I've been stuck on this problem for weeks now and Google is no help, so hopefully some here can help me.
I am programming a software sound mixer in C++, getting audio packets from the network and Windows microphones, mixing them together as PCM, and then sending them back out over the network and to speakers/USB headsets. This works. I have a working setup using the PortAudio library to handle the interface with Windows. However, my supervisors think the latency could be reduced between this software and our system, so in an attempt to lower latency (and better handle USB headset disconnects) I'm now rewriting the Windows interface layer to directly use WASAPI. I can eliminate some buffers and callbacks doing this, and theoretically use the super low latency interface if that's still not fast enough for the higher ups.
I have it only partially working now, and the partially part is what is killing me here. Our system has the speaker and headphones as three separate mono audio streams. The speaker is mono, and the headset is combined from two streams to be stereo. I'm outputting this to windows as two streams, one for a device of the user's choice for speaker, and one of another device of the user's choice for headset. For testing, they're both outputting to the default general stereo mix on my system.
I can hear the speaker perfectly fine, but I cannot hear the headset, no matter what I try. They both use the same code path, they both go through a WMF resampler to convert to 2 channel audio at the sample rate Windows wants. But I can hear the speaker, but never the headset stream.
It's not an exclusive mode problem: I'm using shared mode on all streams, and I've even specifically tried cutting down the streams to only the headset, in case one was stomping the other or something, and still the headset has no audio output.
It's not a mixer problem upstream, as I haven't modified any code from when it worked with PortAudio streams. I can see the audio passing through the mixer and to the output via my debug visualizers.
I can see the data going into the buffer I get from the system, when the system calls back to ask for audio. I should be hearing something, static even, but I'm getting nothing. (At one point, I bypassed the ring buffer entirely and put random numbers directly into the buffer in the callback and I still got no sound.)
What am I doing wrong here? It seems like Windows itself is the problem or something, but I don't have the expertise on Windows APIs to know what, and I'm apparently the most expert for this stuff in my company. I haven't even looked yet as to why the microphone input isn't working, and I've been stuck on this for weeks now. If anyone has any suggestions, it'd be much appreciated.
Check the re-sampled streams: output the stereo stream to the speaker, and output the mono stream to the handset.
Use IAudioClient::IsFormatSupported to check supported formats for the handset.
Verify your code using an mp3 file. Use two media players to play different files with different devices simultaneously.
Related
Does anyone know how to programmatically capture the sound that is being played (that is, everything that is coming from the sound card, not the input devices such as a microphone).
Assuming that you are talking about Windows, there are essentially three ways to do this.
The first is to open the audio device's main output as a recording source. This is only possible when the driver supports it, although most do these days. Common names for the virtual device are "What You Hear" or "Wave Out". You will need to use a suitable API (see WaveIn or DirectSound in MSDN) to do the capturing.
The second way is to write a filter driver that can intercept the audio stream before it reaches the physical device. Again, this technique will only work for devices that have a suitable driver topology and it's certainly not for the faint-hearted.
This means that neither of these options will be guaranteed to work on a PC with arbitrary hardware.
The last alternative is to use a virtual audio device, such as Virtual Audio Cable. If this device is set as the defualt playback device in Windows then all well-behaved apps will play through it. You can then record from the same device to capture the summed output. As long as you have control over the device that the application you want to record uses then this option will always work.
All of these techniques have their pros and cons - it's up to you to decide which would be the most suitable for your needs.
You can use the Waveform Audio Interface, there is an MSDN article on how to access it per PInvoke.
In order to capture the sound that is being played, you just need to open the playback device instead of the microphone. Open for input, of course, not for output ;-)
If you were using OSX, Audio Hijack Pro from Rogue Amoeba probably is the easiest way to go.
Anyway, why not just looping your audio back into your line in and recording that? This is a very simple solution. Just plug a cable in your audio output jack and your line in jack and start recordung.
You have to enable device stero mix. if you do this, direct sound find this device.
