There are examples on Qt website regarding using audio API, but frankly I don't really understand them at all.
What I was imagining is writing array of values (bytes, integers...) into some audio buffer and have the sound card "play" them (actually DAC them).
Pseudocode:
// Square wave?
const int values[] = {255,255,255,255, 0,0,0,0, 255,255,255,255 ...};
// Create output that will buffer the bytes and put them on digi to analog converter
RawAudioOutput output(BIT_RATE_CONSTANT, ... some other parameters ...);
output.start();
output.writeBytes(values, sizeof(values));
Can I accomplish something like that? How would I go about it? I know I can model square wave in Audacity (doesn't sound nice), so I guess it's possible. How?
In Qt, if you want to write an array of values into an audio buffer, the class for that is QAudioOutput. The format of the array of values can vary, the PCM format should be supported by all platforms.
Qt ships with an example that demonstrates the usage of QAudioOutput, have a look at that. In the example, the Generator::generateData() function creates the array of values that are then later sent to the audio device.
Of course playing audio from an array of values is quite low-level. With QMediaPlayer, Qt also provides a high-level class to play sound files (.wav, .mp3), video files and even streams.
Related
So... I'm dealing with a system that has input data coming in buffers (i.e. NOT a file). I want to determine which decoder to create to decompress an audio stream (MP3, WAV, OGG, ...) So obviously I do not know the input format.
I found out that I could determine the format using the av_probe_input_format[23]() functions. That part works great, I get a format pointer that matches the files that I use as input.
AVInputFormat * format(av_probe_input_format3(&pd, true, &score));
I can print the format->name and format->long_name and these are the correct type (so the detection is working as expected).
Now, I'm trying to understand how to convert that AVInputFormat * into a AVCodec * so I can call avcodec_alloc_context3(codec) to create the actual audio decoder.
I found a couple of functions, which I used like so:
AVCodecID const codec_id(av_codec_get_id(format->codec_tag, format->raw_codec_id));
AVCodec * codec(avcodec_find_decoder(codec_id));
Problem 1. the raw_codec_id field is marked as "private" (should not access/use anywhere in your client's code).
Problem 2. the first function always returns AV_CODEC_ID_NONE (0) so of course the second call fails each time.
Am I doing something wrong? Is there is way to instead create a generic decode that will automatically detect the type of audio I have as input? (that is, would that be the only way to make that work?)
Okay, so the fact is that trying to use these functions directly is pretty much futile. The problem I have with the design is that it forces me to actually have a callback and that callback forces me to have a thread (i.e. I have to somehow feed data from a stream, not a file or such!)
So I can use the avformat_open_input() as mentioned by Gyan, only I have to have my own AVIOContext. I was hoping I could just call functions with my incoming data and avoid the pipeline concept. The issue here is some background processes could be servers that use fork() and thus you need to be really careful (i.e. fork() is not friendly with threads).
How can I use NAudio library to get sample float array from mp3 file?
Here is my code:
float[] buffer = new float[2000];
AudioFileReader reader = new AudioFileReader(filePath);
reader.Read(buffer, 0, 2000);
After that buffer is always empty (only zeros inside).
You could also provide me another useful library in C# to realize this.
You're reading the first 2000 samples which is only going to be around 20ms of audio, so it's quite possible that your MP3 starts with a bit of silence. Have you tried reading further into the file?
After making a small video recorder application and being able to play that video again. I would like to make the possibility to pick X seconds from the video and put that into a new .MP4 file (or overwrite the old one, that would be even better).
I am using the MoSync C++ Native UI and VideoViewer. I know I can get the position and that part is all fine, and according to the MoSync documentation.
char buf[BUFFER_SIZE];
maWidgetGetProperty(videoViewHandle,
MAW_VIDEO_VIEW_CURRENT_POSITION,
buf,
BUFFER_SIZE);
int seconds = 5;
//So here I need to make a new file ranging from buf to buf + seconds
However, I have absolutely no clue as to where to look for this. Should I use mp4 header files and create my own mp4 (how is this even done? and is this cross-compatible?).
Will appreciate any advice/help you can offer!
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).
For the implementation of a Windows based page-flip application I need to be able to convert a large number of PDF pages into good quality JPG, not just thumbnails.
The aim is to achieve the best quality / file size for that, similar to Photoshops Save for Web does that.
Currently Im using Datalogics Adobe PDF Library SDK, which does not seem to be able to fullfil that task. I am thus looking for an alternative commcerical C++ or Delphi library which provides a good qualtiy / size / speed.
After doing some search here, I noticed that most posts are about GS & Imagekick, which I have also tested, but I am not satisfied with the output and the speed.
The target is to import the PDFs with 300dpi and convert them with JPG quality 50, 1500px height and an ouput size of 300-500kb.
If anyone could point out a good library for that task, I would be most greatful.
The Gnostice PDFtoolKit VCL may be a candidate. Convert to JPEG is one of the options.
I always recommend Graphics32 for all your image manipulation needs; you have several resamplers to choose. However, I don't think it can read PDF files as images. But if you can generate the big image yourself it may be a good choice.
Atalasoft DotImage (with the PDF rasterizer add-on) will do that (I work on PDF technologies there). You'd be working in C# (or another .NET) language:
ConvertToJpegs(string outfileStem, Stream pdf)
{
JpegEncoder encoder = new JpegEncoder();
encoder.Quality = 50;
int page = 1;
PdfImageSource source = new PdfImageSource(pdf);
source.Resolution = 300; // sets the rendering resolution to 200 dpi
// larger numbers means better resolution in the image, but will cost in
// terms of output file size - as resolution increases, memory used increases
// as a function of the square of the resolution, whereas compression only
// saves maybe a flat 30% of the total image size, depending on the Quality
// setting on the encoder.
while (source.HasMoreImages()) {
AtalaImage image = source.AcquireNext();
// this image will be in either 8 bit gray or 24 bit rgb depending
// on the page contents.
try {
string path = String.Format("{0}{1}.jpg", outFileStem, page++);
// if you need to resample the image, this is the place to do it
image.Save(path, encoder, null);
}
finally {
source.Release(image);
}
}
}
There is also Quick PDF Library
Have a look at DynaPDF. I know its pretty expensive but you can try the starter pack.
P.S.:before buying a product please make sure it meets your needs.