I am developing an app that records audio from microphone to integer array. Array is then passed to FFT and MFCC. I need to make frames about n samples and I need to 50% overlaps them (they cannot be side by side). So I need 3 buffers - when the first is full is passed to FFT. I this moment app should record second half of second buffer and first half of third buffer. FFT will be separate thread (my idea).
So I tried to sample audio using QAudioRecorder and QAudioProbe. I connected audioBufferProbed signal with processBuffer and there i use buffer.constData<int>(). I seems it works. I understood audioBufferProbed is emitted when buffer is full.
I don't know how to associate more buffers with one recorder. Or start writing to second buffer too at half of first buffer.
The easiest way here (since 50% is a nice round number) is to ask Qt for frames half the size of the ones you need for your FFT. You now run your FFT over frames 0&1, 1&2, 2&3, etc.
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I am working on the development of driving software for the hardware implementation by these people. The decoder works properly in overall, but I am struggling making it starting playing the sound at the middle. I suspect that it is common feature of the MP3 decoders as they must have some history of data in order to properly construct current sound (I am not that skilled in MPEG, however have an idea of some basics).
The problem is that this decoder is a black box, and any deepening in its code is an enormous time and effort.
I empirically found out that the sound garbage, when starting somewhere in the middle, happens in no more that 1 (one) seconds after start with file # 320 kbps and 44100 sampling rate. I am actually ok to mute decoder for a second (while it gathers/decodes proper required data for further playback), and then unmute it to continue playback.
I did search on the internet for the matter, did not find anything useful. Tried to invalidate first frames by corrupting frame headers (the easiest that could be done without going into the MP3 headers/data), made things even worse.
Questions:
is there any body of knowledge of how players perform seek in MP3 files and keep non-corrupt sound?
Is my action plan seem valid - mute for 1 second while decoder plays garbage? Is there any way to (easily) calculate the time I must mute output for?
Update: just tried on another file # 128 kbps/48k and maximal garbage time to be about 2 seconds... I can not believe that decoder with so limited resources - input buffer used is 2 kB with some intermediate working buffers, in total must be not more than 36 kB - can keep the history for 2 seconds, or decoder is having problems finding the sync word in the stream... and thus my driver needs to figure out the frame start (by finding out sync word, reading frame header, calculating frame size, and looking after the frame to contain another sync word).
I've found workarounds. The difficulty was that there are actually two problems overlaying each other, but was easy to cope with having structured approach.
The decoder is having issues getting the first sync word of the stream, and works very well when the first bytes supplied to it are FF FB or FF FA. All other bytes - in the middle of the frame - with very high probability, cause major sound corruption, until decoder catches correct sync. Thus I designed the code seeking to the next frame start after the seek point, checking that this is actual start of the frame by calculating frame size and looking at the next frame to contain FFFB/FA.
Having fixed the problem 1 I have had minor corruption left from the decoder starting decoding the frame without historical data. I have solved it by muting the decoder for the first 4 buffering transactions.
Major corruption still happens, but is rare, and it seems that nature of corruption depends on what was in the decoder buffers (not only Huffman input buffer, but other intermediate buffers) before the decoder is instructed to start. My hardware performs clear of the input buffers to 0 when decoder is in reset state, but it seems to be not enough (or just incorrect)...
The decoder itself is a kind of PoC (proof of concept) work, a student term with the aim to prove that they were able to make it; the package is having test bench code, but lacks low level documentation/comments in the code, and is not ready for field implementation and production. In general the fact that it worked for me at all (almost) out of the box makes the honor to the developers and is a mark of high quality of their work. I have reviewed and tried several published projects for MP3 decoders for silicon implementation (FPGA) and concluded that this one is the best available. In addition, the license they provide their work on is generous one.
Update: my research have shown that the most problem lies not in the input buffer (however it is possible to improve the situation by uploading 528 bytes of historical data to the decoder's buffer so that it would be able to grab main data from previous frame), but in the internal state of the decoder. Its documentation says:
To reduce resource usage, part of the RAM for buffering the intermediate data is also shared with Huffman decoding as bit reservoir ...
thus it is a contents of the reservoir and intermediate computed data affecting the decoding. I have confirmed it by starting various set of frames in different sequence, and if set of frames are played in different sequence, nature of garbage changes, or garbage may simply not appear.
Thus, unfortunately, my conclusion: it is not possible to properly seek using this decoder as is. I even do not think it is possible to "fake" playback (to quickly "play" the file till the needed point in buffers) as all three clocks are tied to each other.
I will keep my "best tested" implementation, with the notes on the quality.
