I need to measure quality of a live stream, and I was thinking of measuring PSNR. So I was wondering how can I run this test since it is a live stream, I do not have a refence stream or video. Should I look for a reference video of the same resolution and bitrate, like for example a Big buck bunny, and compare against It?
Thanks
Related
I am looking for a VMAF-like objective user-perception video quality scanner that functions at scale. The use case is a twitch-like streaming service where videos are eligible to be played on demand after the live stream completes. We want to have some level of quality in the on demand library without having to view every live stream. We are encoding the livestreams into HLS playlists after the stream completes, but using VMAF to compare the post-stream mp4 to the post-encoded mp4s in HLS doesn't provide the information needed as the original mp4 could be of low quality due to bandwidth issues during the live stream.
Clarification
Not sure if I get the question correctly. You want to measure the output quality of the transcoded video without using the reference video. Is that correct?
Answer
VMAF is a reference quality metric, which means it simply compares how much subjective distortion was introduced into the transcoded video when compared to the source video. It always needs a reference input video.
I think what you are looking for is a no-reference quality metric(s). Where you can measure the "quality" of video without a reference source video. There are a lot of no-reference quality metrics intended to capture different distortion artifacts in the output video. For example, blurring, blocking, and so on. Then you can make an aggregated metric based on these values depending upon what you want to measure.
Conclusion
So, if I were you, I would start searching for no-reference quality metrics. And then look for tools that can measure those no-reference quality metrics efficiently. Hope that answers your question.
My video stream is encoded with H.264, and audio stream is encoded with AAC. In fact, I get these streams by reading a file whose format is flv. I only decode video stream in order to get all video frames, then I do something by using ffmpeg before encoding them, such as change some pixels. At last I will push the video and audio stream to Crtmpserver. When I pull the live stream from this server, I find the video is not fluent but audio is normal. But when I change gop_size from 12 to 3, everything is OK. What reasons cause that problem, can anyone explain something to me?
Either the CPU, or the bandwidth is not sufficient for your usage. RTMP will always process audio before video. If ffmpeg, or the network is not able to keep up with the live stream, Video frames will be dropped. Because audio is so much smaller, and cheaper to encode, a very slow CPU or congested network will usually have no problems keeping up.
I have a bunch of .png frames and a .mp3 audio file which I would like to convert into a video. Unfortunately, the frames do not correspond to a constant frame rate. For instance, one frame may need to be displayed for 1 second, whereas another may need to be displayed for 3 seconds.
Is there any open-source software (something like ffmpeg) which would help me accomplish this? Any feedback would be greatly appreciated.
Many thanks!
This is not an elegant solution, but it will do the trick: duplicate frames as necessary so that you end up with some resulting (fairly high) constant framerate, 30 or 60 fps (or higher if you need higher time resolution). You simply change which frame is duplicated at the closest new frame to the exact timestamp you want. Frames which are exact duplicates will be encoded to a tiny size (a few bytes) with any decent codec, so this is fairly compact. Then just encode with ffmpeg as usual.
If you have a whole lot of these and need to do it the "right" way: you can indicate the timing either in the container (such as mp4, mkv, etc) or in the codec. For example in an H.264 stream you will have to insert SEI messages of type pic_timing to specify the timing of each frame. Alternately you will have to write your own muxer relying on a container library such as Matroska (mkv) or GPAC (mp4) to indicate the timing in the container. Note that not all codecs/containers support arbitrarily variable frame rate. Only a few codecs support timing in the codec. Also, if timing is specified in both container and codec, the container timing is used (but if you are muxing a stream into a container, the muxer should pick up the individual frame timestamps from the codec).
I'm looking for a way to continuously grab frames, as jpg, from a RTSP stream. I've stumbled upon ffmpeg but it seems that the time between starting it and grabbing the first frame is quite high. Is there any good tool in order to do this?
Regards
I've used gstreamer libraries in the past to extract frames from mobile video
I'm looking to generate web videos from movies taken with my digital camera. What formats should I generate, and at what resolution and bitrate to ensure playback on mobile and desktop devices?
Here's what I was thinking:
Input format: AVI, MOV
Output format: webm, ogv, mp4
Output resolutions: 1080p, 720p, 320p
Not really a programming question but I will answer it anyways:
WebM can be ditched completely. Very few devices support it. mp4 is the most common format that all devices support. Low end phones support 3gpp format instead [cousin of mp4]. If you have it you should be fine for 90% of the devices.
mp4 with h.264/aac is the most common and for devices that don't support those mpeg4 with mp3 will suffice.
How many devices do you have are 1080p resolution. Better to ditch the 1080p and get one SD resolution 480p in there.
Bitrates depend on the encoding profile and content. Just ensure do two pass encoding using ffmpeg and libx264 to get the best quality.
Most mobile devices can display "HD" content fairly well, these days. However, if you're looking to save on bandwidth on peoples data plans, a good resolution would probably be 852x480.
now, depending on if you need near lossless quality, or if you can accept minor artifacts in your video will determine your bitrate. for 1080p and x264, you can get near lossless with about 15mbps, but you could have watchable video with 10-11mbps. im not sure how well the other codecs compare, so you may have to try a couple test runs with a short video.
if you do 720p, you can most certainly get away with 4-6mbps.
with 852x480, you may be successful with as low as 1.5-2mbps.
480x320, or maybe even 320x240 may be a good option, if you suspect people will be watching this on lower end devices or on really slow connection, or very limited bandwidth. you could probably get away with 500kbps for 320x240, and 1mbps for 480x320.
these are all starting points, as each codec and selected encoding options will increase/decrease the quality. but i believe these to be good starting points.