Context: iOS5 AUSampler AudioUnit
I've been digging around trying to determine is there is a programmatic way to determine the number of presets in a DLS or sf2 file. I was hoping it would be available either through 'AudioUnitGetProperty' or 'AudioUnitGetParameter' for an AUSampler. Then of course I want to be able to switch presets on the fly. The Docs don't indicate if this is possible or not.
I'm using the standard code for loading DLS/sf2 per TechNote TN2283. The problem is that with lots of sf2 files it is a trial and error process to find out what the presets are.
-(OSStatus) loadFromDLSOrSoundFont: (NSURL *)bankURL withPatch: (int)presetNumber
OSStatus result = noErr;
// fill out a bank preset data structure
AUSamplerBankPresetData bpdata;
bpdata.bankURL = (CFURLRef) bankURL;
bpdata.bankMSB = kAUSampler_DefaultMelodicBankMSB;
bpdata.bankLSB = kAUSampler_DefaultBankLSB;
bpdata.presetID = (UInt8) presetNumber;
// set the kAUSamplerProperty_LoadPresetFromBank property
result = AudioUnitSetProperty(self.mySamplerUnit,
kAUSamplerProperty_LoadPresetFromBank,
kAudioUnitScope_Global,
0,
&bpdata,
sizeof(bpdata));
// check for errors
NSCAssert (result == noErr,
#"Unable to set the preset property on the Sampler. Error code:%d '%.4s'",
(int) result,
(const char *)&result);
return result;
}
OK - had an answer from an Apple Core Audio engineer:
"There is no API to retrieve the number of presets. The Sampler AU only loads a single instrument at a time from any SF2 or DLS bank, so it does not "digest" the entire bank file (and so has no knowledge of its complete contents)."
Related
I try to make a sip video call using Pjsip/Pjsua on my raspberry pi 3.
Before coding, I'm using the main sample app to test different options. Everything seems to work (registering, audio calling,..) but when I try to start a video call, the programs stops with the following message :
pjsua-armv7l-unknown-linux-gnueabihf: ../src/pjmedia-videodev/v4l2_dev.c:737: vid4lin_stream_get_frame_mmap: Assertion `!"frame buffer is too small for v4l2"' failed.
I've searched a lot, including the source code :
/* get frame from mmap */
static pj_status_t vid4lin_stream_get_frame_mmap(vid4lin_stream *stream, pjmedia_frame *frame)
{
struct v4l2_buffer buf;
pj_time_val time;
pj_status_t status = PJ_SUCCESS;
pj_bzero(&buf, sizeof(buf));
buf.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
buf.memory = V4L2_MEMORY_MMAP;
status = xioctl(stream->fd, VIDIOC_DQBUF, &buf);
if (status != PJ_SUCCESS)
return status;
if (frame->size < buf.bytesused) {
/* supplied buffer is too small */
pj_assert(!"frame buffer is too small for v4l2");
status = PJ_ETOOSMALL;
goto on_return;
}
So I understand that the pjmedia_frame has a "size" inferior to the v4l2 buffer, resulting to my failure.
My question is simple : how can i change this setting ?
I tried evetything in the sample app : changing resolution, bitrate, fps,..
I found some ressources saying to change the h264 profile level.. ok, but where do i set it ? Is it within the v4l2 manager ? or directly in the app ? How can i do it ?
I played with different options in v4l2 to reduce the bitrate/resolution in order to have a small buffer, but still getting the same error.
At this point I'm completely clueless.
For info, I compiled PJsip using openh264 (no libx264) as suggested by PjSip.
Thanks for your help/ideas ;)
According to your question about profile level, you can try with:
const pj_str_t codec_id = {"H264", 4};
pjmedia_vid_codec_param param;
pj_status_t status;
status = pjsua_vid_codec_get_param(&codec_id, ¶m);
param.dec_fmtp.param[0].name = pj_str("profile-level-id");
param.dec_fmtp.param[0].val = pj_str("42e01f");
status = pjsua_vid_codec_set_param(&codec_id, ¶m);
do this anywhere after pjsua_start(). Last two characters in val property are profile level. Description of levels can be found here (link). More information about h264 profile here (link).
I'm not an expert of v4l2, but have little experience with encoding video on rpi3, and I suggest you to use FFmpeg instead of pure openh264, beacuse of support of hardware acceleration (link).
Good luck!
I noticed in the iOS documentation for AVAssetWriterInput you can pass nil for the outputSettings dictionary to specify that the input data should not be re-encoded.
The settings used for encoding the media appended to the output. Pass nil to specify that appended samples should not be re-encoded.