Update 2: I was wrong, it is possible to seek softly, but to mitigate the sound corruption (yes, I am still unsure if I fixed it completely) I had to find another deficiency in the decoder: it is related to timing, decoder assumes that further data is always available in the buffer, while it may not be there yet. (It is actually clear from the test bench code supplied within the IP - the way data was replenished during QA and testing). In the cases I caught the corruption, first frames in the first part of the input buffer RAM were not decoded properly, skipped, and decoder quickly skips to second part of the RAM, assuming new data is there, however driving hardware is not ready yet fetching required data and putting this data into the second part of decoder's buffer RAM, thus corruption persisted for quite a long time with decoder looping skipping "invalid" frames until it catches correct image of the frame and normalizes its pace through the buffer.
Now the solution:
play (almost) 5 frames of silence through decoder before unmuting it. This will ensure all decoder's internal buffers are purged. It will not take much time, however requires some coding;
introduce a possibility to set huffman's decoder starting pointer readptr (in huffctl.v) after reset into the value other than 0. It will give the flexibility to have some history data uploaded into the decoder's buffer and start huffman decoder from the middle of the buffer rather than from its very start;
calculate the position to seek to, it calculates relatively easily for MPEG-1 Layer-3: duration=(filesize-ID3size)/(bitrate/8*1000), newPosition=ID3size+seekTime*(bitrate/8*1000). Duration is needed to check that position to seek to fits into the play time, alternatively newPosition can be used to check against file size. These forumlas do not take into account older tag versions appearing at the end of the file, but they are usually not more than 128 bytes, thus a kind of negligible for timing calculation relative to average MP3 sound file size; it also assumes CBR (VBR will require completely different way, requiring more power and data I/O for accurate seeking). Funny enough I found web pages with incorrect duration calculation formula, thus beware posts by ignorant people with cool job titles;
Seek to the calculated position, find next frame from this position on, calculate frame size, and ensure that there's next valid frame at that distance. New pointer will point to this next frame found at the distance;
find out the main_data_begin lookback pointer of the frame now being pointed to at step 4. Decrease the new pointer by this value so that pointer points within previous frame to the start of the main data for the current frame - it will be a pointer for the decoder data start. Note that it will fail if main data begins in more than one frame back (removal of headers of previous frame(s) will be required for proper operation);
fill decoder's buffer starting pointer identified in step 5, and set decoder's decoding start pointer to the one identified in step 4. While the implementation assumes you fill buffer in halves, do it different from the start: fill the whole buffer instead of just a first half. For this, after reset, set idle bit, check for data request, reset idle bit, perform two 1024 byte transfers to the decoder's buffer (effectively filling it completely), and then set idle bit, then reset it, and then set it again;
after performing step 7 continue normally replenishing 1024 bytes per decoder's request.
Employing this plan I had zero sound corruption cases. As you see it requires some changes to Verilog, but it must be easy if you know basics or hardware, know Verilog amd can perform reverse engineering.
I recently picked up a project where I need to perform a real-time sliding FFT analysis on incoming microphone data. The environment I picked to do this in, is OpenGL and Cinder and using C++.
This is my first experience in audio programming and I am a little bit confused.
This is what I am trying to achieve in my OpenGL application:
So in every frame, there's a part of the incoming data. In a for-loop (therefore multiple passes) a window of the present data will be consumed and FFT analysis will be performed on it. For next iteration of the for-loop, window will advance "hop-size" through the data and etc. until the end of the data is reached.
Now this process must be contiguous. But as you can see in the figure above, as soon as my current app frame ends and when next frame's data comes in, I can't pick up where I left the previous frame (because data is already gone). You can see it in figure where the blue area is in-between two frames.
Now you may say, pick the window-size / hop-size in a way that this never happens but that is impossible since these parameters should left user-configurable in my project.
Suggestions for this kind of processing, oriented towards C++11 is also very welcomed!
Thanks!
Not sure I understand your scenario 100%, but sounds like you may want to use a circular buffer. There is no "standard" circular buffer, but there's one in Boost.
However, you'd need a lock if you plan to do the processing with 2 threads. One thread, for example, would wait on the audio input, then take the buffer lock, and copy from the audio buffer to the circular buffer. The second thread would periodically take the buffer lock and read the next k elements, if there are at least k available in the buffer...
You'd need to adjust the size of the buffer appropriately and make sure you always handle the data faster than the incoming rate to avoid losses in the circular buffer...
Not sure why you mention that the buffer is lock-free and whether that is a requirement, I'd try the circular buffer with locks first as it seems simpler conceptually, and only go lock-free if you have to, because the data structure could be more complicated in this case (but maybe a "producer-consumer" lock-free queue would work)...
HTH.
Thanks for posting a graphic--that illustrates the problem nicely.
