I have a two dump files of raw video and raw audio from an encoder and I want to be able to measure the "Lip-sync". Imagine a video of a hammer striking an anvil. I want to go frame by frame and see that when the hammer finally hits the anvil, there is a spike in amplitude on the audio track.
Because of the speed that everything happens at, I cannot merely listen to the audio, i need to see the waveform in time domain.
Are there any tools out there that will let me see both the video and audio?
If you are concerned about validating a decoder then generally from a validation perspective the goal is to check Audio and Video PTS values against a common real time clock.
Raw YUV and PCM files do not include timestamps. If you know the frame-rate and sample-rate you can use a raw yuv file viewer (I wrote my own) to figure out the time (from start of file) of a given frame in the video, and a tool like Audacity to figure out the time form start of file to a start of tone in the audio file. this still may not tell you the whole story since tools usually embed a delay between the audio and video in the ts/ps file. Or you can hook up ab OScope and go old school.
Related
Has anyone tried to modify the CISCO openh264 library to take JPEG images as input and compress them into P and I frames (output as frames, NOT video) and similarly to modify decoder to take compressed P and I frames and generate uncompressed-frames ?
I have a camera looking at a static scene and taking pictures (1280x720p) every 30 second. The scene is almost static. Currenlty I am using JPEG compression to compress each frame individually and it is resulting in an image size of ~270KB. This compressed frame is transferred via internet to a storage server. Since there is very little motion in the scene, the 'I' frame size will be very small (I think it should be ~20-50KB). So it will be very cost effective to transmit I frames over internet instead of JPEG images.
Can anyone guide me to some project or about how to proceed with this task ?
You are describing exactly what a codec does. It takes images, and compresses them. There relationship in time is irrelevant to the compression step. The decoder than decides how to display or just write them to disk. You don't need to modify open264, what you want to do is exactly what it is designed to do.
I have a real time video stream, and want to cut some video clips from it by accurate timestamp(pts).
When I receiver an avpacket, I decode it, and do something and cache the avpacket. I don't want to re-encode all avpackets, it cost cpu resource.
There are many gop structure in H.264 stream, usually we should cut the video begin at the key frame, and end at the key frame. Otherwise the front some frames in the video clip would display error.
Now I use av_write_frame to make avpacket to video. But sometimes the length of gop is very long, such as it could be 250, 8.3s(30 frame per second). It means the distance between two I-frame could be 250 frames. The video clip is short, I don't want to add too many unused frames.
How should I do? I think i should insert a i-frame at the start position of video clip. Could I change a p-frame to i-frame?
Thanks your reading!
This is not possible in the generic case, but may be in specific cases. Even then, there are no open source/free tools to do this, and I am unaware of any commercial tools. The reason I say it is not possible in the generic case is each frame can reference up to 16 other frames. So you can not just replace a single frame, You will need to replace all referenced frames. Doing this will likely take almost as much CPU as encoding the whole GOP.
I've already loaded the .wav audio to the buffer with XAudio2 (Windows 8.1) and to play it I just have to use:
//start consuming audio in the source voice
/* IXAudio2SourceVoice* */ g_source->Start();
//play the sound
g_source->SubmitSourceBuffer(buffer.xaBuffer());
I wonder, how can I get the frequency value at given time with XAudio2?
The question does not make much sense, a .wav file contains a great many frequencies. It is the blend of them that makes it sound like music to your ears, instead of just an artificial generated tone. A blend that's constantly changing.
A signal processing step is required to convert the samples in the .wav file from the time domain to the frequency domain. Generally known as spectrum analysis, the Fast Fourier Transform (FFT) is the standard technique.
A random Google hit on "xaudio2 fft" produced this code sample. No idea how good it is, but something to play with to get the lay of the land. You'll find more about it in this gamedev question.
I am looking for a fast way to load in a video file and to create images from them at certain intervals ( every second, every minute, every hour, etc.).
I tried using DirectShow, but it just ran too slow for me to start the video file and move to a certain location to get data and to save it out to an image. Even if I disabled the reference clock. Tried OpenCV, but it has trouble opening the AVI file unless I know the exact codec information. So if I know a way to get the codec information out from OpenCV I may give it another shot. I tried to use FFMPEG, but I don't have as much control over it as well as I would wish.
Any advice would be greatly appreciated. This is being developed on a Windows box since it has to be hosted on a Windows box.
MPEG-4 format is not an intra-coded format, so you can't just jump to a random frame and decode it on its own, as most frames only encode the differences from one or more other frames. I suspect your decoding is slow because when you land on a frame for which several other dependent frames to be decoded first.
One way to improve performance would be to determine which frames are keyframes (or sometimes also called 'sync' points) and limit your decoding to those frames, since these can be decoded on their own.
I'm not very familiar with DirectShow capabilities, but I would expect it has some API to expose sync points.
Also, I should mention that the QuickTime SDK on Windows is possibly another good option that you have for decoding frames from movies. You should first test that your AVI movies are played correctly in the QuickTime Player. And the QT SDK does expose sync points, see the section Finding Interesting Times in the QT SDK documentation.
ffmpeg's libavformat might work for ya...
I build a DirectShow graph consisting of my video capture filter
(grabbing the screen), default audio input filter both connected
through spliiter to WM Asf Writter output filter and to VMR9 renderer.
This means I want to have realtime audio/video encoding to disk
together with preview. The problem is that no matter what WM profile I
choose (even very low resolution profile) the output video file is
always "jitter" - every few frames there is a delay. The audio is ok -
there is no jitter in audio. The CPU usage is low < 10% so I believe
this is not a problem of lack of CPU resources. I think I'm time-
stamping my frames correctly.
What could be the reason?
Below is a link to recorder video explaining the problem:
http://www.youtube.com/watch?v=b71iK-wG0zU
Thanks
Dominik Tomczak
I have had this problem in the past. Your problem is the volume of data being written to disk. Writing to a faster drive is a great and simple solution to this problem. The other thing I've done is placing a video compressor into the graph. You need to make sure both input streams are using the same reference clock. I have had a lot of problems using this compressor scheme and keeping a good preview. My preview's frame rate dies even if i use an infinite Tee rather than a Smart Tee, the result written to disk was fine though. Its also worth noting that the more of a beast the machine i was running it on was the less of an issue so it may not actually provide much of a win if you need both over sticking a new faster hard disk in the machine.
I don't think this is an issue. The volume of data written is less than 1MB/s (average compression ratio during encoding). I found the reason - when I build the graph without audio input (WM ASF writer has only video input pint) and my video capture pin is connected through Smart Tree to preview pin and to WM ASF writer input video pin then there is no glitch in the output movie. I reckon this is the problem with audio to video synchronization in my graph. The same happens when I build the graph in GraphEdit. Without audio, no glitch. With audio, there is a constant glitch every 1s. I wonder whether I time stamp my frames wrongly bu I think I'm doing it correctly. How is the general solution for audio to video synchronization in DirectShow graphs?