I have an audio streamer based on ffplay. It works great using ffmpeg 0.11 but when I use ffmpeg 1.0 or the latest 1.2 release the audio seems to be decoded or played weirdly.
Essentially it sounds like chipmunks with mp3 streams and with aac streams I hear tons of static barely hearing the actual stream and the actual stream sounds slow.
Any ideas the possible changes in ffmpeg that could have caused these types of issues?
Similar issue was posted here but no actual answer about what is going on. Supposedly this code reproduces the same issue.
UPDATE 1:
I have done a step by step copy from ffplay and still no luck! :/ The channel and sampling rate look correct so there must be something internally that is returning a weird decoded format?
pCodecCtx->request_sample_fmt = AV_SAMPLE_FMT_S16
doesn't change the behavior! I didn't find any better solution than using swr_convert to convert audio sample from AV_SAMPLE_FMT_FLT to AV_SAMPLE_FMT_S16.
Found a fix... So initially it sounded like something wasn't matching correctly with the decoded raw PCM data and my PCM player. I took a deeper look at what was being returned by the decoder and any potential differences. Turns out that the default decoded format has changed from AV_SAMPLE_FMT_S16 to AV_SAMPLE_FMT_S16P. The fix was to simply specify pCodecCtx->request_sample_fmt = AV_SAMPLE_FMT_S16; before opening the decoder.
Any feedback if this is a bad idea? I'm concerned if there might be issues with other formats and potential performance issues...
Related
I am developing a C# application that records streaming audio to MP3.
I’m new to this but from what I’ve seen so far, the easiest way to do this is record to WAV using NAudio and then create an MP3 version using either LameMP3FileWriter or MediaFoundationEncoder.
I’m running into problems with the conversion, however, as my PC sound system is 5.1 and the MP3 conversion crashes due to the number of channels in the recorded WAV file. It works fine when I reconfigure my sound system to stereo but this is a bit of a pain; firstly, it means I cannot use my 5.1 system when recording the music but more of an issue, for some reason that I cannot figure out, if I set my speakers to stereo, they revert to quadraphonic when the PC (Windows 10) is rebooted!
Can anyone suggest how I can do this conversion without the need to configure my 5.1 sound?
One obvious solution to do something like resampling the WAV file to 2 channels before the conversion to MP3 but that seems something of a ‘long shortcut’. I’m also unclear as to the advantage of recording to WAV in the first place – audio streams are compressed and unlikely to have more than 2 channels to start with so playing it over 5.1, no matter how good it sounds, is really a bit illusory.
It would seem more sensible to just record the stream direct to MP3 but I cannot find any straightforward way of doing that.
Mp3 specification does not handle 5.1. So it seems your mp3 encoders fail with 5.1.
Perhaps you can try an encoder that support MP3 Surround, an mp3 extension for 5.1.
Also, Perhaps you should consider using AAC encondig, a more friendly codec for 5.1.
I need a software that can stream audio with mp3 format.The audio will come from the microphone at the same time.
I have a software that can stream sound with alaw and ulaw codecs.
And I have an another program that can stream recorded mp3 file. Not capture from the microphone.
I can make stream with VLC.Dotnet wrapper but I didn't succeed with directshow.(namely microphone)
Here my Vlc.Dotnet code;
myVlcControl.Play("dshow://");
myVlcControl.Play(new Uri("dshow://"));
It did not work with this codes. I don't know what causes the problem.
My second software can stream sound that captures form microphone in real time. But its codec format is alaw not mp3. I did not find any converter that convert linear to mp3 file. I find a converter that convert linear to alaw. this is the link Linear to Alaw Codec
I know the LAME and NAudio but it converts wav file to mp3. I need linear to mp3(like in the link)
I am very confused. I really do not know which way to go.
1. Find a codec linear to mp3 (It's very complicated) ?
2. Learn VLC direct Show usage on .NET ?
Thank you so much in advance.
*VLC.DotNet, axVLCPlugin21, LAME, ffmpeg....
As soon as I've successfully solved this problem with VLC.Dotnet wrapper. The problem is compiling with x64 architectural. When I was compiled with x86 architectural, The problem solved.
