difference between how AVAssetReader and AudioFileReadPackets reads Audio - ios

consider these two scenarios for reading/writing data from Audio files (for the purpose of sending over a network):
Scenario 1: Audio File Services:
Using AudioFileReadPackets from Audio File Services. This generates audio packets that you can easily send over the network. On the receiving side you use AudioFileStreamOpen and AudioFileStreamParseBytes to parse the data.
AudioFileStreamParseBytes then has two callback functions: AudioFileStream_PropertyListenerProc and AudioFileStream_PacketsProc. These guys are called when a new property is discovered in the stream and when packets are received from the stream, respectively. Once you receive the packets, you can feed it to an audio queue using Audio Queue Service which plays the file just fine.
Note: This method does NOT work with music files stored in the iPod library, which brings us to the 2nd scenario:
Scenario 2: AVAssetReader:
With AVAssetReader you can read from the iPod music library and send packets over the network. Typically you would load the packets directly on an Audio Queue similar to above. However, in this scenario you will have to create a thread to ensure that you block receiving packets when the queue is full, and unblock when queue buffers are available (see this example).
Question:
Is it possible to use AVAssetReader to send packets over, only to have it read by AudioFileStreamParseBytes? (the motive would be that the AudioFileStreamParseBytes's callbacks will handle the threading/blocking business and save you that pain). I tried doing it like so:
1. first read the audio file using AVAssetReader
//NSURL *assetURL = [NSURL URLWithString:#"ipod-library://item/item.m4a?id=1053020204400037178"];
AVURLAsset *songAsset = [AVURLAsset URLAssetWithURL:assetURL options:nil];
NSError * error = nil;
AVAssetReader* reader = [[AVAssetReader alloc] initWithAsset:songAsset error:&error];
AVAssetTrack* track = [songAsset.tracks objectAtIndex:0];
// Note: I don't supply an audio format description here, rather I pass on nil to keep the original
// file format. In another piece of code (see here: http://stackoverflow.com/questions/12264799/why-is-audio-coming-up-garbled-when-using-avassetreader-with-audio-queue?answertab=active#tab-top) I can extract the audio format from the track, let's say it's an AAC format.
AVAssetReaderTrackOutput* readerOutput = [AVAssetReaderTrackOutput assetReaderTrackOutputWithTrack:track
outputSettings:nil];
[reader addOutput:readerOutput];
[reader startReading];
2. set up the streamer
// notice how i manually add the audio file type (for the file hint parameter)
// using the info from step one.. If i leave it as 0, this call fails and returns
// the typ? error, which is :The specified file type is not supported.
streamer->err = AudioFileStreamOpen((__bridge void*)streamer,
ASPropertyListenerProc, ASPacketsProc,
kAudioFileAAC_ADTSType, &(streamer->audioFileStream));
3. once I receive the data, I parse the bytes:
streamer->err = AudioFileStreamParseBytes(streamer->audioFileStream, inDataByteSize, inData, 0);
problem: When I do it this way.. I send the bytes and the AudioFileStreamParseBytes does not fail. However, the callbacks *AudioFileStream_PropertyListenerProc* and *AudioFileStream_PacketsProc* are never called. Which makes me think that the parser has failed to parse the bytes and extract any useful information out of them.. in the documentation for AudioStreamParseBytes it states:* You should provide at least more than a single packet’s worth of audio file data, but it is better to provide a few packets to a few seconds data at a time.* I'm sending over 900 bytes, which is just below GKSession's data limit. I'm pretty sure 900 bytes is enough (when testing this under scenario 1, the total bytes was 417 each time and it worked fine).
Any ideas?

