi, > > > > One very good reason for running AAC at 96 kHz is > > that it interoperates with > > 96 kHz D/A converter, and these converters permit an > > analog anti-imaging > > filter with much lower slope (e.g. passband at 20 > > kHz, stopband at 48 kHz) > > and hence less phase distortion in the conversion. > > > > Schuyler Quackenbush > > --- > > Dr. Schuyler Quackenbush, Audio Research Labs > > 336 Park Ave, Suite 200, Scotch Plains, NJ 07076 > > office: 908 490 0700 srq@audioresearchlabs.com > > mobile: 908 612 9423 > > fax: 908 842 9151 www.audioresearchlabs.com > > > > > > > > > -----Original Message----- > > > From: mp4-tech-bounces@lists.mpegif.org > > > [mailto:mp4-tech-bounces@lists.mpegif.org]On > > Behalf Of D.Domazet > > > Sent: Tuesday, October 26, 2023 6:12 AM > > > To: mp4-tech@lists.mpegif.org > > > Subject: [Mp4-tech][audio] High sampling rates in > > AAC - why? > > > > > > > > > Hi all, > > > As we know, AAC supports high sampling rates, for > > example 88.2kHz > > > and 96kHz. > > > What is the purpose of this? If I was to test the > > encoder for quality at > > > this sampling rates, how would I do that? Is > > down-sampling a must since > > > humans can't hear this high freqs? Or AAC is used > > by bats and whales... > > > > > > Daniel > > > > > > _______________________________________________ > > > NOTE: Please use clear subject lines for your > > posts. Include > > > [audio, [video], [systems], [general] or another > > apppropriate > > > identifier to indicate the type of question you > > have. > > > > > > Note: Conduct on the mailing list is subject to > > the Antitrust > > > guidelines found at > > > > > > http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > > > > > > > > > _______________________________________________ > > NOTE: Please use clear subject lines for your posts. > > Include [audio, [video], [systems], [general] or > > another apppropriate identifier to indicate the type > > of question you have. > > > > Note: Conduct on the mailing list is subject to the > > Antitrust guidelines found at > > > http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php > > > > > > > __________________________________ > Do you Yahoo!? > Yahoo! Mail Address AutoComplete - You start. We finish. > http://promotions.yahoo.com/new_mail > From rlei ati.com Wed Oct 27 14:49:48 2004 From: rlei ati.com (Ryan Lei) Date: Wed Oct 27 14:49:17 2004 Subject: [Mp4-tech] [System][audio] Message-ID: Hi, all Does anybody know where I can find 3gpp2 files with QCELP, EVRC, SMV audio data in it? QuickTime Pro can only create 3gpp2 file with QCELP information inside mp4a box, but no other 2 codec support. Thanks Ryan Lei, Ph.D Handheld Products Group | ATI Technologies Inc. | 905.882.2600x2172 | www.ati.com -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041027/e6a8197a/attachment.html From jlee nextreaming.com Thu Oct 28 10:04:29 2004 From: jlee nextreaming.com (Jae-Yong Lee) Date: Thu Oct 28 01:55:21 2004 Subject: [Mp4-tech] [System][audio] Message-ID: Try Xenon at www.nextreaming.com which support 3gpp2 as well as 3gpp. Jay ________________________________ From: mp4-tech-bounces@lists.mpegif.org on behalf of Ryan Lei Sent: Thu 2023-10-28 02:49 To: mp4-tech@lists.mpegif.org Subject: [Mp4-tech] [System][audio] Hi, all Does anybody know where I can find 3gpp2 files with QCELP, EVRC, SMV audio data in it? QuickTime Pro can only create 3gpp2 file with QCELP information inside mp4a box, but no other 2 codec support. Thanks Ryan Lei, Ph.D Handheld Products Group | ATI Technologies Inc. | 905.882.2600x2172 | www.ati.com From lhq haiersoft.com.cn Thu Oct 28 10:55:22 2004 From: lhq haiersoft.com.cn (=?gb2312?B?u6rG5g==?