I've been struggling with the following problem and can't figure out a solution. The provided java server application sends pcm audio data in chunks over a websocket connection. There are no headers etc. My task is to play these raw chunks of audio data in the browser without any delay. In the earlier version, I used audioContext.decodeAudioData because I was getting the full array with the 44 byte header at the beginning. Now there is no header so decodeAudioData cannot be used. I'll be very grateful for any suggestions and tips. Maybe I've to use some JS decoding library, any example or link will help me a lot.
Thanks.
1) Your requirement "play these raw chunks of audio data in the browser without any delay" is not possible. There is always some amount of time to send audio, receive it, and play it. Read about the term "latency." First you must get a realistic requirement. It might be 1 second or 50 milliseconds but you need to get something realistic.
2) Web sockets use tcp. TCP is designed for reliable communications, congestion control, etc. It is not design for fast low latency communications.
3) Give more information about your problem. Is you client and server communicating over the Internet or over a local Lan? This will hugely effect your performance and design.
4) The 44 byte header was a wav file header. It tells the type of data (sample rate, mono/stereo, bits per sample). You must know this information to be able to play the audio. IF you know the PCM type, you could insert it yourself and use your decoder as you did before. Otherwise, you need to construct an audio player manually.
Streaming audio over networks is not a trivial task.
Related
I would like to make an app (Target pc windows) that let you modify the micro input in real time, like introducing sound effects or even modulating your voice.
I searched over the internet and only found people telling that it would not be possible without using a virtual audio cable.
However I know some apps with similar behavior (voicemod, resonance) not using a virtual audio cable so I would like some help about how can be done (just the name of a library capable would be enough) or where to start.
Firstly, you can use professional ready-made software for that - Digital audio workstation (DAW) in combination with a huge number of plugins for that.
See 5 steps to real-time process your instrument in the DAW.
And What is (audio) direct monitoring?
If you are sure you have to write your own, you can use libraries for real-time audio processing (as far as I know, C++ is better for this than C#).
These libraries really works. They are specially designed for realtime.
https://github.com/thestk/rtaudio
http://www.portaudio.com/
See also https://en.wikipedia.org/wiki/Csound
If you don't have a professional sound interface yet, but want to minimize a latency, read about Asio4All
The linked tutorial worked for me. In it, a sound is recorded and saved to a .wav.
The key to having this stream to a speaker would be opening a SourceDataLine and outputting to that instead of writing to a wav file. So, instead of outputting on line 59 to AudioSystem.write, output to a SourceDataLine write method.
IDK if there will be a feedback issue. Probably good to output to headphones and not your speakers!
To add an effect, the AudioInputLine has to be accessed and processed in segments. In each segment the following needs to happen:
obtain the byte array from the AudioInputLine
convert the audio bytes to PCM
apply your audio effect to the PCM (if the effect is a volume change over time, this could be done by progressively altering a volume factor between 0 to 1, multiplying the factor against the PCM)
convert back to audio bytes
write to the SourceDataLine
All these steps have been covered in StackOverflow posts.
The link tutorial does some simplification in how file locations, threads, and the stopping and starting are handled. But most importantly, it shows a working, live audio line from the microphone.
We're looking for a way to send per frame metadata (for example an ID) with H264 encoded frames from a server to a client.
We're currently developing a remote rendering application, where both client and server side are actively involved.
The server renders a high quality image with all effects, lighting etc.
The client also has model-informations and renders a diffuse image that is used when the bandwidth is too low or the images have to be warped in order to avoid stuttering .
So far we're encoding the frames on the server side with ffmpeg and streaming them with live555 to the client, who receives an rtsp-stream and decodes the frames again using ffmpeg.
For our application, we now need to send per frame metadata.
We want the client to tell the server where the camera is right now.
Ideally we'd be able to send the client's view matrix to the server, render the corresponding frame and send it back to the client together with its view matrix. So when the client receives a frame, we need to know exactly at what camera position the frame was rendered.
Alternatively we could also tag each view matrix with an ID, send it to the server, render the frame and tag it with the same ID and send it back. In this case we'd have to assign the right matrix to the frame again on the client side.
After several attempts to realize the above intent with ffmpeg we came to the conclusion that ffmpeg does not provide the required functionality. ffmpeg only provides a fix, predefined set of fields for metadata, that either cannot store a matrix or can only be set for every key frame, which is not frequently enough for our purpose.