The short answer is that it simply doesn't make sense to have packets of audio data be parsed by AudioFileStreamParseBytes.. in the docs AudioFileStreamParseBytes is a function dependent on the existence of an audio file (thus the parameter inAudioFileStream.. which is defined as the ID of the parser to which you wish to pass data. The parser ID is returned by the AudioFileStreamOpen function.)
so lesson learned: don't try to pigeon hole iOS functions to fit your situation.. it should be the other way around.
What I ended up doing was feeding the data directly to an Audio Queue.. without going through all these unnecessary intermediary functions.. a more in depth way would be feeding the data to audio units.. but my application didn't need that level of control

Related

Writing buffers of Streamed mp3 packets to wav file using ExtAudioFileWrite ios

I am working on online radio app I managed to play the streamed mp3 packets from the Icecast server using AudioQueueServices, what I am struggling with is implementing a recording feature.
Since the streaming is in mp3 format I can not write the Audio packets directly to file using AudioFileWritePackets.
To leverage The automatic conversion of Extended Audio I am using ExtAudioWriteFile to write to a wav file. I have setup the AudioStreamBasicDescription of the incoming packets using the FileStreamOpen call back function AudioFileStream_PropertyListenerProc and the destination format I populated manually.The code successfully creates the file and writes the packet to it but on playback what I hear is a white noise;
Here is my code
// when the recording button is pressed this function creates the file and setup the asbd
-(void)startRecording{
recording = true;
OSStatus status;
NSURL *baseUrl=[self applicationDocumentsDirectory];//returns the document direcotry of the app
NSURL *audioUrl = [NSURL URLWithString:#"Recorded.wav" relativeToURL:baseUrl];
//asbd setup for the destination file/wav file
AudioStreamBasicDescription dstFormat;
dstFormat.mSampleRate=44100.0;
dstFormat.mFormatID=kAudioFormatLinearPCM; dstFormat.mFormatFlags=kAudioFormatFlagsNativeEndian|kAudioFormatFlagIsSignedInteger|kAudioFormatFlagIsPacked;
dstFormat.mBytesPerPacket=4;
dstFormat.mBytesPerFrame=4;
dstFormat.mFramesPerPacket=1;
dstFormat.mChannelsPerFrame=2;
dstFormat.mBitsPerChannel=16;
dstFormat.mReserved=0;
//creating the file
status = ExtAudioFileCreateWithURL(CFBridgingRetain(audioUrl), kAudioFileWAVEType, &(dstFormat), NULL, kAudioFileFlags_EraseFile, &recordingFilRef);
// tell the EXtAudio File ApI what format we will be sending samples
//recordasbd is the asbd of incoming packets populated in AudioFileStream_PropertyListenerProc
status = ExtAudioFileSetProperty(recordingFilRef, kExtAudioFileProperty_ClientDataFormat, sizeof(recordasbd), &recordasbd);
}
// a handler called by packetproc call back function in AudiofileStreamOpen
- (void)handlePacketsProc:(const void *)inInputData numberBytes:(UInt32)inNumberBytes numberPackets:(UInt32)inNumberPackets packetDescriptions:(AudioStreamPacketDescription *)inPacketDescriptions {
if(recording){
// wrap the destination buffer in an audiobuffer list
convertedData.mNumberBuffers= 1;
convertedData.mBuffers[0].mNumberChannels = recordasbd.mChannelsPerFrame;
convertedData.mBuffers[0].mDataByteSize = inNumberBytes;
convertedData.mBuffers[0].mData = inInputData;
ExtAudioFileWrite(recordingFilRef,recordasbd.mFramesPerPacket * inNumberPackets, &convertedData);
}
}
My questions are:
Is my approach right can I write mp3 packets to wav file this way If so what am I missing ??
If my approach is wrong please tell me any other way you think is right.A nudge in the right direction is more than enough for me
I am so grateful for any help I have read every SO question I could get my hands on this topic, I also looked closely at apples Convertfile example but I could not figure out what I am missng
Thanks in advance for any help
Why not write the raw mp3 packets directly to a file? Without using ExtAudioFile at all.
They will form a valid mp3 file and will be much smaller than the equivalent wav file.

