The presentation timestamp PTS is a timestamp metadata field in an MPEG transport stream or MPEG program stream that is used to achieve synchronization of programs' separate elementary streams for example Video, Audio, Subtitles when presented to the viewer. Presentation time stamps have a resolution of 90kHz, suitable for the presentation synchronization task.
The PCR or SCR has a resolution of 27MHz which is suitable for synchronization of a decoder's overall clock with that of the usual remote encoder, including driving TV signals such as frame and line sync timing, colour sub carrier, etc.
Decoding of N elementary streams is synchronized by adjusting the decoding of streams to a common master time base rather than by adjusting the decoding of one stream to match that of another. A transport stream may contain multiple programs and each program may have its own time base. The time bases of different programs within a transport stream may be different. Because PTSs apply to the decoding of individual elementary streams, they reside in the PES packet layer of both the transport streams and program streams.
End-to-end synchronization occurs when encoders save time stamps at capture time, when the time stamps propagate with associated coded data to decoders, and when decoders use those time stamps to schedule presentations.
Synchronization of a decoding system with a channel is achieved through the use of the SCR in the program stream and by its analog, the PCR, in the transport stream.
The SCR and PCR are time stamps encoding the timing of the bit stream itself, and are derived from the same time base used for the audio and video PTS values from the same program. Since each program may have its own time base, there are separate PCR fields for each program in a transport stream containing multiple programs. In some cases it may be possible for programs to share PCR fields. From Wikipedia, the free encyclopedia. Redirected from Presentation time stamp.
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The dark mode beta is finally here. Change your preferences any time. Stack Overflow for Teams is a private, secure spot for you and your coworkers to find and share information. I have a confusion about the timestamp of h RTP packet. The frame rate of my encoder is not exactly 30 FPS, it is variable. So, I cannot use any fixed timestamp. Could any one tell me the timestamp of the following encoded packet. So you determine that on the fly for each frame.
That way you can send 10 frames in one second 10fpsand in other second you can send 30 frames 30 fps. You only need to set the RTP timestamp correctly. And if I get your question, you are in doubt how to do this Let the starting time stamp be 0, you add the wall clock time in milliseconds multiplied by to the last RTP timestamp, or you can use any time scale you want.
To make the decoder decode 10fps video at 30fps, add to RTP timestamp for each packet And as you can see, the decoder uses RTP timestamps to know when to display each one, and it doesnt mind if the video was encoded at 30 or 10 fps. Also, if the video is 30 fps, that doesnt mean that for each second there will be 30 RTP packets. Sometimes there can be more thenso you can not have a formula that ensures the correct RTP timestamp calculation. The instant used for sampling the frame before encoding is called the PTS presentation timestamp.
It's out of the scope of the encoder, you must remember it in your data flow when you capture the frames. This also means the client will have to decode in the order received and not reorder the frames in the PTS order. I'm not using the term DTS here for a reason, because you don't need the decoding timestamp for this to work, only the order. In that case, you have to implement some application logic to reorder the units, refer to RFC for details.
Learn more. Asked 10 years, 1 month ago. Active 3 years, 3 months ago.H.265 (HEVC) vs H.264 (AVC) Compression: Explained!
Viewed 17k times.When I first made this tutorial, all of my syncing code was pulled from ffplay. Today, it is a totally different program, and improvements in the ffmpeg libraries and in ffplay. While this code still works, it doesn't look good, and there are many more improvements that this tutorial could use. How Video Syncs So this whole time, we've had an essentially useless movie player.
It plays the video, yeah, and it plays the audio, yeah, but it's not quite yet what we would call a movie. So what do we do? Fortunately, both the audio and video streams have the information about how fast and when you are supposed to play them inside of them. Audio streams have a sample rate, and the video streams have a frames per second value.
However, if we simply synced the video by just counting frames and multiplying by frame rate, there is a chance that it will go out of sync with the audio. Instead, packets from the stream might have what is called a decoding time stamp DTS and a presentation time stamp PTS. To understand these two values, you need to know about the way movies are stored. The two other kinds of frames are called "I" frames and "P" frames "I" for "intra" and "P" for "predicted".
I frames contain a full image. P frames depend upon previous I and P frames and are like diffs or deltas. B frames are the same as P frames, but depend upon information found in frames that are displayed both before and after them! So let's say we had a movie, and the frames were displayed like: I B B P.
Now, we need to know the information in P before we can display either B frame. Because of this, the frames might be stored like this: I P B B. This is why we have a decoding timestamp and a presentation timestamp on each frame. The decoding timestamp tells us when we need to decode something, and the presentation time stamp tells us when we need to display something.