Now we're considering using live555. So far we have an on demand Server, witch gets a VideoSubsession with a H264VideoStreamDiscreteFramer to contain our own FramedSource class. In this class we load the encoded AVPacket (from ffmpeg) and send its data-buffer over the network. Now we need a way to send some kind of metadata with every frame to the client.
Do you have any ideas how to solve this metadata problem with live555 oder another library?
Thanks for your help!
It seems this question was answered in the comments:
pipe the output of ffmpeg through a custom tool that embedded the data
in the 264 elementary stream via an SEI
Someone also gave the following answer, which was deleted a few years ago for dubious reasons (it is brief but does seem to contain sufficient information):
You can do so using MPEG-4. See details for MPEG-4 Part 14 for
details.
I'm searching for a way to analyse the content of internet radios. I want to write a ruby client that can get the current track, next track, band, bpm and other meta information from a stream (e.g. a radio on shoutcast).
Does anybody know how to do this? And how do I record that stream into a mp3 or aac file?
Maybe there is a library that can already do this, I haven't one so far.
regards
I'll answer both of your questions.
Metadata
What you are seeking isn't entirely possible. Information on the next track is not available (keep in mind not all stations are just playing songs from a playlist... many offer live content). Advanced metadata such as BPM is not available. All you get is something like this:
Some Band - Some Song
The format of {artist} - {song title} isn't always followed either.
With those caveats, you can get that metadata from a stream by connecting to the stream URL and requesting the metadata with the following request header:
Icy-MetaData: 1
That tells the server to send the metadata, which is interleaved into the stream. Every 8KB or so (specified by the server in a response header), you'll find a chunk of metadata to parse. I have written up a detailed answer on how to parse that here: Pulling Track Info From an Audio Stream Using PHP The prior question was language-specific, but you will find that my answer can be easily implemented in any language.
Saving Streams to Disk
Audio playing software is generally very resilient to errors. SHOUTcast servers are built on this principal, and are not knowledgeable about the data going through them. They just receive data from an encoder, and when the client requests the stream, they start sending that data at an arbitrary point.
You can use this to your advantage when saving stream data. It is possible to simply write the stream data as it comes in to a file. Most audio players will play them without problem. I have tested this with MP3 and AAC.
If you want a more conformant file, you will have to use a library or parse the stream yourself to split on the appropriate frames, and then handle bit reservoir issues in your code. This is a lot of work, and generally isn't worth doing unless you find your files have real compatibility problems.
I'm writing a DirectShow source filter which is registered as a CLSID_VideoInputDeviceCategory, so it can be seen as a Video Capture Device (from Skype, for example, it is viewed as another WebCam).
My source filter is based on the VCam example from here, and, for now, the filter produces the exact output as this example (random colored pixels with one Video output pin, no audio yet), all implemented in the FillBuffer() method of the one and only output pin.
Now the real scenario will be a bit more tricky - The filter uses a file handle to a hardware device, opened using the CreateFile() API call (opening the device is out of my control, and is done by a 3Party library). It should then read chunks of data from this handle (usually 256-512 bytes chunk sizes).
The device is a WinUSB device and the 3Party framework just "gives" me an opened file handle to read chunks from.
The data read by the filter is a *.mp4 file, which is streamed from the device to the "handle".
This scenario is equivalent to a source filter reading from a *.mp4 file on the disk (in "chunks") and pushing its data to the DirectShow graph, but without the ability to read the file entirely from start to end, so the file size is unknown (Correct?).
I'm pretty new to DirectShow and I feel as though I'm missing some basic concepts. I'll be happy if anyone can direct me to solutions\resources\explanations for the following questions:
1) From various sources on the web and Microsoft SDK (v7.1) samples, I understood that for an application (such as Skype) to build a correct & valid DirectShow graph (so it will render the Video & Audio successfully), the source filter pin (inherits from CSourceStream) should implement the method "GetMediaType". Depending on the returned value from this implemented function, an application will be able to build the correct graph to render the data, thus, build the correct order of filters. If this is correct - How would I implement it in my case so that the graph will be built to render *.mp4 input in chunks (we can assume constant chunk sizes)?
2) I've noticed the the FillBuffer() method is supposed to call SetTime() for the IMediaSample object it gets (and fills). I'm reading raw *.mp4 data from the device. Will I have to parse the data and extract the frames & time values from the stream? If yes - an example would b great.
3) Will I have to split the data received from the file handle (the "chunks") to Video & Audio, or can the data be pushed to the graph without the need to manipulate it in the source filter? If split is needed - How can it be done (the data is not continuous, and is spitted to chunks) and will this affect the desired implementation of "GetMediaType"?
