I'm using Apple Video Toolbox framework to compress raw frames captured by the device camera.
My callback is being called with a CMSampleBufferRef object that contains CMBlockBuffer.
The CMBlockBuffer object contain the H264 elementary stream but I didn't find any way to get a pointer to the elementary stream.
When I printed into the console the CMSampleBufferRef object I got:
(lldb) po blockBufferRef
CMBlockBuffer 0x1701193e0 totalDataLength: 4264 retainCount: 1 allocator: 0x1957c2c80 subBlockCapacity: 2
[0] 4264 bytes # offset 128 Buffer Reference:
CMBlockBuffer 0x170119350 totalDataLength: 4632 retainCount: 1 allocator: 0x1957c2c80 subBlockCapacity: 2
[0] 4632 bytes # offset 0 Memory Block 0x10295c000, 4632 bytes (custom V=0 A=0x0 F=0x18498bb44 R=0x0)
It seems that the CMBlockBuffer object that I managed to get pointer to is contaning another CMBlockBuferRef (4632 bytes) which is not accessible.
Can anyone post how to access the H264 elemantry stream?
Thank you!
I've been struggling with this myself for quite some time now, and have finally figured everything out.
The function CMBlockBufferGetDataPointer gives you access to all the data you need, but there are a few not very obvious things you need to do to convert it to an elementary stream.
AVCC vs Annex B format
The data in the CMBlockBuffer is stored in AVCC format, while elementary streams are typically following the Annex B specification (here is an excellent overview of the two formats). In the AVCC format, the 4 first bytes contains the length of the NAL unit (another word for H264 packet). You need to replace this header with the 4 byte start code: 0x00 0x00 0x00 0x01, which functions as a separator between NAL units in an Annex B elementary stream (the 3 byte version 0x00 0x00 0x01 works fine too).
Multiple NAL units in a single CMBlockBuffer
The next not very obvious thing is that a single CMBlockBuffer will sometimes contain multiple NAL units. Apple seems to add an additional NAL unit (SEI) containing metadata to every I-Frame NAL unit (also called IDR). This is probably why you are seeing multiple buffers in a single CMBlockBuffer object. However, the CMBlockBufferGetDataPointer function gives you a single pointer with access to all the data. That being said, the presence of multiple NAL units complicates the conversion of the AVCC headers. Now you actually have to read the length value contained in the AVCC header to find the next NAL unit, and continue converting headers until you have reached the end of the buffer.
Big-Endian vs Little-Endian
The next not very obvious thing is that the AVCC header is stored in Big-Endian format, and iOS is Little-Endian natively. So when you are reading the length value contained in an AVCC header pass it to the CFSwapInt32BigToHost function first.
SPS and PPS NAL units
The final not very obvious thing is that the data inside the CMBlockBuffer does not contain the parameter NAL units SPS and PPS, which contains configuration parameters for the decoder such as profile, level, resolution, frame rate. These are stored as metadata in the sample buffer's format description and can be accessed via the function CMVideoFormatDescriptionGetH264ParameterSetAtIndex. Note that you have to add the start codes to these NAL units before sending. The SPS and PPS NAL units does not have to be sent with every new frame. A decoder only needs to read them once, but it is common to resend them periodically, for example before every new I-frame NAL unit.
Code Example
Below is a code example taking all of these things into account.
static void videoFrameFinishedEncoding(void *outputCallbackRefCon,
void *sourceFrameRefCon,
OSStatus status,
VTEncodeInfoFlags infoFlags,
CMSampleBufferRef sampleBuffer) {
// Check if there were any errors encoding
if (status != noErr) {
NSLog(#"Error encoding video, err=%lld", (int64_t)status);
return;
}
// In this example we will use a NSMutableData object to store the
// elementary stream.
NSMutableData *elementaryStream = [NSMutableData data];
// Find out if the sample buffer contains an I-Frame.
// If so we will write the SPS and PPS NAL units to the elementary stream.