AudioUnitRender and ExtAudioFileWrite error -50 in Swift: Trying to convert MIDI to Audio File

I'm trying to convert a MIDI file to an Audio File (.m4a) in Swift.
Right now I'm using MIKMIDI as a tool to sequence and playback MIDI files, however it does not include the ability to save the playback into a file. MIKMID's creator outlines the process to do this here. In an attempt to capture and save the output to an audio file, I've followed this example to try and replace the MIKMIDI Graph's RemoteIO node with a GeneralIO node in Swift. When I try to save the output to a file using AudioUnitRender and ExtAudioFileWrite, they both return error -50 (kAudio_ParamError).
var channels = 2
var buffFrames = 512
var bufferList = AudioBufferList.allocate(maximumBuffers: 1)
for i in 0...bufferList.count-1{
var buffer = AudioBuffer()
buffer.mNumberChannels = 2
buffer.mDataByteSize = UInt32(buffFrames*sizeofValue(AudioUnitSampleType))
buffer.mData = calloc(buffFrames, sizeofValue(AudioUnitSampleType))
bufferList[i] = buffer
result = AudioUnitRender(generalIOAudioUnit, &flags, &inTimeStamp, busNum, UInt32(buffFrames), bufferList.unsafeMutablePointer)
inTimeStamp.mSampleTime += 1
result = ExtAudioFileWrite(extAudioFile, UInt32(buffFrames), bufferList.unsafeMutablePointer)
}
What is causing error -50, and how can I resolve it to render the MIDI (offline) to .m4a files?
UPDATE: I have resolved the ExtAudioFileWrite error -50 by changing mNumberChannels and channels to = 1. Now I get a one second audio file with noise. AudioUnitRender still returns error -50.
There are a couple of problems with your code:
your AudioBufferList doesn't agree with the client format, try
let bufferList = AudioBufferList.allocate(maximumBuffers: Int(clientFormat.mChannelsPerFrame))
you're replacing the wrong node from the AUGraph, and connecting the remaining node to itself, resulting in an infinite loop on AudioUnitRender.
But the main problem is that you are not implementing the solution that the author suggested. You wish that you could call AudioUnitRender with sample timestamps, faster than realtime, but the author said no, you'll have to manually convert sample time to hosttime and implement the better part of a midi player if you want that.
So you could do that (sounds hard), or file a feature request, or maybe record to file in realtime as you listen to the music by adding a render notification to the graph's remote IO audio unit with AudioUnitAddRenderNotify and writing the samples during the kAudioUnitRenderAction_PostRender phase.

