I am working on a project that is audio steganography and i need any new scheme along with its implementation. My input is text that will be encoded in the audio file. I want some suggestion on new encoding scheme related to phase coding. I have thought of using the complete intial phase to hide the text to increase the data transmission. Can anyone help me with the code.
Here's a technical paper where professionals describe methods of Audio Steganography:
http://airccse.org/journal/jma/3311ijma08.pdf
Related
I have built a source client using Portaudio and LAME which streams the microphone input to an Icecast server to be listened to online via the HTML5 tag. I have managed to (supposedly) get the quality of the stream to MP3 320kbps at 44.1kHz and am looking for a way to confirm this using tests and or benchmarks.
I have an indication that these stats are somewhat correct from looking at stream inspectors in software such as iTunes and VLC, but I am looking to get a more in-depth data set.
What I basically want is to be able to test how much of the original file is being lost over the stream and if or how much the quality changes depending on environmental conditions of the broadcaster or streamer.
Does anyone know of any tools, frameworks to get some hard numbers or representations of this data?
If VLC tells you the stream is 320kbit CBR, then it is.
It sounds like what you're looking for is a comparison of the actual audio content. This is highly subjective. MP3 is built to use features of how our hearing works to save bandwidth. For example, quiet sounds are masked by loud sounds. High frequencies are harder to hear and are simply rolled off.
You can compare the spectral analysis between the original PCM-sampled waveform and the MP3 decoded waveform, but this doesn't tell you how humans interpret that sound. For that, you would have to survey humans.
I have an C-Code for a video codec. It takes in a compressed format as an input and give out a YUV data buffer. As a standalone application i'm able to render the YUV generated using OpenGL.
Note: This codec is currently not supported by VLC/gstreamer.
My task now is to create a player using this code (that is with features such as play, pause, step, etc.). Instead of re-inventing the whole wheel, i think it would be better if i'm able to integrate my codec into gstreamer player code(for Linux).
Is it possible to achieve the above? Is there some tutorial using which i can proceed? I have searched a lot on net but was unable to find anything specific to my requirement. Any information or links specific to the above problem will be of great help to me. Thanks in advance.
-Regards
Since the codec and container are of new MIME types, you will have to implement a new GstElement for demuxer and codec. A simple example (for audio) is available in this location. I presume this should provide a good starting reference for you.
Some additional links:
To create a decoder plugin, you can refer to the vorbisdec implementation.
To create a demuxer, you can refer to the oggdemuxer implementation.
Reference to factory make
I'm wondering if it's possible to draw an audio channel of a video or audio file as an image using ffmpeg, or if there's another tool that would do it on Win2k8 x64. I'm doing this as part of an encoding process after a user uploads a video or audio file.
I'm using ColdFusion 10 to handle the upload and calling cfexecute to run ffmpeg.
I need the image to look something like this (without the horizontal lines):
You can do this programmatically very easily.
Study the basics of FFmpeg. I suggest you to compile this sample. It explains how to open a video/audio, identify the streams and loop over the packets.
Once you have the data packet (in this case you are interested only in the audio packets). You will decode it (line 87 of this document) and obtain the raw data of an audio. It's the waveform itself (the analogue "bitmap" for an audio).
You could also study this sample. This second example is how to write a video/audio file. You don't want to write any video, but with this sample you can easily understand how the audio raw data packet works, if you see the functions get_audio_frame() and write_audio_frame().
You need to have some knowledge about creating a bitmap. Any platform has an easy way to do that.
So, the answer for you: YES, IT IS POSSIBLE TO DO THIS WITH FFMPEG! But you have to code a little bit in order to get what you want...
UPDATE:
Sorry, there are ALSO built-in features for this:
You could use those filters... or
showspectrum, showwaves, avectorscope
Here are some examples on how to use it: FFmpeg Filters - 12.22 showwaves.
now i'm working on a project for creating audio unit instrument that provide the basic waveform and also provide the audio sampler. i have a problem with how to implement the audio unit instrument base to support the audio file browsing and also wonder about the Audio unit SDK that support this situation to making a sampler.
the sampler can combine with wave form then generate the new sound
This is not an IOS audio unit. and i have not much knowledge about this sampler structure
i have been search a lot, but their are no related knowledge and some source code that i can understand. pls help me for at least browsing the audio file from Au Instrument and slicing the audio data in a time domain. so i can use DSP to work with it.
regard.
I suggest taking a look at the FilterDemo source code. It illustrates the most important aspects of the relationship between parameters, properties, UI, and the underlying DSP code. I have had some success with using the FilterDemo source code as a basis for converting raw DSP code, as well as AU plugins with only generic parameters (and therefore no UI), into fully integrated AU plugins with customized UI.
Also, pay close attention to the warnings, embedded in some of the source code, about renaming your UI elements, as there is a flat namespace to contend with.
I'm trying to develop an online application where the user writes some text and the software sings it back to the user.
I can currently generate the audio file with the words spoken by the computer using espeak, but I have no idea how to make it sound like a song, how to add rhythm to it.
I'm able to change the pitch and tempo using rubberband, but that's as far as I've gotten.
Does anyone have a clue how to make this happen?
If you want to use rubberband to change duration and pitch, then I think the hard part is going to be mapping from phonemes/syllables in the text to corresponding audio ranges in the speech systhesis output, for which I have no simple suggestion. (Ideally you'd get inside the speech synthesiser so that it would provide you with the mapping from phonemes to audio location.)
A simpler alternative might be to try Speech Synthesizer Markup Language - SSML. It has a "pitch" and "duration" elements that can absolutely specify pitch in Hz and duration in seconds. You can also specify volume, for controlling dynamics.
Given this, you could try to convert the text into a SSML document, and mark up words/syllables/phonemees with pitch/duration and volume attributes.
I've ended up using Festival's singing mode. It sounds reasonably well, except for the fact it only works with English voices.