BOOL isIFrame = NO;
CFArrayRef attachmentsArray = CMSampleBufferGetSampleAttachmentsArray(sampleBuffer, 0);
if (CFArrayGetCount(attachmentsArray)) {
CFBooleanRef notSync;
CFDictionaryRef dict = CFArrayGetValueAtIndex(attachmentsArray, 0);
BOOL keyExists = CFDictionaryGetValueIfPresent(dict,
kCMSampleAttachmentKey_NotSync,
(const void **)¬Sync);
// An I-Frame is a sync frame
isIFrame = !keyExists || !CFBooleanGetValue(notSync);
}
// This is the start code that we will write to
// the elementary stream before every NAL unit
static const size_t startCodeLength = 4;
static const uint8_t startCode[] = {0x00, 0x00, 0x00, 0x01};
// Write the SPS and PPS NAL units to the elementary stream before every I-Frame
if (isIFrame) {
CMFormatDescriptionRef description = CMSampleBufferGetFormatDescription(sampleBuffer);
// Find out how many parameter sets there are
size_t numberOfParameterSets;
CMVideoFormatDescriptionGetH264ParameterSetAtIndex(description,
0, NULL, NULL,
&numberOfParameterSets,
NULL);
// Write each parameter set to the elementary stream
for (int i = 0; i < numberOfParameterSets; i++) {
const uint8_t *parameterSetPointer;
size_t parameterSetLength;
CMVideoFormatDescriptionGetH264ParameterSetAtIndex(description,
i,
¶meterSetPointer,
¶meterSetLength,
NULL, NULL);
// Write the parameter set to the elementary stream
[elementaryStream appendBytes:startCode length:startCodeLength];
[elementaryStream appendBytes:parameterSetPointer length:parameterSetLength];
}
}
// Get a pointer to the raw AVCC NAL unit data in the sample buffer
size_t blockBufferLength;
uint8_t *bufferDataPointer = NULL;
CMBlockBufferGetDataPointer(CMSampleBufferGetDataBuffer(sampleBuffer),
0,
NULL,
&blockBufferLength,
(char **)&bufferDataPointer);
// Loop through all the NAL units in the block buffer
// and write them to the elementary stream with
// start codes instead of AVCC length headers
size_t bufferOffset = 0;
static const int AVCCHeaderLength = 4;
while (bufferOffset < blockBufferLength - AVCCHeaderLength) {
// Read the NAL unit length
uint32_t NALUnitLength = 0;
memcpy(&NALUnitLength, bufferDataPointer + bufferOffset, AVCCHeaderLength);
// Convert the length value from Big-endian to Little-endian
NALUnitLength = CFSwapInt32BigToHost(NALUnitLength);
// Write start code to the elementary stream
[elementaryStream appendBytes:startCode length:startCodeLength];
// Write the NAL unit without the AVCC length header to the elementary stream
[elementaryStream appendBytes:bufferDataPointer + bufferOffset + AVCCHeaderLength
length:NALUnitLength];
// Move to the next NAL unit in the block buffer
bufferOffset += AVCCHeaderLength + NALUnitLength;
}
}
Thanks Anton for an excellent answer! Am putting a naive Swift-port of your solution for people interested in using the concepts discussed here straight in their Swift-based projects.
public func didEncodeFrame(frame: CMSampleBuffer)
{
print ("Received encoded frame in delegate...")
//----AVCC to Elem stream-----//
var elementaryStream = NSMutableData()
//1. check if CMBuffer had I-frame
var isIFrame:Bool = false
let attachmentsArray:CFArray = CMSampleBufferGetSampleAttachmentsArray(frame, false)!
//check how many attachments
if ( CFArrayGetCount(attachmentsArray) > 0 ) {
let dict = CFArrayGetValueAtIndex(attachmentsArray, 0)
let dictRef:CFDictionaryRef = unsafeBitCast(dict, CFDictionaryRef.self)
//get value
let value = CFDictionaryGetValue(dictRef, unsafeBitCast(kCMSampleAttachmentKey_NotSync, UnsafePointer<Void>.self))
if ( value != nil ){
print ("IFrame found...")
isIFrame = true
}
}
//2. define the start code
let nStartCodeLength:size_t = 4
let nStartCode:[UInt8] = [0x00, 0x00, 0x00, 0x01]
//3. write the SPS and PPS before I-frame
if ( isIFrame == true ){
let description:CMFormatDescriptionRef = CMSampleBufferGetFormatDescription(frame)!
