I'm looking to set the playback rate of an audio file. Is there any way to do this for both iOS and Android? I could fork the repository and set it up myself, but I'm hoping to avoid that as it could take some time.
I'm currently using cocos2d-x v2.2.2.
Thanks!
Since Cocos2d-x v2.2.2 has no such API for changing playback rate. Check API for SimpleAudioEngine :
And you don't want to write your own code. Therefore simple answer is you can't change playback rate. You might try newer version Cocos2d-x v3.0 beta or you have to write your own code using API's provided by respective platforms.
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I am looking for an example code on how to capture microphone audio using Naudio + WASAPI.
(I am not interested in direct to disk recording, what i need is to process the input buffer in realtime in order to do some audio effects.)
I've searched a lot, but could not find any decent sample online.
Can you please help?
P.S. BASS library and C# examples are welcome as well!
The NAudio source code comes with a demo app that shows how to capture audio using WASAPI. Look in NAudioDemo\RecordingDemo\RecordingPanel.cs.
MSDN has a lot of code samples, though not covering NAudio they do have a few samples that show in detail how to use the Windows Audio Session API.
Since WASAPI is a native-only API there are both sample projects that show you how to use that API from a native-only app Here as well as samples that show you how to build a native component that wraps the API for consumption from a CSharp application. I couldn't find the direct link to the C#/C++-sample but it's included in the Windows 8 App Samples package. Then there's the option of writing a managed wrapper for the API altogether but unless you enjoy pain and are looking for an adventure in marshaling I wouldn't recommend it...
If you're developing for Windows Phone then there's a VOIP-sample in the WP8 SDK that covers how to capture and render PCM audio data using WASAPI.
As Mark pointed out, the size of the pcm data buffer might differ over time and this is in part due to the fact that WASAPI is a low-latency Audio API and therefore has as little abstraction between the consumer (your app) and the producer (the driver) as possible. Though there's nothing that stops you from doing some fixes size buffering of your own and only pass on the data to your app when your own buffer is full.
I'm working on a C++ application which takes microphone input, processes it, and plays back some audio. The processing will incorporate a database located on a server. For ease of creating UI and for maximum portability, I'm thinking it would be nice to have the front end be done in HTML. Essentially, I want to record audio in a browser, send that audio to the server for processing, and then receive audio from the server which will then be played back inside the browser.
Obviously, it would be nice if HTML5 supported microphone input, but it does not. So, I will need to create a plugin of some kind in order to make this happen. NPAPI scares me because of the security issues involved, so I was looking into PPAPI and Native Client. Native Client does not yet support microphone input, and I believe that the PPAPI audio input API would be limited to a dev build of Chrome. FireBreath doesn't look like it supports any microphone function either. So, I believe my options are:
Write my own NPAPI plugin to record the audio
Use Flash to get microphone input
Bail on browsers altogether and just make a native application
The target audience for this is young children and people who aren't computer-adept. I'd like to make it as portable and simple to use as possible. Any suggestions?
If you can do it all in Flash and have the relevant knowledge, that would probably be the best solution:
You can avoid writing platform-specific code, delivery/updating is easy and Flash has broad coverage so users don't need to install any custom plugins.
FireBreath doesn't look like it supports any microphone function either.
You can write your own (platform-dependent) code for audio recording with FireBreath, just like you could in a plain NPAPI plugin. FireBreath just makes it easier for you to write the plugin, the result is still a NPAPI (and ActiveX) plugin with access to native APIs etc.
You can use Capturing Audio & Video features in HTML5, see this link for more information.
We are developing an application which takes audio from the microphone and does some analysis. We found during the analysis, that AGC is implemented on the microphone subsystem. Also I have heard that VAD is used.
Are there any other post processing done on the audio(PCM) before it is delivered to the application?
Is it possible for the application to disable the AGC and VAD post processing? Is it possible in JavaME or using some proprietary API, such as Nokia/Samsung?
See my answers to my own questions:
Unknown.
Impossible in JavaME. If you are working on Symbian/S60
devices, you could check if Qt or Symbian C++ has such capability. For example, I found the following info on the web, but did not check it: "There is an API called SetGain/GetMaxGain in CMdaAudioInputStream, but in S60 phones the range is between 1-1, so not very useful using this API. But you can use CVoIPAudioUplinkStream which allows you to dynamically control the audio gain and other codec properties". Try if you are interested in...
I'm working on a desktop application built with XNA. It has a Text-To-Speech application and I'm using Microsoft Translator V2 api to do the job. More specifically, I'm using is the Speak method (http://msdn.microsoft.com/en-us/library/ff512420.aspx), and I play the audio with SoundEffect and SoundEffectInstance classes.
The service works fine, but I'm having some issues with the audio. The quality is not very good and the volume is not loud enough.
I need a way to improve the volume programmatically (I've already tried some basic solutions in CodeProject, but the algorithms are not very good and the resulting audio is very low quality), or maybe use another api.
Are there some good algorithms to improve the audio programmatically? Are there other good text-to-speech api's out there with better audio quality and wav support?
Thanks in advance
If you are doing off-line processing of audio, you can try using Audacity. It has very good tools for off-line processing of audio. If you are processing real-time streaming audio you can try using SoliCall Pro. It creates virtual audio device and filters all audio that it captures.
I'm writing a cross-platform program that involves scrolling a waveform along with uncompressed wav/aiff audio playback. Low latency and accuracy are pretty important. What is the best cross-platform audio library for audio playback when synchronizing to an external clock? By that I mean that I would like to be able to write the playback code so it sends events to a listener many times per second that includes the "hearing frame" at the moment of the notification.
That's all I need to do. No recording, no mixing, no 3d audio, nothing. Just playback with the best possible hearing frame notifications available.
Right now I am considering RTAudio and PortAudio, mostly the former since it uses ALSA.
The target platforms, in order of importance, are Mac OSX 10.5/6, Ubuntu 10.11, Windows XP/7.
C/C++ are both fine.
Thanks for your help!
The best performing cross platform library for this is jack. Properly configured, jack on Linux can outperform Windows asio easily (in terms of low latency processing without dropouts). But you cannot expect normal users to use jack (the demon should be started by the user before the app is started, and it can be a bit tricky to set up). If you are making an app specifically for pro-audio I would highly recommend looking in to jack.
Edit:
Portaudio is not as high-performance, but is much simpler for the user (no special configuration should be needed on their end, unlike jack). Most open source cross platform audio programs that I have used use portaudio (much moreso than openal), but unlike jack I have not used it personally. It is callback based, and looks pretty straightforward though.
OpenAL maybe an option for you.