I want to take advantage of this feature to pass in a stream of raw H.264 NALs, but I am having trouble adapting my raw byte streams into a CMSampleBuffer that I can pass into AVAssetWriterInput's appendSampleBuffer method. My stream of NALs contains only SPS/PPS/IDR/P NALs (1, 5, 7, 8). I haven't been able to find documentation or a conclusive answer on how to use pre-encoded H264 data with AVAssetWriter. The resulting video file is not able to be played.
How can I properly package the NAL units into CMSampleBuffers? Do I need to use a start code prefix? A length prefix? Do I need to ensure I only put one NAL per CMSampleBuffer? My end goal is to create an MP4 or MOV container with H264/AAC.
Here's the code I've been playing with:
-(void)addH264NAL:(NSData *)nal
{
dispatch_async(recordingQueue, ^{
//Adapting the raw NAL into a CMSampleBuffer
CMSampleBufferRef sampleBuffer = NULL;
CMBlockBufferRef blockBuffer = NULL;
CMFormatDescriptionRef formatDescription = NULL;
CMItemCount numberOfSampleTimeEntries = 1;
CMItemCount numberOfSamples = 1;
CMVideoFormatDescriptionCreate(kCFAllocatorDefault, kCMVideoCodecType_H264, 480, 360, nil, &formatDescription);
OSStatus result = CMBlockBufferCreateWithMemoryBlock(kCFAllocatorDefault, NULL, [nal length], kCFAllocatorDefault, NULL, 0, [nal length], kCMBlockBufferAssureMemoryNowFlag, &blockBuffer);
if(result != noErr)
{
NSLog(#"Error creating CMBlockBuffer");
return;
}
result = CMBlockBufferReplaceDataBytes([nal bytes], blockBuffer, 0, [nal length]);
if(result != noErr)
{
NSLog(#"Error filling CMBlockBuffer");
return;
}
const size_t sampleSizes = [nal length];
CMSampleTimingInfo timing = { 0 };
result = CMSampleBufferCreate(kCFAllocatorDefault, blockBuffer, YES, NULL, NULL, formatDescription, numberOfSamples, numberOfSampleTimeEntries, &timing, 1, &sampleSizes, &sampleBuffer);
if(result != noErr)
{
NSLog(#"Error creating CMSampleBuffer");
}
[self writeSampleBuffer:sampleBuffer ofType:AVMediaTypeVideo];
});
}
Note that I'm calling CMSampleBufferSetOutputPresentationTimeStamp on the sample buffer inside of the writeSampleBuffer method with what I think is a valid time before I'm actually trying to append it.
Any help is appreciated.
I managed to get video playback working in VLC but not QuickTime. I used code similar to what I posted above to get H.264 NALs into CMSampleBuffers.
I had two main issues:
I was not setting CMSampleTimingInfo correctly (as my comment above states).
I was not packing the raw NAL data correctly (not sure where this is documented, if anywhere).
To solve #1, I set timing.duration = CMTimeMake(1, fps); where fps is the expected frame rate. I then set timing.decodeTimeStamp = kCMTimeInvalid; to mean that the samples will be given in decoding order. Lastly, I set timing.presentationTimeStamp by calculating the absolute time, which I also used with startSessionAtSourceTime.
To solve #2, through trial and error I found that giving my NAL units in the following form worked:
[7 8 5] [1] [1] [1]..... [7 8 5] [1] [1] [1]..... (repeating)
Where each NAL unit is prefixed by a 32-bit start code equaling 0x00000001.
Presumably for the same reason it's not playing in QuickTime, I'm still having trouble moving the resulting .mov file to the photo album (the ALAssetLibrary method videoAtPathIsCompatibleWithSavedPhotosAlbum is failing stating that the "Movie could not be played." Hopefully someone with an idea about what's going on can comment. Thanks!
I'm working with the MusicPlayer API. I understand that when you load in a .mid as a sequence, the API creates a default AUGraph for you that includes an AUSampler. This AUSampler uses a simple sine-wave based instrument to synthesize the notes in the .mid
My question is, how does one change the default instrument in the AUSampler? I understand that you can use SoundFont2 files (.sf2) and add them using the AudioUnitSetProperty method. But, how does one access this default AUGraph? Do you have to open the graph before you can edit the AudioUnit or is opening a graph only for editing connections between nodes?