//how many params
var numParams:size_t = 0
CMVideoFormatDescriptionGetH264ParameterSetAtIndex(description, 0, nil, nil, &numParams, nil)
//write each param-set to elementary stream
print("Write param to elementaryStream ", numParams)
for i in 0..<numParams {
var parameterSetPointer:UnsafePointer<UInt8> = nil
var parameterSetLength:size_t = 0
CMVideoFormatDescriptionGetH264ParameterSetAtIndex(description, i, ¶meterSetPointer, ¶meterSetLength, nil, nil)
elementaryStream.appendBytes(nStartCode, length: nStartCodeLength)
elementaryStream.appendBytes(parameterSetPointer, length: unsafeBitCast(parameterSetLength, Int.self))
}
}
//4. Get a pointer to the raw AVCC NAL unit data in the sample buffer
var blockBufferLength:size_t = 0
var bufferDataPointer: UnsafeMutablePointer<Int8> = nil
CMBlockBufferGetDataPointer(CMSampleBufferGetDataBuffer(frame)!, 0, nil, &blockBufferLength, &bufferDataPointer)
print ("Block length = ", blockBufferLength)
//5. Loop through all the NAL units in the block buffer
var bufferOffset:size_t = 0
let AVCCHeaderLength:Int = 4
while (bufferOffset < (blockBufferLength - AVCCHeaderLength) ) {
// Read the NAL unit length
var NALUnitLength:UInt32 = 0
memcpy(&NALUnitLength, bufferDataPointer + bufferOffset, AVCCHeaderLength)
//Big-Endian to Little-Endian
NALUnitLength = CFSwapInt32(NALUnitLength)
if ( NALUnitLength > 0 ){
print ( "NALUnitLen = ", NALUnitLength)
// Write start code to the elementary stream
elementaryStream.appendBytes(nStartCode, length: nStartCodeLength)
// Write the NAL unit without the AVCC length header to the elementary stream
elementaryStream.appendBytes(bufferDataPointer + bufferOffset + AVCCHeaderLength, length: Int(NALUnitLength))
// Move to the next NAL unit in the block buffer
bufferOffset += AVCCHeaderLength + size_t(NALUnitLength);
print("Moving to next NALU...")
}
}
print("Read completed...")
}
Related
I'm struggling with adding a number to a AVFame/AVPacket before I encode it and retrieve the number back when I decode it. My original question was here but I wasn't able to get it work with the AVFrame metadata or side_data. I have looked at various posts in stackflow like this or on the internet ( using AVDictionary, AVDictionaryEntry ... ) but nothing so far. I have managed to store a number in the metadata object of the frame but it is not there when I decode the package. Anyone have an idea what I need to do? Are my coding settings not correct and is therefor my custom data not available when I decode the packet/frame?
If you're able to use MKV as the container format, you can attach the number as BlockAdditional side-data to the AVPacket, and retrieve it from the AVPacket that is read from the container during playback. Here is a crude example, using strings since they are easy to read.
Note: if you want to correlate the side-data to AVFrames, you will probably need to do some additional tracking to match each AVPacket to the corresponding AVFrame. In FFmpeg, the codec will not preserve the side-data through decoding.
AVPacket* packet; // from avcodec_receive_packet
int64_t frame_number; // number you want to attach
// Allocate the side data on the packet.
size_t side_data_size = 256;
uint8_t* side_data = av_packet_new_side_data(
packet, AV_PKT_DATA_MATROSKA_BLOCKADDITIONAL, side_data_size);
// Set the BlockAddID. Currently only 1 is supported.