In my Android app, I use Android NDK to play music by doing the following:
extract audio samples from an OGG file using the Vorbis library
process the audio samples
redirect the processed samples to the audio output using the Oboe library
In order to avoid underruns, I do the first 2 steps in a separate thread, to extract and process the sound a little bit in advance (it adds a bit of latency, but this is not a problem for my app). That solution works great on every device I've tested so far.
But for some reason, when I pair my device with a bluetooth speaker and when I play music, I have what seems to be underruns on some devices like Samsung S7 or Nokia 1 (but not on every device).
This bug looks so random to me that I don't know where to start. It acts like the bluetooth connection is using quite a lot of CPU so my app doesn't have enough resource to run properly.
Does anyone have experienced something similar? Should I do anything in my code to handle the bluetooth connection so it doesn't use CPU (for example to avoid audio resampling)?
Thanks for your help.
Android + Bluetooth audio is a world of pain. The major thing to appreciate about Bluetooth is the audio sink runs at a rate independent of other audio devices, which is why the native mediaplayer will do things like display video in accordance with whatever rate the attached audio device consumes samples, essentially slaving itself to the clock of the BT audio device. If you want to drive the speed from Android (i.e. SystemClock timebase) you'll need to use a timestretching AudioTrack. (This can be done, but driver support is erratic and overall system stability tanks).
Firstly, you want to eliminate the devices themselves being problems. Can you play the ogg files in a media player to a Bluetooth speaker from the S7 or Nokia 1 without problems? If so, it's your code!
It sounds to me like the speaker is consuming samples faster than the device is producing them, for whatever reason. Basically check your callbacks to make sure whenever the audio subsystem requests more data you are actually providing it. Be sure to drive your decoding pipeline according to the callbacks being made and not the system clock or any other assumptions about timing.
Finally, Bluetooth audio, at least A2DP, as opposed to directly streaming MP3, is going to require some processing to recompress the audio as it is sent out, but those devices should have plenty of headroom for this, maybe even special DSPs. I've done it with 1080P video playback at the same time before, and it starts to fall apart with two videos at once!
I'm writing a movie player using VLC engine on Mac Os but I encounter a little problem :
On the VLC doc you can read about this (libvlc_media_player_set_time()) function :
Set the movie time (in ms).
This has no effect if no media is being played. Not all formats and protocols support this.
But I can't find any information about the formats and protocols supported. As my application is not very smooth, I was wondering if this might be the cause.
Nowadays, this works for virtually all local files (except for MPEG TS encapsulated streams in some corner cases). For networking, this depends on the availability of seeking - i.e. you cannot seek in live streams as they are coming in as it is happening. Same for multicast streams. However, it works just fine for VoD.
However, this should not be related to your app being smooth or not, so please elaborate if this answer doesn't solve your issue.
I'm writing a program similar to StreamMyGame with the difference of the client being free and more importantly, open source, so I can port it to other devices (in my case an OpenPandora), or even make an html5 or flash client.
Because the objective of the program is to stream video games, latency should be reduced to a minimum.
Right now I can capture video of Direct 3D 9 games at a fixed frame rate, encode it using libx264 and dumping it to disk, and send input remotely, but I'm stumped at sending the video and eventually the audio through the network.
I don't want to implement a way just to discover that it introduces several seconds of delay and I don't care how it is done as long as it is done.
Off of my head I can think several ways:
My current way, encode video with libx264 and audio with lame or as ac3 and send them with live555 as a RTSP feed, though the library is not playing nice with MSVC and I’m still trying to understand its functioning.
Have the ffmpeg library do all the grunt work, where it encodes and sends (I guess I'll have to use ffserver to get an idea on how to do it)
Same but using libvlc, perhaps hurting encoding configurability in the process.
Using several pipes with the independent programs (ie: piping data to x264.exe or ffmpeg.exe)
Use other libraries such as pjsip or JRTPLIB that might simplify the process.
The hard way, sending video and audio through an UDP channel and figuring out how to synchronizing everything at the client (though the reason to use RTSP is to avoid this).