All you really need here is a buffer of size (window - 1) where you can store zero or more samples from the "previous" frame for processing in the "next" one. In C++ this would be:
std::vector<Sample> interframeBuffer;
interframeBuffer.reserve(windowSize - 1);
Then when you are within windowSize samples from the end of the current frame, rather than process the samples you store them with interframeBuffer.push_back(sample). When you start processing the next frame, you first do:
for (const Sample& sample : interframeBuffer) {
process(sample);
}
interframeBuffer.clear();
You should use a single vector the whole time, clearing it and repopulating it as needed, to avoid memory allocation. That's why we call reserve() at the top--to avoid latency later on. Calling clear() doesn't release the memory, it just resets the size() to zero.
I am writing a C program that reads serial data from COM3 (these data are actually pixel intensities of a stream of video frames); once one frame is received completely, the program re-assembles the frame and displays it using OpenGL; next frame comes, display the next frame. (so in the end it looks like a video)
To me, it seems that I need one thread to receive data and another thread to display? Since the program must not stop receiving data.
I have finished data receiving and frame re-assembling part but I have no idea how the display part work.
Can anybody give me any clue how to do this?...
No, you don't have to do this on different threads. Consider this pseudocode:
while (true) {
if (data_present())
read_data();
display();
}
From what I understood from your question, you want to present raster data on the screen. In this case, instance the data in contiguous memory buffer, create a texture of it and render it on a quad or two triangles covering the whole screen.
I am attempting to create a DirectShow source filter based on the pushsource example from the DirectShow SDK. This essentially outputs a set of bitmaps to a video. I have set up a filter graph which uses Async_reader with a Wave Parser for audio and my new filter to push the video (the filter is a CSourceStream and I populate my frames in the FillBuffer function). These are both connected to a WMASFWriter to output a WMV.
Each bitmap can last for several seconds so in the FillBuffer function I'm calling SetTime on the passed IMediaSample with a start and end time several seconds apart. This works fine when rendering to the screen but writing to disk results in a file which is too short in duration. It seems like the last bitmap is being ignored when writing a WMV (it is shown as the video ends rather than lasting for the intended duration). This is the case both with my filter and a modified pushsource filter (in which the frame length has been increased).
I've seen additional odd behaviour in that it was not possible to have a video that wasn't a multiple of 10 seconds in length at one point whilst I was trying to make this work. I'm not sure what this was, but I though I'd mention it incase it's relevant.
I think the end time is simply ignored. Normally video samples only have a start time because they are a point in time. If there is movement in the video, the movement is fluent, though the video are just points in time.
I think the solution is simple. Because video stays the same until the next frame is received, you can just add a dummy frame at the end of your video. You can simply repeat the previous frame.
I'm doing some processing on some very large video files (often up to 16MP), and I need a way to store these videos in a format that allows seeking to specific frames (rather than to times, like ffmpeg). I was planning on just rolling my own format that concatenates all of the individually zlib compressed frames together, and then appends an index on the end that links frame numbers to file byte indices. Before I go about this though, I just wanted to check to make sure I'm not duplicating the functionality of another format/library. Has anyone heard of a format/library that allows lossless compression and random access of videos?
The reason it is hard to seek to a specific frame in most video codecs is that most frames depend on another frame or frames, so frames must be decoded as a group. For this reason, most libraries will only let you seek to the closest I-frame (Intra-frame - independently decodable frame). To actually produce an image from a non-I-frame, data from other frames is required, so you have to decode a number of frames worth of data.
The only ways I have seen this problem solved involve creating an index of some kind on the file. In other words, make a pass through the file and create an index of what frame corresponds to a certain time or section of the file. Since the seeking functions of most libraries are only able to seek to an I frame so you may have to seek to the closest I-frame and then decode from there to the exact frame you want.
If space is not of high importance, I would suggest doing it like you say, but use JPEG compression instead of zlib as it will give you a lot higher compression ratio since it exploits the fact you are dealing with image data.
If space is an issue, P frames (depend on previous frame/frames) can greatly reduce the size of the file. I would not mess with B frames (depend on previous and future frame/frames) since they make it much harder to get things right.
I have solved the problem of seeking to a specific frame in the presence of B and P frames in the past using ffmpeg (libavformat) to demux the video into packets (1 frame's worth of data per packet) and concatenate these into a single file. The important thing is to keep and index into that file so you can find packet bounds for a given frame. If the frame is an I-frame, you can just feed that frame's data into an ffmpeg decoder and it can be decoded. If the frame is a B or P frame, you have to go back to the last I-frame and decode forward from there. This can be quite tricky to get right, especially for B-frames since they are often sent in a different order than how they are displayed.
Some formats allow you to change the number of key frames per second.
For example, I've used ffmpeg to encode to flv at 25 frames per second with 25 key frames per second, and then used a player that was fine in moving to key frames. Basically this allowed me to do frame by frame seeking.
Also the last time I checked quicktime can do frame by frame seek without having to have each frame being a key frame.
May not be applicable to you but that's my thoughts.