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.
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 using cocos2d's CCRenderTexture to record video of my game. But if recording video in retina display resolution will cost lot of CPU and memory, so I want to use low resolution for video record but keep retina-resolution for normal game play. is it possible?
I've tried "[[CCDirector sharedDirector] enableRetinaDisplay:NO];" during record video, but it seems not work. the generated output totally wrong.
This is not feasible.
You'd have to render each frame twice, once on the screen, then onto the render texture. A serious drop in framerate is inevitable even if you lower the resolution of the render texture somehow.
The reason is simply that you'll also have to write each render texture as an image to flash memory. This is extremely slow. You'll also end up with a huge amount of data. If each (PNG/JPG) image file ends up being a reasonably small 50 KB then one second of recorded data at 60 fps will consume 3 Megabytes of flash memory. One minute would be around 180 Megabytes.
To record a demo of your game, most games follow the simple principle of recording the user input, and then playing back the user input as if the user had issued these commands. This requires careful planning, no breaking changes when updating the app (or invalidating old demos), and no use of non-deterministic randomizers (ie seeded with time).
If you need to record a demo for making a trailer video, there's plenty of screengrabbing solutions around. Some even specialize in grabbing iPhone video, either from the device (usually requires a source code/library component) or from the Simulator.
You should check out Kamcord SDK for recording game play. Check at http://kamcord.com/
Kamcord has a built-in gameplay video and audio recording technology for iOS. It allows you, the game developer, to capture gameplay videos with an API. Your users can then replay and share these gameplay videos via YouTube, Facebook, Twitter, and email.
I'm making a game for iOS platform and i want to use openAL for playing audio effects in the game (except the background music). I want to have for example 20-30 sounds with duration of 1-3 seconds each. Because i want these sounds to be played with no delay i have to load from file, decompress and store in the memory. But decompressed audio (as i understand) uses a lot of memory. So am I on the right way? Or there is another way ? Thanks
Loading sounds does incur a delay, and can cause the rest of the app to jitter. For best performance you definitely want to preload sound effects.
Memory use is a concern, but as long as you're not loading too much audio data into memory at a time, you'll be OK.
44KHz mono audio data will occupy 88,000 bytes per second when uncompressed. Stereo is double that, but usually for sound effects you don't want stereo anyway. So if you had 30 sounds loaded, each of 3 second duration, you'd have 90 seconds of sound using 7.5MB of memory. You can of course halve that memory usage by using a 22050 Hz source before compressing it to AAC (which preserves the source sample rate).
What I do is maintain a cache of audio buffers that I can flush when the app starts to use too much memory like so: https://github.com/kstenerud/ObjectAL-for-iPhone/blob/master/ObjectAL/ObjectAL/OALSimpleAudio.m#L441
For the lowest latency sound effects, you will want to play uncompressed audio already in memory. 2 or 3 MBytes is not a lot of memory to allocate for low latency sound effects.
I am developing a game for the iPhone and iPad using cocos2d, and I need to be able to play a sound exactly when another one completes.
I have a soundtrack that is chopped up in smaller pieces, and there are no room for the tinyest gap between playback when one finishes and one starts.
Btw. I cannot glue the sounds together into a single file and just play that since the order of the files will be rearranged runtime.
How can I achieve this?
With CocosDenshion you can register a delegate with
[[CDAudioManager sharedManager] setBackgroundMusicCompletionListener:self
selector:#selector(musicDidFinish)];
CDAudioManager class reference
This delegate will be called whenever the background music ends. This of course only works if you play your sound files as background music (with the playBackgroundMusic method).
If that doesn't work for you, have a look at ObjectAL. You'll have more options and greater flexibility. For example, with ALSource you can queue multiple ALBuffer objects which represent sound files. That means whenever the source's buffer count decreases to 1 you just queue the next buffer to achieve uninterrupted, sequential playback of multiple sound files (any format).
Because ObjectAL is so awesome (well, I think so :) ) it's included and ready to use in Kobold2D.
You can use a single Audio Queue or the RemoteIO Audio Unit, and just fill the callback buffers with raw/PCM audio samples from any file in any order.
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.