Stream video while downloading iOS

I am using iOS 7 and I have a .mp4 video that I need to download in my app. The video is large (~ 1 GB) which is why it is not included as part of the app. I want the user to be able to start watching the video as soon as is starts downloading. I also want the video to be able to be cached on the iOS device so the user doesn't need to download it again later. Both the normal methods of playing videos (progressive download and live streaming) don't seem to let you cache the video, so I have made my own web service that chunks up my video file and streams the bytes down to the client. I start the streaming HTTP call using NSURLConnection:
self.request = [[NSMutableURLRequest alloc] initWithURL:self.url];
[self.request setTimeoutInterval:10]; // Expect data at least every 10 seconds
[self.request setHTTPMethod:#"GET"];
self.connection = [[NSURLConnection alloc] initWithRequest:self.request delegate:self startImmediately:YES];
When I receive a data chunk, I append it to the end of the local copy of the file:
- (void)connection:(NSURLConnection *)connection didReceiveData:(NSData *)data
{
NSFileHandle *handle = [NSFileHandle fileHandleForWritingAtPath:[self videoFilePath]];
[handle truncateFileAtOffset:[handle seekToEndOfFile]];
[handle writeData:data];
}
If I let the device run, the file is downloaded successfully and I can play it using MPMoviePlayerViewController:
NSURL *url=[NSURL fileURLWithPath:self.videoFilePath];
MPMoviePlayerViewController *controller = [[MPMoviePlayerViewController alloc] initWithContentURL:url];
controller.moviePlayer.scalingMode = MPMovieScalingModeAspectFit;
[self presentMoviePlayerViewControllerAnimated:controller];
However, if I start the player before the file is completely downloaded, the video starts playing just fine. It even has the correct video length displayed at the top scrubber bar. But when the user gets to the position in the video that I had completed downloading before the video started, the video just hangs. If I close and reopen the MPMoviePlayerViewController, then the video plays until it gets to whatever location I was then at when I launched the MPMoviePlayerViewController again. If I wait until the entire video is downloaded, then the video plays without a problem.
I am not getting any events fired, or error messages printed to the console when this happens (MPMoviePlayerPlaybackStateDidChangeNotification and MPMoviePlayerPlaybackDidFinishNotification are never sent after the video starts). It seems like there is something else that is telling the controller what the length of the video is other than what the scrubber is using...
Does anyone know what could be causing this issue? I am not bound to using MPMoviePlayerViewController, so if a different video playback method would work in this situation I am all for it.
Related Unresolved Questions:
AVPlayer and Progressive Video Downloads with AVURLAssets
Progressive Video Download on iOS
How to play an in downloading progress video file in IOS
UPDATE 1
I have found that the video stall is indeed because of the file size when the video starts playing. I can get around this issue by creating a zero-ed out file before I start the download and over overwrite it as I go. Since I have control over the video streaming server, I added a custom header so I know the size of the file being streamed (default file size header for a streaming file is -1). I am creating the file in my didReceiveResponse method as follows:
- (void)connection:(NSURLConnection *)connection didReceiveResponse:(NSURLResponse *)response
{
// Retrieve the size of the file being streamed.
NSHTTPURLResponse *httpResponse = (NSHTTPURLResponse *)response;
NSDictionary *headers = httpResponse.allHeaderFields;
NSNumberFormatter * formatter = [[NSNumberFormatter alloc] init];
[formatter setNumberStyle:NSNumberFormatterDecimalStyle];
self.streamingFileSize = [formatter numberFromString:[headers objectForKey:#"StreamingFileSize"]];
// Check if we need to initialize the download file
if (![[NSFileManager defaultManager] fileExistsAtPath:self.path])
{
// Create the file being downloaded
[[NSData data] writeToFile:self.path atomically:YES];
// Allocate the size of the file we are going to download.
const char *cString = [self.path cStringUsingEncoding:NSASCIIStringEncoding];
int success = truncate(cString, self.streamingFileSize.longLongValue);
if (success != 0)
{
/* TODO: handle errors here. Probably not enough space... See 'man truncate' */
}
}
}
This works great, except that truncate causes the app to hang for about 10 seconds while it creates the ~1GB file on disk (on the simulator it is instant, only a real device has this problem). This is where I am stuck now - does anyone know of a way to allocate a file more efficiently, or a different way to get the video player to recognize the size of the file without needing to actually allocate it? I know some filesystems support "file size" and "size on disk" as two different properties... not sure if iOS has something like that?