=) Date: Thu Oct 28 01:58:20 2004 Subject: [Mp4-tech] Help! how can i imiprove my video profile and level Message-ID: <200410280155.i9S1tJHA025049@lists1.magma.ca> hello every one, Now i use the mpeg4ip to encode my raw v/a, my video format is *.avi with 112*128, and the profile@level is simple@L1, i use the ffmpeg -s 4cif to improve my video resoultion, but the pix's granularity is so big and the quality is much poor, and i try to improve my videm's profile and level, howvever i don't know how to do it. and if i improve the profile and level, whethe my video quality can improve or not? in a word, my question is how to improve my video resoultion and don't reduce the video quality so much. Regards. Eric ¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡lhq@haiersoft.com.cn ¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡¡2004-10-28 From prashant sasken.com Thu Oct 28 10:11:21 2004 From: prashant sasken.com (Prashant Bhujang) Date: Thu Oct 28 01:59:36 2004 Subject: [Mp4-tech] Timestamp of AU Message-ID: <41806A61.8040002@sasken.com> Hi, How do I find out the timestamp for an AU in H.264? thanks Prashant -- Prashant Bhujang Sasken Communication Technologies Ltd, 139/25, Amar Jyothi Layout, Domlur PO Bangalore 560071 Tel: 25355501 Extn: 1124 email: prashant@sasken.com SASKEN BUSINESS DISCLAIMER This message may contain confidential, proprietary or legally Privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, Disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email From anupkc01 yahoo.com Wed Oct 27 22:26:52 2004 From: anupkc01 yahoo.com (anup kc) Date: Thu Oct 28 02:01:08 2004 Subject: [Mp4-tech] Enhanced HE-AAC baseline version In-Reply-To: Message-ID: <20041028042652.91312.qmail@web53002.mail.yahoo.com> >Could you please point out to which part of the 3GPP specification you are >referring? Hi Andreas I was referring to the Encoder Spec doc- 3GPP TS 26.405 V1.0.0 (2004-05)- where in page 16 it said that "frame_class – Is set to 0. num_env_idx – Is set to 0 if all values in the iid and icc vectors after quantization are unchanged since previous frame. Is set to 1 for all other cases." If the maximum value of num_env_idx is 1 with frameclass always set to 0 ,doesnt it mean that the maximum number of envelopes is 1 ? -Anup Andreas Schneider wrote: Hello Anup, --------------------------------- Do you Yahoo!? Take Yahoo! Mail with you! Get it on your mobile phone. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041027/f7bc7826/attachment.html From kexu ee.cuhk.edu.hk Thu Oct 28 14:46:12 2004 From: kexu ee.cuhk.edu.hk (Xu Ke) Date: Thu Oct 28 11:28:26 2004 Subject: [Mp4-tech] [video]the CAVLC code design in H.264 Message-ID: <200410280545.i9S5jLQk029593@cuees8.ee.cuhk.edu.hk> Hi, I want to download the documents which are all refered in the white paper(H.264 / MPEG-4 Part 10 White Paper Variable-Length Coding). Where can I download the document JVT Document JVT-C028, G.Bjontegaard and K. Lillevold, ¡°Context-Adaptive VLC Coding of Coefficients¡±, Fairfax, VA, May 2002? I can not access it from the site ftp://ftp.imtc-files.org/jvt-experts/ Best regards, Xu Ke From rli accfast.com.tw Thu Oct 28 15:09:59 2004 From: rli accfast.com.tw (Richard Li) Date: Thu Oct 28 11:31:00 2004 Subject: [Mp4-tech] AAC: fill_element() Message-ID: Hi, According to ISO/IEC 13818-7:2003(E) Table 26 fill_element() { cnt = count; if(cnt == 15) { cnt += esc_count -1; } ... } Can anyone tell me why cnt value gets increased with esc_count if cnt equals 15? Best Regards, Richard T. J. Li From vinayak.shivaram rediffmail.com Thu Oct 28 10:14:58 2004 From: vinayak.shivaram rediffmail.com (Vinayak Shivaram) Date: Thu Oct 28 11:32:51 2004 Subject: [Mp4-tech] [video] video packets Message-ID: <20041028091429.25314.qmail@webmail18.rediffmail.com> An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041028/848ffa08/attachment.html -------------- next part -------------- If i have read the standard rightly when the hec is '1' in the video packet header the VOP header is repeated again this is done for error resilience i think i.e. we have two vop headers is there any way of knowing if the vop header(say any one!)corrupted if so which header do i retain. From prashanthp gmail.com Thu Oct 28 20:31:17 2004 From: prashanthp gmail.com (Prashanth P) Date: Thu Oct 28 11:34:49 2004 Subject: [Mp4-tech] Bit rate and compression Ratio Message-ID: <71c3c4ac0410280701721729ea@mail.gmail.com> Hi, Questions: 1) what is the difference between data rate and compression ratio? 