Please feel free to correct me if I'm using incorrect terminology.
Thanks :-)
This is a good question. On the one hand this is doable, but there is some specific involved.
First of all, your filter registered under CLSID_VideoInputDeviceCategory category is expected to behave as a live video source. By doing so you make it discoverable by applications (such as Skype as you mentioned), and those applications will be attempting to configure video resolution, they expect video to go at real time rate, some applications (such as Skype) are not expecting compressed video such H.264 there or would just reject such device. You can neither attach audio right to this filter as applications would not even look for audio there (not sure if you have audio on your filter, but you mentioned .MP4 file so audio might be there).
On your questions:
1 - You would have a better picture of application requirement by checking what interface methods applications call on your filter. Most of the methods are implemented by BaseClasses and convert the calls into internal methods such as GetMediaType. Yes you need to implement it, and by doing so you will - among other - enable your filter to connect with downstream filter pins by trying specific media types you support.
Again, those cannot me MP4 chunks, even if such approach can work in other DirectShow graphs. Implementing a video capture device you should be delivering exactly video frames, preferably decompressed (well those could be compressed too, but you are going to immediately have compatibility issies with applications).
A solution you might be thinking of is to embed a fully featured graph internally to which you inject your MP4 chunks, then the pipelines parse those, decodes and delivers to your custom renderer, taking frames on which you re-expose them off your virtual device. This might be a good design, though assumes certain understanding of how filters work internally.
2 - Your device is typically treated as/expected to be a live source, which means that you deliver video in realtime and frames are not necessarily time stamped. So you can put times there and yes you definitely need to extract time stamps from your original media (or have it done by internal graph as mentioned in item 1 above), however be prepared that applications strip time stamps especially for preview purposes, since the source is "live".
3 - Getting back to audio, you cannot implement audio on the same virtual device. Well you can, and this filter might be even working in a custom built graph, but this is not going to work with applications. They will be looking for separate audio device, and if you implement such, they will instantiate it separately. So you are expected to implement both virtual video and virtual audio source, and implement internal synchronization behind the scenes. This is where timestamps will be important, by providing them correctly you will keep lip sync in live session to what it was originally on the media file you are streaming from.
What i want to do is the following procedure:
Get a frame from the Webcam.
Encode it with an H264 encoder.
Create a packet with that frame with my own "protocol" to send it via UDP.
Receive it and decode it...
It would be a live streaming.
Well i just need help with the Second step.
Im retrieving camera images with AForge Framework.
I dont want to write frames to files and then decode them, that would be very slow i guess.
I would like to handle encoded frames in memory and then create the packets to be sent.
I need to use an open source encoder. Already tryed with x264 following this example
How does one encode a series of images into H264 using the x264 C API?
but seems it only works on Linux, or at least thats what i thought after i saw like 50 errors when trying to compile the example with visual c++ 2010.
I have to make clear that i already did a lot of research (1 week reading) before writing this but couldnt find a (simple) way to do it.
I know there is the RTMP protocol, but the video stream will always be seen by one peroson at a(/the?) time and RTMP is more oriented to stream to many people. Also i already streamed with an adobe flash application i made but was too laggy ¬¬.
Also would like you to give me an advice about if its ok to send frames one by one or if it would be better to send more of them within each packet.
I hope that at least someone could point me on(/at?) the right direction.
My english is not good maybe blah blah apologies. :P
PS: doesnt has to be in .NET, it can be in any language as long as it works on Windows.
Many many many many thanks in advance.
You could try your approach using Microsoft's DirectShow technology. There is an opensource x264 wrapper available for download at Monogram.
If you download the filter, you need to register it with the OS using regsvr32. I would suggest doing some quick testing to find out if this approach is feasible, use the GraphEdit tool to connect your webcam to the encoder and have a look at the configuration options.
Also would like you to give me an advice about if its ok to send frames one by one or if it would be better to send more of them within each packet.
This really depends on the required latency: the more frames you package, the less header overhead, but the more latency since you have to wait for multiple frames to be encoded before you can send them. For live streaming the latency should be kept to a minimum and the typical protocols used are RTP/UDP. This implies that your maximum packet size is limited to the MTU of the network often requiring IDR frames to be fragmented and sent in multiple packets.
My advice would be to not worry about sending more frames in one packet until/unless you have a reason to. This is more often necessary with audio streaming since the header size (e.g. IP + UDP + RTP) is considered big in relation to the audio payload.