I'm making an iPhone game application using Core Audio, Extended Audio File Services.
It works OK, but when I first call AudioOutputUnitStart, it takes about 1-2 seconds.
After the second call, no problem.
For a game application, 1-2 seconds is very noticeable.
(I tested this on iPhone simulator, and iPhone 3GS)
Also, if I leave the game for about 10 seconds, first call of AudioOutputUnitStart after the leave also takes time.
I'm using a 5-second AAC 160kbps file to play.
For a 5-second file, why not convert that file into a linear PCM wav or aif file? It shouldn't be that much bigger, plus you don't have to involve the AAC decoder in the process.
Related
We have a VOIP application, that records and plays audio. As such, we are using PlayAndRecord (kAudioSessionCategory_PlayAndRecord) audio session category.
So far, we have used it successfully with iPhone 4/4s/5 with both iOS 6 and iOS 7 where call audio and tones played clearly and were audible.
However, with iPhone 5s, we observed that both the call audio and tones sound robotic/garbled in speaker mode. When using earpiece/bluetooth/headset, sound is clear and audible.
iOS Version used with iPhone 5s: 7.0.4
We are using audiounits for recording/playing of call audio.
When setting audio properties like session category, audio route, session mode etc., we tried both the older (deprecated) AudioSessionSetProperty() and AVAudioSession APIs.
For playing tones, we are using AVAudioPlayer. Playing of tones during the VOIP call and also when pressing keypad controller within the app produces robotic sound.
When instantiating the audio component using AudioComponentInstanceNew, we set componentSubType to kAudioUnitSubType_VoiceProcessingIO.
When replacing kAudioUnitSubType_VoiceProcessingIO with kAudioUnitSubType_RemoteIO, we noticed that the sound of call audio and tones was no longer robotic, it was quite clear, but the volume level was very low when using speaker mode.
In summary, keeping all the other audio APIs the same:
kAudioUnitSubType_VoiceProcessingIO: Volume is high (desirable) but sound of tones and call audio was robotic in speaker mode.
kAudioUnitSubType_RemoteIO: Sound of tones and call audio was clear but it is not audible.
STEPS TO REPRODUCE
- Set audio session category to playAndRecord.
- Set audio route to speaker
- Set all the other audio properties like starting audio unit, activating the audio session, instantiating the audio components.
- Set the input and render callbacks
- Try both options
1. Play tones using AVAudioPlayer
2. Play call audio
Any suggestions on how to get over this issue. Raised as an issue with Apple but no response yet from them.
i have shared the code here github link
The only difference between kAudioUnitSubType_VoiceProcessingIO and kAudioUnitSubType_RemoteIO is that voiceProcessing includes code to tune out acoustic echo i.e. tunes out the noise from the speaker so the microphone doesn't pick it up. Its been a long time since I've played with the audio framework but I remember that to sound off there could be any number of things,
Are you doing any work in the audio callbacks that could be taking a long time?
The callbacks run on realtime threads. if your processing takes too long you can miss data. Would be helpful to track the data over a fixed period of time to see are you capturing it all. Use something like wireShark to sniff the network. Record the number of packets and see did the phone capture the same.
Are you modifying any of the audio?
Do you have a circular buffer that might be causing an issue?
I've had several issues doing this and one was using a third party circular buffer that was described as low latency and efficient ... it wasn't. I answered my own question here and included my circular buffer implementation that greatly improved my audio as the issue was I was skipping data.
Give this a go and let me know:
iOS UI are causing a glitch in my audio stream
Please be aware that some of this code is unique to the audio format ALaw, 0xD5 is a byte of silence in ALaw, if you are using linear PCM or any other that will probably be a noise of some kind.
I am trying to see if it is possible to record a video from the iPhone's camera and write this to a file. I then want the video to start playing on the screen a set time after. This all needs to happen continuously. For example, I want the video on the screen to always be 20 seconds behind what the camera is recording.
Some background:
I have a friend who is a coach and would like for his players to be able to see their last play. This could be accomplished by a feed going to a TV from an iPad always 20 seconds behind what is recorded. This needs to continually run until practice is over. (I would connect the iPad to the TV either with a cable or AirPlay to an Apple TV). The video would never need to be saved and should just be discarded after playing.
Is this even possible with the APIs AVFoundation offers? Will the iPhone let you write to a file and read from a file at the same time to accomplish this? Any other better way to accomplish this?
Thanks for your time.
Instead of writing to a file, how about saving your frames in a circular buffer big enough to hold X seconds of video?
The way I would start to do this would be to look at what's provided in AVCaptureVideoDataOutput and its delegate methods (where you can get the frame data from).
I am playing a song in Cocos2D iphone using this line.