2) How to calculate the MIPS rate given the Processor Clock Frequency? Rgds Prashanth From snd codingtechnologies.com Thu Oct 28 18:38:43 2004 From: snd codingtechnologies.com (Andreas Schneider) Date: Thu Oct 28 11:55:05 2004 Subject: [Mp4-tech] Enhanced HE-AAC baseline version In-Reply-To: <20041028042652.91312.qmail@web53002.mail.yahoo.com> Message-ID: Hi Anup, anup kc wrote on 28/10/2023 06:26:52 AM: > >Could you please point out to which part of the 3GPP specification you are > >referring? > Hi Andreas > > I was referring to the Encoder Spec doc- 3GPP TS 26.405 V1.0.0 > (2004-05)- where in page 16 it said that > "frame_class ? Is set to 0. > num_env_idx ? Is set to 0 if all values in the iid and icc vectors > after quantization are unchanged since previous frame. > Is set to 1 for all other cases." > If the maximum value of num_env_idx is 1 with frameclass always set > to 0 ,doesnt it mean that the maximum number of envelopes is 1 ? This is correct. However, this is just the specification of the reference encoder. It points out that this particular encoder is internally limited to 1 envelope. Hence this is not a normative statement that only 1 ps-envelope shall be used. In fact, since baseline-ps and the unrestricted version of ps share the same bitstream format, a compliant encoder may also signal features that go beyond the capabilities of a baseline decoder (like IPD/OPD). So the limitation to baseline just places restrictions on decoder behaviour (i.e. disregard IPD/OPD-data, map 34 stereo bands to 20). I hope this helps, Andreas > > -Anup > > > Andreas Schneider wrote: > Hello Anup, > Do you Yahoo!? > Take Yahoo! Mail with you! Get it on your mobile phone. -- Andreas Schneider, Research Engineer Coding Technologies GmbH Deutschherrnstr. 15-19 90429 Nuernberg, Germany phone: +49 (0) 911 92891 -26 fax: +49 (0) 911 92891 -99 mailto:snd@CodingTechnologies.com From garysull windows.microsoft.com Thu Oct 28 11:53:02 2004 From: garysull windows.microsoft.com (Gary Sullivan) Date: Thu Oct 28 15:25:54 2004 Subject: [Mp4-tech] [video]the CAVLC code design in H.264 Message-ID: <91D7F2CEE3425A4A9D11311D09FCE24608D062A9@WIN-MSG-10.wingroup.windeploy.ntdev.microsoft.com> The JVT ftp site has moved to standards.polycom.com. You can find the document there (in a directory with a name something like 2002_05_Fairfax). Best Regards, Gary Sullivan ________________________________ From: mp4-tech-bounces@lists.mpegif.org on behalf of Xu Ke Sent: Thu 10/28/2004 1:46 PM To: mp4-tech Subject: [Mp4-tech] [video]the CAVLC code design in H.264 Hi, I want to download the documents which are all refered in the white paper(H.264 / MPEG-4 Part 10 White Paper Variable-Length Coding). Where can I download the document JVT Document JVT-C028, G.Bjontegaard and K. Lillevold, "Context-Adaptive VLC Coding of Coefficients", Fairfax, VA, May 2002? I can not access it from the site ftp://ftp.imtc-files.org/jvt-experts/ Best regards, Xu Ke _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041028/88ba8edd/attachment.html From stream_guy yahoo.com Thu Oct 28 11:59:16 2004 From: stream_guy yahoo.com (video guy) Date: Thu Oct 28 15:29:01 2004 Subject: [Mp4-tech] Help with H.264 start code prefix In-Reply-To: <417F210F.9060309@interrasystems.com> Message-ID: <20041028175916.14939.qmail@web51309.mail.yahoo.com> Hi Madan, Thanks for the reply. That definitely gave some insight. You mentioned to use a buffer that