I am trying to implement a file converter using libav. Currently I am testing just the audio convertion (for example aac input and mp3 output). I am using the code from this question Conversion from mp3 to aac/mp4 container (FFmpeg/c++) but the resulting output file sounds corrupted (too slow, noisy, distorted).
This is the result from converting AAC to mp3. AAC input format is AV_SAMPLE_FMT_FLTP and mp3 output format is AV_SAMPLE_FMT_S16P. It looks like part of the one channel is inverted on the other channel?
I am using avresample to convert the audio data. I just cant find out what is wrong, i already looked through the examples but as i can see i am doing it just like in the examples.
Here is my converter class:
http://pastebin.com/c6hvrRaM (.h)
http://pastebin.com/u6iAPHZ9 (.cpp)
I know this is pretty much but I am desperated...
Related
I am trying to convert the Ogg file to mp3/wav formats. I used:
FFmpeg
pyaudio
dlls
But nothing worked out.
Also, I am trying to first read the ogg data from an HTTP URL then want to convert it to mp3/wav, and then using speech_recognition converting to text.
If I don't use any method I get the following error.
Error: Audio file could not be read as PCM WAV, AIFF/AIFF-C, or Native FLAC; check if the file is corrupted or in another format.
Please suggest any libraries.
Code snippet:
audio_data = Data.get("audio")
if '.wav'or'.mp3' not in audio_data:
file = ("newspeech.mp3")
new_audio = urllib.request.urlretrieve(audio_data,file)
I am trying to use libav to convert an MP4 file to an MP3 file. Basically trying to achieve what ffmpeg -i filename.mp4 filename.mp3 does. I've found this official example. But when I run it with an input MP4 and an output MP3 I get an error:
Invalid audio stream. Exactly one MP3 audio stream is required.
I am not at all familiar with this library but I think I have narrowed the problem down to this line:
ret = avcodec_parameters_copy(out_stream->codecpar, in_codecpar);
It seems to copy all streams for a video file but we only need one for the MP3 file? I am not sure. There doesn't seem to be a function to copy only the parameters relevant to audio. I checked the sources, avcodec_parameters_copy does a simple memcpy.
Questions:
Is this the actual problem?
How do I solve it?
Am I on the right track to achieve the goal of extracting audio from a video file? I've seen this question (and other similar questions like this and this) on here but none seem to have a complete code example. The C API documentation for this library is also a little lacking.
You can have multiple audio tracks in mp4 file, but only one such track in an mp3 file. The easiest fix for the remuxing example would be to replace lines 101-103:
if (in_codecpar->codec_type != AVMEDIA_TYPE_AUDIO ||
stream_index != 0) {
This, naturally, is relevant only if the output is mp3.
PS, make sure that your input mp4 uses the MP3 audio codec. If it does not (and most have AAC or AC3 these days), it's not enough to remux the file, you also need to decode and re-encode the audio stream.
I have a simple encoding/ decoding application using Windows Imaging Component API. The issue I'm having is that when I use either the JPEGXR or BMP formats, everything works fine. However, when I use the JPEG codec - the encoder works fine and I can visually verify the generated JPEG image, but when I try to decode that stream, I get a WINCODEC_ERR_BADHEADER (0x88982f61)
Here's the line that fails:
hr = m_pFactory->CreateDecoderFromStream(
pInputStream,
NULL,
WICDecodeMetadataCacheOnDemand,
&pDecoder);
Here pInputStream is an IStream created from a byte array (output of the encoder - a black box which outputs a byte vector).
Please help! This is driving me nuts!
When passing stream as an argument, make sure to pre-seek it to proper initial position (esp. seek it back to the beginning if you just wrote data into it and expect further retrieval). APIs are typically not expected to seek, because this way they let you provide data in the middle of a bigger stream.
Thanks for taking some time to read my question.
I'm developping a C++ application using Qt and windows API.
I'm recording the microphone output in small 10s audio files in raw format, and I want to convert them to aac format.
I have tried to read as many things as I could, and thought it would be a great idea to start from windows media foundation transcode API.
Problem is, I can't seem to use a .raw or .pcm file in the "CreateObjectFromUrl" function, and so I'm pretty much stuck here for the moment. It keeps on failing. The hr return code equals 3222091460. I have tried to pass an .mp3 file to the function and of course it works, so no url-human-failure involved.
MF_OBJECT_TYPE ObjectType = MF_OBJECT_INVALID;
IMFSourceResolver* pSourceResolver = NULL;
IUnknown* pUnkSource = NULL;
// Create the source resolver.
hr = MFCreateSourceResolver(&pSourceResolver);
if (FAILED(hr))
{
qDebug() << "Failed !";
}
// Use the source resolver to create the media source.
hr = pSourceResolver->CreateObjectFromURL(
sURL, // URL of the source.
MF_RESOLUTION_MEDIASOURCE, // Create a source object.
NULL, // Optional property store.
&ObjectType, // Receives the created object type.
&pUnkSource // Receives a pointer to the media source.
);
The MFCreateSourceResolver works fine, but CreateObjectFromURL does not succeed :(
So I have two questions for you folks :
Is it possible to encode raw audio files to aac files using windows media foundation ?
If yes, what should I read to accomplish what I want ?
I want to point out that I can't just use ffmpeg or libav because I can't afford any license for my software, and don't want it to be under the GPL license. But if there are alternatives to windows media foundations to encode raw audio files to aac, I would be glad to hear them.
And finally, sorry for my bad english, this is obviously not my native language and I'm sorry if I made your eyes bleed. (and happy if I made you laugh)
Have a nice day
The hr return code equals 3222091460
Those are HRESULT codes. Use this "ShowHresult" tool to have them conveniently decoded for you. The code means 0xC00D36C4 MF_E_UNSUPPORTED_BYTESTREAM_TYPE "The byte stream type of the given URL is unsupported."
The problem is basically that there is no support for these raw files, .WAV is a good source for raw audio - the file holds both format descriptor and the payload.
You can obviously read data from the raw audio file yourself and compress into AAC using Media Foundation's AAC Encoder via its IMFTransform interface. This is reasonably easy and you have AAC data on the output to e.g. write into raw .AAC.
Alternate options to Media Foundation is DirectShow (there are suitable codecs, though I thought it might be not so easy to start), libfaac, FFmpeg's libavcodec (available under LGPL, not GPL).
I want to create a program, which gets a video-file from Qt, converts that video file to TIFF-files and sends them to an algorithm which handles these TIFF-Files.
My questions:
is it possible with ffmpeg or avcodec not to convert a video-file to TIFF-files first on harddrive and send them to the algorithm after that, but to convert frame for frame and send it to the algorithm right away?
The more important question: Is it possible to do that not with an external process with ffmpeg.exe, but with ffmpeg.dll? Or is it only possible with avcodec.dll? (It doesn't have to be "on-the-fly" like at my point above) How can I create a ffmpeg.dll with header and lib?
for exporting tif :
http://www.repaire.net/forums/cinema-numerique/215306-projet-dencodage-dcp.html
Creating a tiff from second 29 in a mpeg, using ffmpeg dd201110 can be done like this:
ffmpeg -i 'test.mpg' -vframes 1 -compression_level 0 -ss 29 'test.tiff'
YMMV :-D
If you dont want to store the image as a file, take a look at ffmpeg-php
http://ffmpeg-php.sourceforge.net/
$movie->getFrame([Integer framenumber])
Returns a frame from the movie as an ffmpeg_frame object.
$frame->toGDImage()
Returns a truecolor GD image of the frame.
There may be C code underneath you can reuse..