I figured out how to do this, and it is much simpler than my original idea.
First, since my video is in .mp4, the MPMoviePlayerViewController or AVPlayer class can play it directly from a web server - I don't need to implement anything special and they can still seek to any point in the video. This must be part of how the .mp4 encoding works with the movie players. So, I just have the raw file available on the server - no special headers required.
Next, when the user decides to play the video I immediately start playing the video from the server URL:
NSURL *url=[NSURL fileURLWithPath:serverVidelFileURLString];
controller = [[MPMoviePlayerViewController alloc] initWithContentURL:url];
controller.moviePlayer.scalingMode = MPMovieScalingModeAspectFit;
[self presentMoviePlayerViewControllerAnimated:controller];
This makes it so the user can watch the video and seek to any location they want. Then, I start downloading the file manually using NSURLConnection like I had been doing above, except now I am not streaming the file, I just download it directly. This way I don't need the custom header since the file size is included in the HTTP response.
When my background download completes, I switch the playing item from the server URL to the local file. This is important for network performance because the movie players only download a few seconds ahead of what the user is watching. Being able to switch to the local file as soon as possible is key to avoid downloading too much duplicate data:
NSTimeInterval currentPlaybackTime = videoController.moviePlayer.currentPlaybackTime;
[controller.moviePlayer setContentURL:url];
[controller.moviePlayer setCurrentPlaybackTime:currentPlaybackTime];
[controller.moviePlayer play];
This method does have the user downloading two video files at the same time initially, but initial testing on the network speeds my users will be using shows it only increases the download time by a few seconds. Works for me!
You gotta create an internal webserver that acts like a proxy! Then set your player to play the movie from the localhost.
When using HTTP protocol to play a video with MPMoviePlayerViewController, the first thing the player does is to ask for the byte-range 0-1 (first 2 bytes) just to obtain the file length. Then, the player asks for "chunks" of the video using the "byte-range" HTTP command (the purpose is to save some battery).
What you have to do is to implement this internal server that delivers the video to the player, but your "proxy" must consider the length of your video as the full length of the file, even if the actual file hasn't been completely downloaded from the internet.
Then you you set your player to play a movie from " http:// localhost : someport "
I've done this before... it works perfectly!
Good luck!
I can only assume that the MPMoviePlayerViewController caches the file length of the file when you started it.
The way to fix (just) this issue is to first determine how large the file is. Then create a file of that length. Keeping an offset pointer, as the file downloads, you can overwrite the "null" values in the file with the real data.
So you get to a specific point in the download, start the MPMoviePlayerViewController, and let it run. I'd also suggest you use the "F_NOCACHE" flag (with fcntl()) so you bypass the file block cache (which means you will lower your memory footprint).
The downside to this architecture is that if you get stalled, and the movie player gets ahead of you, well, the user is going to have a pretty bad experience. Not sure if there is any way for you to monitor and take preemptive action.
EDIT: its quite possible that the video is not read sequentially, but certain information requires the player to essentially look ahead for something. If so, then this is doomed to fail. The only other possible solution is to use some software tool to sequentially order the file (I'm no video expert so cannot comment from experience on any of the above).
To test this out, you can construct a "damaged" video of varying lengths, and test that to see what works and what does not. For instance, suppose you have a 100Meg file. Write a little utility program, and over write the last 50Megs of data with zeros. Now play this video. Its should fail 1/2 through. If it fails right away, well, you now know that its seeking in the file.
If non sequential, its possible that its looking at the last 1000 bytes or so, in which case if you don't overwrite that things work as you want. If you get lucky and this is the case, you would eventually download the last 1000 bytes, then then start from the front of the file.
It really gets down to finding some way before introducing real networking into the picture, to play a partial file. You will surely find it easier to artificially introduce the networking conditions without really doing it real time.