accumulates the packets from the server and extract the NAL units. I am writing the sent data into a file at the reciever side, and giving it as an input to the decoder. And, I am not sure about the byte stream structure of the encoded video. Can you please throw some light on that. I have been using the JM73 version. I am very new to this, so will be having a lot of questions and hope to get answers from you. :) Thanks a bunch. aar. Madan wrote: Hi Buddy !! At the decoder end, keep a buffer that accumulates the packets from the server. From this buffer extract the NAL packets. From the bitstream at the reciving end. Use this algo to extract the NAL units for(;;) { if next 24 bits are 0x000001 { startCodeFound = true break; } else { flush 8 bits } }// for(;;) if(true == startCodeFound) { //startcode found // Flush the start code found flush 24 bits //Now navigate up to next start code and put the in between stuff // in the nal structure. for(;;) { get next 24 bits & check if it equals to 0x000001 if(false == (next 24 bits == 000001)) { // search for pattern 0x000000 check if next 24 bits are 0x000000 if(false == result) { // copy the byte into the buffer copy one byte to the Nal unit } else { break; } } else { break; } }//for(;;) once u got the nal , decode it ... Hope this helps u :-) video guy wrote: Hello, I'm trying to use the start code prefix in H.264 but not sure how to go ahead with it. Here is my problem, I want to send the h.264 encoded byte stream (which is stored as test.264 file) from one system to another using a client server program in C, and, simulate a packet loss environment. At the reciever, some random data will be lost. But, i'm not able to run the decoder due to the lost data. Is there anyway I could make the decoder work ? I have no idea how to use the start code prefix. Any help in this regard would be greatly appreciated. Thank you aar. --------------------------------- Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. --------------------------------- _______________________________________________NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have.Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php -- Thanx & Regards !! from Madan Interra Systems India Pvt. Ltd. A10, Sec9,NOIDA Ph: 0120-2442273/4 Ext 324 --------------------------------- Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041028/899b6c29/attachment.html From amylu cmu.edu Fri Oct 29 13:07:01 2004 From: amylu cmu.edu (Amy, Meihsuan Lu) Date: Fri Oct 29 13:04:32 2004 Subject: [Mp4-tech] Re: Random drop data in MPEG4 encoded bitstream to simulate wirelessmedia streaming Message-ID: <200410291606.i9TG6cEN004231@smtp.andrew.cmu.edu> So anyone knows any decoder that can tolerate random packet loss? (at least not crash!) Thanks. > ------------------------------ > Date: Tue, 26 Oct 2023 19:13:14 +0100 > From: "Bhumin Pathak" > Subject: [Mp4-tech] Random drop data in MPEG4 encoded bitstream to > simulate wireless media streaming > To: > Message-ID: > Content-Type: text/plain; charset="us-ascii" > > Hi there, > > MS FGS decoder is not made to recover from such errors which you are > trying to introduce. In few other decoders frames with such errors and > all frames dependent on it are dropped without being decoded and > decoding starts again from next Intra-refresh frame. MS FGS decoder is > not made to do so and it just crashes !! > > Bhumin > Bhumin H. Pathak > > ============================== > Communications Research Group. > Oxford Brookes University, > Gipsy Lane, Oxford > OX3 0BP - U.K. > > > > From GRAKURAM maxis.com.my Sat Oct 30 01:17:34 2004 From: GRAKURAM maxis.com.my (Rakuram Gandhi) Date: Fri Oct 29 13:08:07 2004 Subject: [Mp4-tech] Re: Mp4-tech Digest, Vol 15, Issue 31 Message-ID: Please be informed that I will be on leave from 30 October 2023 to 8 November 2004. (Both days inclusive). During my