Thanks :)
I've written a tutorial on this but here but here's an outline of the process:
Function to load a Sound Font file (taken from the Apple documentation):
-(OSStatus) loadFromDLSOrSoundFont: (NSURL *)bankURL withPatch: (int)presetNumber {
OSStatus result = noErr;
// fill out a bank preset data structure
AUSamplerBankPresetData bpdata;
bpdata.bankURL = (__bridge CFURLRef) bankURL;
bpdata.bankMSB = kAUSampler_DefaultMelodicBankMSB;
bpdata.bankLSB = kAUSampler_DefaultBankLSB;
bpdata.presetID = (UInt8) presetNumber;
// set the kAUSamplerProperty_LoadPresetFromBank property
result = AudioUnitSetProperty([pointer to your AUSampler node here],
kAUSamplerProperty_LoadPresetFromBank,
kAudioUnitScope_Global,
0,
&bpdata,
sizeof(bpdata));
// check for errors
NSCAssert (result == noErr,
#"Unable to set the preset property on the Sampler. Error code:%d '%.4s'",
(int) result,
(const char *)&result);
return result; }
Then you need to load the Sound Font from your Resources folder:
NSURL *presetURL = [[NSURL alloc] initFileURLWithPath:[[NSBundle mainBundle] pathForResource:#"Name of sound font" ofType:#"sf2"]];
// Initialise the sound font
[self loadFromDLSOrSoundFont: (NSURL *)presetURL withPatch: (int)10];
Hope this helps!
You might take a look at the Audiograph example. It doesn't use soundFonts but should give you an idea of how to set up a graph.
When I use the MusicPlayer I always generate the midi note data from code/GUI and create the AUGraph (with a mixer) from scratch. There are ways to derive/extract the default generated AUGraph & AUSampler resulting from loading a midi file (example code below) but I never had success setting a new soundFont this way. On the other hand, creating the AUGraph from scratch and then loading an .sf2 file works great.
AUGraph graph;
result = MusicSequenceGetAUGraph (sequence, &graph);
MusicTrack firstTrack;
result = MusicSequenceGetIndTrack (sequence, 0, &firstTrack);
AUNode myNode;
result = MusicTrackGetDestNode(firstTrack,&myNode);
AudioUnit mySamplerUnit;
result = AUGraphNodeInfo(graph, myNode, 0, &mySamplerUnit);
Using CoreAudio, I am able to get the sampleRate (frames per second) and the file size, but in order to get the "total" time of the song, I need to know the Real file size of that compressed mp3.
AudioStreamBasicDescription asbd;
UInt32 asbdSize = sizeof(asbd);
// get the stream format.
err = AudioFileStreamGetProperty(inAudioFileStream, kAudioFileStreamProperty_DataFormat, &asbdSize, &asbd);
if (err)
{
[self failWithErrorCode:AS_FILE_STREAM_GET_PROPERTY_FAILED];
return;
}
sampleRate = asbd.mSampleRate;
Is there any way I can know the real size of the song using Objective-C?
Thanks in advance.
See the answer to this question
There's a property you can ask in AudioFileGetProperty called kAudioFilePropertyEstimatedDuration that should do the trick.
I find Apple's documentation quite limited on AudioFileStreamSeek and I cannot find any examples of actual usage anywhere. I have a working streaming audio player, but I just can't seem to get AudioFileStreamSeek to work as advertised...
Any help tips or a little example would be greatly appreciated!
I am told this works:
AudioQueueStop(audioQueue, true);
UInt32 flags = 0;
err = AudioFileStreamParseBytes(audioFileStream, length, bytes,
kAudioFileStreamParseFlag_Discontinuity);
OSStatus status = AudioFileStreamSeek(audioFileStream, framePacket.mPacket,
¤tOffset, &flags);
NSLog(#"Setting next byte offset to: %qi, flags: %d", (long long)currentOffset, flags);
// then read data from the new offset set by AudioFileStreamSeek
[fileHandle seekToFileOffset:currentOffset];
NSData* data = "" readDataOfLength:4096];
flags = kAudioFileStreamParseFlag_Discontinuity;
status = AudioFileStreamParseBytes( stream, [data length], [data bytes], flags);
if (status != noErr)
{
NSLog(#"Error parsing bytes: %d", status);
}
Unless I'm mistaken, this is only available in the 3.0 SDK, and therefore under NDA. Maybe you should take this to the Apple Beta forums?
I stand corrected. AudioFileStreamSeek doesn't show up if you do a search in the online 2.2.1 documentation. You have to manually dig into the docs to find it.
Don't forget to add the data offset (kAudioFileStreamProperty_DataOffset) to the byte offset returned by AudioFileStreamSeek. The return value is an offset into the audio data and ignores the data offset.
It's also a good idea to stop and then re-start the AudioQueue before/after seeking.
Matt Gallagher uses AudioFileStreamSeek in his example "Streaming and playing an MP3 stream".
Look at Matt's code AudioStreamer.m:
SInt64 seekPacket = floor(newSeekTime / packetDuration);
err = AudioFileStreamSeek(audioFileStream, seekPacket, &packetAlignedByteOffset, &ioFlags);