Your way, if I didn't think of something.
The second option would really be the best as it would reduce the number of libraries (integrate swscale, libx264, the audio codec and the sender library), simplify the development and bringing more codec variety (CELT looks promising) but I worry about latency as it might have a longer pipeline.
100 ms would already be too much, especially when you consider you might be adding another 150 ms of latency when it is used trough broadband.
Does any of you have experience with these libraries, to recommend me to switch to ffmpeg, keep wrestling live555 or do anything else (even if I didn’t mentioned it)?
I had very good results of streaming large blocks of data with low latency using UDT4 library. But first I would suggest checking ffmpegs network capabilities, so you have a native solution in all operations.
I wanted to get some ideas one how some of you would approach this problem.
I've got a robot, that is running linux and uses a webcam (with a v4l2 driver) as one of its sensors. I've written a control panel with gtkmm. Both the server and client are written in C++. The server is the robot, client is the "control panel". The image analysis is happening on the robot, and I'd like to stream back the video from the camera to the control panel for two reasons:
A) for fun
B) to overlay image analysis results
So my question is, what are some good ways to stream video from the webcam to the control panel as well as giving priority to the robot code to process it? I'm not interested it writing my own video compression scheme and putting it through the existing networking port, a new network port (dedicated to video data) would be best I think. The second part of the problem is how do I display video in gtkmm? The video data arrives asynchronously and I don't have control over main() in gtkmm so I think that would be tricky.
I'm open to using things like vlc, gstreamer or any other general compression libraries I don't know about.
thanks!
EDIT:
The robot has a 1GHz processor, running a desktop like version of linux, but no X11.
Gstreamer solves nearly all of this for you, with very little effort, and also integrates nicely with the Glib event system. GStreamer includes V4L source plugins, gtk+ output widgets, various filters to resize / encode / decode the video, and best of all, network sink and sources to move the data between machines.
For prototype, you can use the 'gst-launch' tool to assemble video pipelines and test them, then it's fairly simply to create pipelines programatically in your code. Search for 'GStreamer network streaming' to see examples of people doing this with webcams and the like.
I'm not sure about the actual technologies used, but this can end up being a huge synchronization ***** if you want to avoid dropped frames. I was streaming a video to a file and network at the same time. What I eventually ended up doing was using a big circular buffer with three pointers: one write and two read. There were three control threads (and some additional encoding threads): one writing to the buffer which would pause if it reached a point in the buffer not read by both of the others, and two reader threads that would read from the buffer and write to the file/network (and pause if they got ahead of the producer). Since everything was written and read as frames, sync overhead could be kept to a minimum.
My producer was a transcoder (from another file source), but in your case, you may want the camera to produce whole frames in whatever format it normally does and only do the transcoding (with something like ffmpeg) for the server, while the robot processes the image.
Your problem is a bit more complex, though, since the robot needs real-time feedback so can't pause and wait for the streaming server to catch up. So you might want to get frames to the control system as fast as possible and buffer some up in a circular buffer separately for streaming to the "control panel". Certain codecs handle dropped frames better than others, so if the network gets behind you can start overwriting frames at the end of the buffer (taking care they're not being read).
When you say 'a new video port' and then start talking about vlc/gstreaming i'm finding it hard to work out what you want. Obviously these software packages will assist in streaming and compressing via a number of protocols but clearly you'll need a 'network port' not a 'video port' to send the stream.
If what you really mean is sending display output via wireless video/tv feed that's another matter, however you'll need advice from hardware experts rather than software experts on that.
Moving on. I've done plenty of streaming over MMS/UDP protocols and vlc handles it very well (as server and client). However it's designed for desktops and may not be as lightweight as you want. Something like gstreamer, mencoder or ffmpeg on the over hand is going to be better I think. What kind of CPU does the robot have? You'll need a bit of grunt if you're planning real-time compression.
On the client side I think you'll find a number of widgets to handle video in GTK. I would look into that before worrying about interface details.