So, I've been working on a project that is going to require the playback of Midi. I have already done this with a little bit of code that looks like this:
mciSendString(L"play C:\\aha.mid", NULL, 0, NULL);
And that works perfectly fine for playing a Midi file.
But, what I also need to do is to get the note events from the playback of the midi file itself. I've seen a great deal of libraries that can help with playback or with reading the raw data of the midifile, but not both, and not both at the same time. I've tried Midifile, did some searching around the Juce library, but nothing seems to get the functionality that I need.
Is there something out there that can solve my problem or am I just thinking about this problem the wrong way? Maybe there's some way to get the current notes being played on the Microsoft wavetable (would be much easier if that's possible).
I'm trying to develop a little application in which you can load a mp3 file and play it in variable speeds! (I know it already exists :-) )
I'm using Qt and C++. I already have the basic player but I'm stuck with the rate thing, because I want to change the rate smoothly (like in Mixxx) without stopping the playback! The QMediaPlayer always stops if I change the value and creates a gap in the sound. Also I don't want the pitch to change!
I already found something called "SoundTouch" but now I'm completely clueless what to do with it, how to process my mp3 data and how to get it to the player! The "SoundTouch" Library is capable of doing what I want, i got that from the samples on the homepage.
How do I have to import the mp3 file, so I can process it with the SoundTouch functions
How can I play the output from the SoundTouch function? (Perhaps QMediaPlayer can do the job?)
How is that stuff done live? I have to do some kind of stream I guess? So I can change the speed during play and keep on playing without gaps. Graphicaly in my head it has to be something that sits between the data and the player, where all data has to go through live, with a small buffer (20-50 ms or so) behind to avoid gaps during processing future data.
Any help appreciated! I'm also open to any another solution then "SoundTouch" as long as I can stay with Qt/C++!
(Second thing: I want to view a waveform overview aswell as moving part of it (around actual position of the song), so I could also use hints on how to get the waveform data)
Thanks in advance!
As of now (Qt 5.5) this is impossible to do with QMediaPlayer only. You need to do the following:
Decode the audio using GStreamer, FFMpeg or (new) QAudioDecoder: http://doc.qt.io/qt-5/qaudiodecoder.html - this will give you raw PCM stream;
Apply SoundTouch or some other library to this raw data to change the pitch. If GPL is ok, take a look at http://nsound.sourceforge.net/examples/index.html, if you develop proprietary stuff, STK might be a better choice: https://ccrma.stanford.edu/software/stk/
Output the modified data into audio device by using QAudioOutput.
This strategy uses Qt as much as possible, and brings you the best platform coverage (you still lose Android though as it does not support QAudioOutput)
I have a application (qt c++) that reads data from USB-device, decodes that data into 24bit RGB pixels which are stored in a uchar array.
Framerate is ~10 FPS. Framesize is 128x4096.
Question is: How to encode these frames into VP8 or h.264 video in real time?
No external processes are allowed, everything needs to run inside my application.
ffmpeg is an option but how to include it to my project and use it? Documentation is rather bad to say the least. Also x264 could be an option but same question as to ffmpeg. And it's also quite expensive, 1$ for unit and minimum of 10000.
Simple guide would be helpful but I doubt there exists one.
Application should run in Windows and Linux.
The problem with the VP8 SDK is that the examples only encode to IVF. That codec appears to have been shut down by Microsoft due to a security flaw (buffer overflow). It's pretty hard to even get the VP8 project setup when you can't even check the results. It at least uses a BSD license scheme and its supposedly unencumbered with patents.
The VP8 SDK has some routines for converting formats, but they are buried in the source tree.
An option not mentioned is the Intel Media SDK, but that locks you to windows.
There is also Theora and Dirac.
X264 has an encoder, but it would be expensive to get a commercial license.
GPLv2 source code is not "free". I don't care what they try to get you to believe.
There is also a project called "Revel - the Really Easy Video Encoding Library". That is a path to getting MPEG-4 part 2 files encoded. H264 is MPEG-4 part 10. H264 is also called AVC. Revel is also GPL'd.
Ffmpeg is a catch all utility that tries to create a wrapper around the various encoders/decoders. If you use the x264 encoder with it, it becomes GPLv2.
The VP8 SDK has documentation and even some sample code