absence, please contact Shirley (x7693) for urgent matters. Thank you, Rakuram From Dakshinamoorthy.R lntinfotech.com Sat Oct 30 12:44:09 2004 From: Dakshinamoorthy.R lntinfotech.com (Dakshinamoorthy R) Date: Sat Oct 30 13:00:50 2004 Subject: [Mp4-tech] Bit rate and compression Ratio Message-ID: Dear Prasanth, 1) what is the difference between data rate and compression ratio? Ans: data rate is bits/sec say 64kbps ---> 64 kilo bits per second compression ratio: ratio between input file size of encoder to output file size of encoder ( like 4:1, 11:1 etc..) If your compression ratio is higher you can transmit, your data in low bitrate channel( mobile application need high compression ratio since band with is less) high compression is always trade-off with quality (more compression will decrease the quality). 2) How to calculate the MIPS rate given the Processor Clock Frequency? Ans: if your DSP processor execute one instruction per clock cycle and clock frequency is 300MHz then the MIPS for that processor is 1 x 300 = 300 mips some commercially available leading processors supports 8 instruction execution per clock cycle in that case the MIPS of the processor will be 8 x 300 = 2400 MPIS (assuming Processor clock freq is 300 Mhz) Regards, R.Dakshinamoorthy Project Leader, DSP- Group, Embedded System, L&T Infotech Ltd, Chennai. INDIA Phone +91 44 22527072 Fax +91 44 22523514 Prashanth P Sent by: mp4-tech-bounces@lists.mpegif.org 28/10/2023 07:31 PM Please respond to Prashanth P To: mp4-tech@lists.mpegif.org cc: Subject: [Mp4-tech] Bit rate and compression Ratio Hi, Questions: 1) what is the difference between data rate and compression ratio? 2) How to calculate the MIPS rate given the Processor Clock Frequency? Rgds Prashanth _______________________________________________ NOTE: Please use clear subject lines for your posts. Include [audio, [video], [systems], [general] or another apppropriate identifier to indicate the type of question you have. Note: Conduct on the mailing list is subject to the Antitrust guidelines found at http://www.mpegif.org/public/documents/vault/mp-out-30042-Antitrust.php ______________________________________________________________________ ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041030/279f82c1/attachment.html From u8921053 mail.ndhu.edu.tw Sat Oct 30 18:15:59 2004 From: u8921053 mail.ndhu.edu.tw (Alan) Date: Sat Oct 30 13:05:06 2004 Subject: [Mp4-tech] where can get the infomation about the specifics of mpeg-4 for animation? Message-ID: <000801c4be61$213df430$6700a8c0@alan> any one knows the specifics of MPEG-4 not the detail.. I have search that from ISO,but still no idea. I only want to know how much joints ( bones..brabrabra) it store not how it store,but what it store. i want store some animation data for education research. if someone know that, can tell me please. i will be very appreciate. Alan. -------------- next part -------------- An HTML attachment was scrubbed... URL: /pipermail/mp4-tech/attachments/20041030/ab9a9baf/attachment.html From bhargo netapp.com Sat Oct 30 09:31:55 2004 From: bhargo netapp.com (Sunil Bhargo) Date: Sat Oct 30 13:06:53 2004 Subject: [Mp4-tech] stdp and stsh atom/box Message-ID: Hi, I have following questions regarding the stdp and stsh atom/box ( i don't have the specifications) :- 1. If the file has both the stss and stsh atom/box and the operation is seek then how does the server decide which frame to honour (to be sent) from the stsh and stss ? Or is it that when both the atom/box are present then always stsh, sync_sample_number is sent. 2. In the case of stdp what does the value of priority value indicate, does it indicate the priority in which the frames will be dropped in case of thinning or is it something else ? Any clarification will be higly useful. thanks in advance, sunil bhargo Conventional opinion is the ruin of our souls - Mevlana Rumi