But what we really want is the PTS of our newly decoded raw frame, so we know when to display it. Now, while it's all well and good to know when we're supposed to show a particular video frame, but how do we actually do so? Here's the idea: after we show a frame, we figure out when the next frame should be shown. Then we simply set a new timeout to refresh the video again after that amount of time.
As you might expect, we check the value of the PTS of the next frame against the system clock to see how long our timeout should be. This approach works, but there are two issues that need to be dealt with.
First is the issue of knowing when the next PTS will be. Now, you might think that we can just add the video rate to the current PTS — and you'd be mostly right.
However, some kinds of video call for frames to be repeated. This means that we're supposed to repeat the current frame a certain number of times. This could cause the program to display the next frame too soon. So we need to account for that.
The second issue is that as the program stands now, the video and the audio chugging away happily, not bothering to sync at all. We wouldn't have to worry about that if everything worked perfectly.
But your computer isn't perfect, and a lot of video files aren't, either. So we have three choices: sync the audio to the video, sync the video to the audio, or sync both to an external clock like your computer. For now, we're going to sync the video to the audio. Now let's get into the code to do all this.
The offset in an FLV file is always in milliseconds. Then look into the ISO ,8. This transport strategy ensures that both frames that the B-frame bridges are in the decoder before the B-frame is processed. I think I have understood the CTS. Because B-frames may depends fowarding frames to decode, so the CTS means when this B-frame can be decoded, usually that means all the depended frames are received.
Asked 8 years, 8 months ago. Active 5 years, 5 months ago. Viewed 10k times. The h. Thank you. Wang from Next Door Mr. Wang from Next Door 8, 6 6 gold badges 41 41 silver badges 64 64 bronze badges. It happens because b-frame requires frames following after it do be decoded. Wang from Next Door Aug 15 '11 at Active Oldest Votes. What about frame type?In order to get a grasp on what is happening on our machines, we first need to understand the underlying concepts.
Only then can we talk about implementation details. CPU load is expensive and codecs are complex. When video decoding with software on computers became popular, it was revolutionary. With the introduction of QuickTime 1. Up until this point, only specialized computers with certain graphics hardware were able to play color video.
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The solution was to add a dedicated decoding chip by C-Cube to the motherboard for all the heavy lifting. This chip can be found on the Wallstreet, Lombard, and Pismo PowerBook generations, as well as on their professional desktop equivalents.
In the mid 00s, a new kid arrived on the block and remains the dominant video codec on optical media, digital television broadcasting, and online distribution: H. However, this came at a cost: increased CPU load and the need for dedicated decoding hardware, especially on embedded devices like the iPhone.
Without the use of the provided DSPs, battery life suffers — typically by a factor of 2. It was highly optimized with handcrafted assembly code, and further improved upon in OS X InAdobe Flash Video switched from its legacy video codecs 2 to H. This changed in OS X It is a small framework wrapping the H. This is based on whether or not it can handle the input buffer of the encoded data that is provided. If the decoder can handle the input, it will return the decoded frames. There are four different error states available, none of them being particularly verbose.
It is a beast! It can compress and decompress video at real time or faster. Hardware accelerated video decoding and encoding can be enabled, disabled, or enforced at will. Furthermore, without actual testing, there is no way of knowing the supported codec, codec profile, or encoding bitrate set available on any given device.
Therefore, a lot of testing and device-specific code is needed. Internally, it appears to be similar to the modularity of the original QuickTime implementation. The external API is the same across different operating systems and platforms, while the actual available feature set varies. Based on tests conducted on OS X The A7 SoC added support for H. This feature is typically used in the broadcasting and content creation industries. This is expected to change in subsequent iOS releases, 4 but might be limited to future devices.
The only exception is for contents purchased from the iTunes Store which are protected by FairPlay2, the deployed Digital Rights Management. HLS typically consists of small chunks of H.Whatever clock is set by encoder try generating the pts of audio according to that.
Search everywhere only in this topic. Advanced Search. Classic List Threaded. Nisar Ahmed. Thanks for your answer, when I set timestamp of decoder, the frame rate jumps up to a very large value. Coming back to this issue, I have tested 2 scenarios. I am simply setting pts to the timestamp received from encoder and dts to 0 but the resulting file is showing very large frame rate value.
The framerate is correct this time but the quicktime player displays a white frame at the start of the movie. VLC plays fine but I also want Quicktime player to play the file normally as well. Which inputs matter in this kind of scenarios and how they relate to each other?
The output movie 's frame rate is 20fps when it should be Coming back to this issue, I have tested 2 scenarios a. Muxer does not change the pts or dts if you have provided by your own, may be you are unable to set pts, dts in your muxer.
Note: your a part wont work because your h elementary stream contain b frame, where dts is important part. Free forum by Nabble. Edit this page.