[[SimpleAudioEngine sharedEngine] playEffect:#"song.mp3"];
Suppose the mp3 is 1 min long, I want to play the song from 10 sec to 20 sec. Is it possible in Cocos2D iphone. Please help.
I don't think it is possible to "play the song from 10 sec to 20 sec" using the audio engine of cocos2d-iphone (which is named CocosDeshion).
SimpleAudioEngine is the main interface to CocosDeshion. Basically in cocos2d development we use audio files in two way - sound effect (short) and background music (long and usually loop). playEffect: is for the short ones, while playBackgroundMusic: and playBackgroundMusic:loop: is for the long ones. preloadBackgroundMusic: provides you the ability to cache the music in memory to avoid lag while playing, since they are often in compressed format.
As you can see in the documentation of SimpleAudioEngine (link above), there isn't such method for you to play a part of an audio file. Even the underlying class CDAudioManager doesn't support this.
I suggest to extract the 0:10 - 0:20 part as a separate audio file. But if you have a special reason to do this, you may want to use audio playback interfaces (such as AVAudioPlayer) in iOS SDK directly. Please refer to: http://developer.apple.com/library/ios/#documentation/AudioVideo/Conceptual/MultimediaPG/UsingAudio/UsingAudio.html
I'm struggling with an AudioQueue audio player I implemented. I initially thought it was truncating the 1st 1/2 of audio that it played but upon loading larger files I notice gaps every other 1/2-1 second. I've run it in debug and I've confirmed that I'm loading the queue correctly with audio. (There are no big zero regions loaded in the queue.) It plays without issue (no gaps) on the simulator but on device I get gaps as if its missing every other chunk of audio. In my app I decompress then pull audio from a memory NSMutableData object. I feed this data into the audio queue. I have a corresponding implementation in the same app that plays wave audio and this example works without issue on long and short audio clips. I'm comparing the wave implementation to the other which does the decompression. the only difference between the two is how I discover the audio meta data and where I get the audio samples for enqueuing. In the wave implementation I use AudioFileGetProperty and AudioFileReadPackets to get this data. In the other case I derive the data before hand using cached ivars loaded during callbacks from my decompressor. The meta data matches for both compressed and wave implementations. I've run the code in instruments and I don't see anything taking more than 1ms in my audio packet delivery/enqueuing logic during playback. I'm completely lost. Please speak up if you have any idea how to solve the situation.
I finally resolved this issue. I found that if I skipped the 1st 44 bytes (the exact size of a wave header) of audio then it plays correctly on the device. It pays correctly on the sim regardless of wether I skip 44 or not. Strange and I'm not sure why but that's the way it works.
I'm working on a project which is using Audio Toolbox for recording and playback of PCM data, and I'm having trouble with playback. In the simulator, I can record and play audio just fine, using a custom class to handle storing and sourcing PCM bytes for the recording and playback buffers as needed. On device (iPhone (3.0.1) and iPod 2G (3.1.2)) recording works fine, the audio files produced are correct, but in-app playback stutters, like it's only playing part of each playback buffer. My buffers are one second long, and I've got 3 buffers, which are preloaded before playback starts; stuttering occurs during those first 3 seconds as well, which I think rules out a latency problem.
I've written Audio Toolbox code before that worked, and I'm not doing anything strange here except that I'm using my own class to source PCM data instead of AudioFileReadBytes()
I know the data that comes out of my source is good, because it plays right in the sim, and it writes to disk as a correct audio file
I've played around with sample rates a bit; I'm normally using 11025Hz sampling to cut down on file size (it's all voice, so it sounds fine). at 44100Hz, but with the same size of buffers, I get the same stuttering problem, but the audio segments come a lot faster, about 4 times faster. That's why I think it's only playing part of each buffer.
The only reason I can conceive that it would only play part of each buffer is a latency problem... like the audio toolbox code is running out of full buffers while I'm still filling an empty one. But that would cause it to play the preloaded buffers correctly, and then start stuttering, and that doesn't happen, it stutters the whole way through
I've tried humongous buffers, like 10MB buffers, and I just get silence and a single stutter of audio at the end of playback. I've also tried preloading more buffers than normal, like 10 seconds worth of audio, and it behaves the same.
The audio session is being set with AVAudioSession, not the Audio Toolbox calls, and it's being set to the Playback category for playback
I have no idea how to try and attack this problem, it makes no sense to me that it works fine on the simulator but not the device.
Code for the playing callback and the set up for the audio queue services: http://pastebin.com/mfaa546c
It turns out that the use of NSData's GetBytes:length: was causing the problem. The buffer filled with that method was playing incorrectly. However, doing a memcpy from that buffer to another buffer would prevent the problem.