I'm writing a program to windows store in c++ which plays back the microphone. I have to modify the bits before sending that to the speakers. Firstly I wanted to play back the microphone without any effect bit it is lagging. The frequency and the bit rate is the same (24 bit, 192000Hz) but I also tried with (24 bit, 96000Hz). I debugged it and it seems that the speaker is faster therefore it has to wait for the data from the microphone like the squeakers would work in a higher frequency but according to the settings it doesn't. Dose anyone have a sightliest idea what is the problem here?
When you say that there are some 'lag', do you mean that there are some delay between when you feed the audio capture device with data and when the playback device renders the data or do you mean that the audio stream is 'chopped' with small pauses in between each sample being rendered?
If there's delay in playback I would take a look at with what latency value you've initialized the audio capture client.
If there are small pauses then I would recommend you using double buffering of sample data so that one buffer is being rendered while the other is being re-fetched from the audio capture device.
Related
I'm building an acoustic cancelling device based on Pyaudio, fourier transforms and. c-Media usb audio card. The software is threaded, using the producer/consumer model.
The device detects pure tones in the environment (reads chunks of microphone audio), uses fourier to detect the pure tone, and so far so good it works like a charm.
The final step however is getting tricky. I'm aiming to generate a 100ms wave (sine wave), which holds a certain amounts of periods of the frequency to be cancelled.
This wave buffer has to be played with Pyaudio on a separate thread continuously, which also must increase the phase little by little till detecting the amplitude of the tone in the environment drops. This is basically destructive interference.
My problem is when using Pyaudio.stream.write(), the buffer keeps overruning, since i have NO IDEA, what the function is doing internally. I have tried with many combinations of "frame_buffer_size" and audio lenght and no matter what i do, the buffer is overrun.
Ideally, the buffer does not have to be recalculated with a different phase on each run... instead, i'm trying pyaudio to read a different part of the buffer (window) to start writing the sine wave on a different origin each time.
I have no idea how to do that.
Long story short, how would you:
1) create a thread to fill a circular buffer continuously with audio data.
2) create a pyaudio consumer thread that continuously reads the buffer without overruning.
3) manipulate the volume on realtime
My output data must be 44100 hz, little endian, 16bit signed int. Any hints, advise, references or suggestions will be greatly appreciated.
Is it possible to play a simple and short video on smart eye glasses?
I know that it can play audio and it can show images one after the other. It should not be too much work from there I am just guessing.
There is no direct support for video playback, but as Ahmet says, you can approach this with showing Bitmaps as fast as possible.
The playback speed depends on the connection - so it is recommended to use High performance mode - wifi connection to achieve highest frame rate (setPowerMode)
Also take a look at showBitmapWithCallback which provides you a callback right after previous frame gets rendered, so you can show another one.
Yes, it is possible. You can grab frames of the video and display them one after another, as bitmaps.
That should give you a video playback view on the SmartEyeglass.
I'm trying to capture an AVI video, using DirectShow AVIMux and FileWriter Filters.
When I connect SampleGrabber filter instead of the AVIMux, I can clearly see that the stream is 30 fps, however upon capturing the video, each frame is duplicated 4 time and I get a 120 frames instead of 30. The movie is 4 times slower than it should be and only the first frame in the set of 4 is a Key Frame.
I tried the same experiment with 8 fps and for each image I received, I had 15 frames in the video. And in case of 15 fps, I got each frame 8 times.
I tried both writing the code in C++ and testing it with Graph Edit Plus.
Is there any way I can control it? Maybe some restrictions on the AVIMux filter?
You don't specify your capture format which could have some bearing on the problem, but generally it sounds like the graph when writing to file has some bottleneck which prevents the stream from continuing to flow at 30fps. The camera is attempting to produce frames at 30fps, and it will do so as long as buffers are recycled for it to fill.
But here the buffers aren't available because the file writer is busy getting them onto the disk. The capture filter is starved and in this situation it increments the "dropped frame" counter which travels with each captured frame. AVIMux uses this count to insert an indicator into the AVI file which says in effect "a frame should have been available here to write to file, but isn't; at playback time repeat the last frame". So the file should have placeholders for 30 frames per second - some filled with actual frames, and some "dropped frames".
Also, you don't mention whether you're muxing in audio, which would be acting as a reference clock for the graph to maintain audio-video sync. When capture completes if also using an audio stream, AVIMux alters the framerate of the video stream to make the duration of the two streams equal. You can check whether AVIMux has altered the framerate of the video stream by dumping the AVI file header (or maybe right click on the file in explorer and look at the properties).
If I had to hazard a guess as to the root of the problem, I'd wager the capture driver has a bug in calculating the dropped frame count which is in turn messing up AVIMux. Does this happen with a different camera?
I'm writing a video player. For audio part i'm using XAudio2. For this i have separate thread that is waiting for BufferEnd event and after this fills buffer with new data and call SubmitSourceBuffer.
The problem is that XAudio2(driver or sound card) has huge delays before playing next buffer if buffer size is small (1024 bytes). I made measurements and XAudio takes up to two times long for play such chunk. (1024 bytes chunk of 48khz raw 2-channeled pcm should be played in nearly 5ms, but on my computer it's played up to 10ms). And nearly no delays if i make buffer 4kbytes or more.
I need such small buffer to be able making synchronizations with video clock or external clock (like ffplay does). If i make my buffer too big then end-user will hear lot of noises in output due to synchronization stuff.
Also i have made measurements on all my functions that are decoding and synchronizing audio or anything else that could block or produce delays, they take 0 or 1 ms to execute, so they are not the problem 100%.
Does anybody know what can it be and why it's happenning? Can anyone check if he has same delay problems with small buffer?
I've not experienced any delay or pause using .wav files. If you are using mp3 format, it may add silence at the beginning and end of the sound during the compress operation thus causing a delay in your sound playing. See this post for more information.
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?