CMSampleBufferRef pool to write H.264 AVCC stream

I'm using AVAssetWriter/AVAssetWriterInput to write H.264 raw data to an MP4 file. As I'm receiving the data from a remote server, I use the following CoreMedia APIs to get a sample buffer (CMSampleBufferRef) containing the H.264 data in AVCC format that is in turned appended to an MP4 file by sending to an AVAssetWriterInput the message (BOOL)appendSampleBuffer:(CMSampleBufferRef)sampleBuffer:
CMBlockBufferCreateWithMemoryBlock to create a memory block
CMBlockBufferReplaceDataBytes to write the H.264 in AVCC format to the memory block
CMSampleBufferCreate to create a sample buffer with the memory block and a format descriptor containing the H.264 "extradata"
Everything works as expected, the only problem with this approach is that I'm periodically calling the above APIs and what I would really like is instead to be able to reuse the resources allocated - in particular CMSampleBufferRef and CMBlockBufferRef. Basically, I would like to have a pool of CMSampleBuffer's and be able to update its memory content and format descriptor as I'm receiving new H.264 data from the remote server.
I know that exists AVAssetWriterInputPixelBufferAdaptorthat gives access to a CVPixelBufferPool but, I can't use it in my case because as far as I know, to properly instantiate a pixel buffer adaptor, at minimum I need to be able to pass the video frame dimensions which I would' know until I parse the stream. Further, I don't know how to write the H.264 "extradata" with a CVPixelBuffer. So, I'm thinking that I need to stick with CMSampleBuffer. Unfortunately, it seems that CoreMedia APIs don't offer the possibility to update the memory block nor the format descriptor of a sample buffer once created (as far as I can tell, I only have access to immutable references of those objects). Thus, the best I can do so far is to reuse the memory block CMBlockBufferRef but I'm still recreating the sample buffer. My code is below. Hopefully someone here will have some ideas on how to implement a pool of CMSampleBuffer's or perhaps a more efficient way to write H.264 AVCC stream to MP4?
- (CMSampleBufferRef)sampleBufferWithData:(NSData*)data formatDescriptor:(CMFormatDescriptionRef)formatDescription
{
OSStatus result;
CMSampleBufferRef sampleBuffer = NULL;
// _blockBuffer is a CMBlockBufferRef instance variable
if (!_blockBuffer)
{
size_t blockLength = MAX_LENGTH;
result = CMBlockBufferCreateWithMemoryBlock(kCFAllocatorDefault,
NULL,
blockLength,
kCFAllocatorDefault,
NULL,
0,
blockLength,
kCMBlockBufferAssureMemoryNowFlag,
&_blockBuffer);
// check error
}
result = CMBlockBufferReplaceDataBytes([data bytes], _blockBuffer, 0, [data length]);
// check error
const size_t sampleSizes = [data length];
CMSampleTimingInfo timing = [self sampleTimingInfo];
result = CMSampleBufferCreate(kCFAllocatorDefault,
_blockBuffer,
YES,
NULL,
NULL,
formatDescription,
1,
1,
&timing,
1,
&sampleSizes,
&sampleBuffer);
// check error
return sampleBuffer;
}
If you are receiving raw H.264 data, then there is not much do to and no need to deal with CoreMedia at all.
Buffer all VCL NAL units until you get SPS/PPS NAL units. Create the extradata from them, then just append all buffered and new VCL NAL units to the file. In case you are received the NAL units in Annex B format you need to convert them to AVCC format (basically replacing the start code with a length code)
You only need to work with 'CMSampleBuffer' if you want to decode uncompressed pictures or if you want to decode compressed pictures. As you are already working with a raw H.264 stream and just want to write it into an MP4 file, just do so. No need to touch CoreMedia at all here.
Regarding CoreMedia: you wrap your video information in a CMBlockBuffer. This buffers together with a CMVideoFormatDescriptor (generated from SPS/PPS) plus CMTime make up a CMSampleBuffer. And multiple CMSampleBuffers make up a 'CMSampleBufferPool'.
'CVPixelBuffer' and 'CVPixelBufferPool' are not involved. These are either the input or output of a 'VTCompressionSession' or "VTDecompressionSession' when dealing with encoding/decoding h.264 video.
As said in your case, no need to touch any of the core framworks at all as you are just creating a file.
An overview about Annex B and AVCC stream format can be found here: Possible Locations for Sequence/Picture Parameter Set(s) for H.264 Stream

Why I am receiving only a few audio samples per second when using AVAssetReader on iOS?

I'm coding something that:
record video+audio with the built-in camera and mic (AVCaptureSession),
do some stuff with the video and audio samplebuffer in realtime,
save the result into a local .mp4 file using AVAssetWritter,
then (later) read the file (video+audio) using AVAssetReader,
do some other stuff with the samplebuffer (for now I do nothing),
and write the result into a final video file using AVAssetWriter.
Everything works well but I have an issue with the audio format:
When I capture the audio samples from the capture session, I can log about 44 samples/sec, which seams to be normal.
When I read the .mp4 file, I only log about 3-5 audio samples/sec!
But the 2 files look and sound exactly the same (in QuickTime).
I didn't set any audio settings for the Capture Session (as Apple doesn't allow it).
I configured the outputSettings of the 2 audio AVAssetWriterInput as follow:
NSDictionary *settings = #{
AVFormatIDKey:#(kAudioFormatLinearPCM),
AVNumberOfChannelsKey:#(2),
AVSampleRateKey:#(44100.),
AVLinearPCMBitDepthKey:#(16),
AVLinearPCMIsNonInterleaved:#(NO),
AVLinearPCMIsFloatKey:#(NO),
AVLinearPCMIsBigEndianKey:#(NO)
};
I pass nil to the outputSettings of the audio AVAssetReaderTrackOutput in order to receive samples as stored in the track (according to the doc).
So, the sample rate should be 44100Hz from the CaptureSession to the final file. Why I am reading only a few audio samples? And why is it working anyway? I have the intuition that it will not work well when I'll have to work with the samples (I need to update their timestamps for example).
I tried several other settings (such as kAudioFormatMPEG4AAC), but AVAssetReader can't read compressed audio formats.
Thanks for your help :)

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