// BlockAdditional data with BlockAddID==1 is intended to store
// data for the codec to use.
uint64_t additional_id = 1;
AV_WB64(side_data, additional_id);
// Put your data in the side_data after the ID.
std::string msg = "frame_number=" + std::to_string(frame_number);
std::strcpy(reinterpret_cast<char*>(side_data + 8), msg.c_str());
Then during playback, you can retrieve the side-data like this:
AVPacket* packet; // from av_read_frame
// Get side-data from packet.
int side_data_size;
uint8_t* side_data = av_packet_get_side_data(
&packet, AV_PKT_DATA_MATROSKA_BLOCKADDITIONAL, &side_data_size);
// Print it, or parse it how you want.
std::string msg = (char*)(side_data + 8);
std::cout << msg << std::endl;
I had a need to transmit sound over the network and for this I chose libraries "PortAudio" and "Opus". I am new to working with sound and therefore i don’t know many thing.I am new to working with sound and therefore i don’t know many things, but i read the documentation and looked at some examples, but i still have some problems with encoding/decoding with Opus. I do not understand how to correctly restore the original encoded PСM.I have some sequence of actions:
Some consts
const int FRAMES_PER_BUFFER = 960;
const int SAMPLE_RATE = 48000;
int NUM_CHANNELS = 2;
int totalFrames = 2 * SAMPLE_RATE; /* Record for a few seconds. */
int numSamples = totalFrames * 2;
int numBytes = numSamples * sizeof(float);
float *sampleBlock = nullptr;
int bytesOfPacket = 0;
unsigned char *packet = nullptr;
I get PСM to sampleBlock
paError = Pa_ReadStream(**&stream, sampleBlock, totalFrames);
if (paError != paNoError) {
cout << "PortAudio error : " << Pa_GetErrorText(paError) << endl;
std::system("pause");
}
Encoding sampleBlock
OpusEncoder *encoder;
int error;
int size;
encoder = opus_encoder_create(SAMPLE_RATE, NUM_CHANNELS, OPUS_APPLICATION_VOIP, &error);
size = opus_encoder_get_size(NUM_CHANNELS);
encoder = (OpusEncoder *)malloc(size);
packet = new unsigned char[480];
error = opus_encoder_init(encoder, SAMPLE_RATE, NUM_CHANNELS, OPUS_APPLICATION_VOIP);
if (error == -1) {
return -1;
}
bytesOfPacket = opus_encode_float(encoder, sampleBlock, FRAMES_PER_BUFFER, packet, 480);
opus_encoder_destroy(encoder);
Ok, i received a encoded packet to Opus
Decoding
OpusDecoder *decoder;
int error;
int size;
decoder = opus_decoder_create(SAMPLE_RATE, NUM_CHANNELS, &error);
size = opus_decoder_get_size(NUM_CHANNELS);
decoder = (OpusDecoder *)malloc(size);
error = opus_decoder_init(decoder, SAMPLE_RATE, NUM_CHANNELS);
opus_decode_float(decoder, packet, bytesOfPacket, sampleBlock, 480, 0);
opus_decoder_destroy(decoder);
Here i am trying to decode the Opus back to the PCM and save the result to the sampleBlock
Playing the sound
paError = Pa_WriteStream(**&stream, sampleBlock, totalFrames);
if (paError != paNoError) {
cout << "PortAudio error : " << Pa_GetErrorText(paError) << endl;
std::system("pause");
}
I get silence. I don't really understand the subtleties in working with sound since i am new to this business. Help please understand what is wrong.
As for your settings you're encoding 20ms of audio per opus_encode_float call. I don't see any iteration over this call so I suppose you don't hear anything because you encode only 20ms of audio. You should pass to opus_encode_float 20ms worth of samples with your sampleBlock pointer incrementing it through the whole buffer x times.
Try to encode more audio and remember that you have to add some sort of framing to decode it. You cannot just feed the whole buffer to the decoder. You should feed the decoder one time for each encoder call with the same data that each encoder call outputs.
Damiano
I am converting from the following format:
const int four_bytes_per_float = 4;
const int eight_bits_per_byte = 8;
_stereoGraphStreamFormat.mFormatID = kAudioFormatLinearPCM;
_stereoGraphStreamFormat.mFormatFlags = kAudioFormatFlagsNativeFloatPacked | kAudioFormatFlagIsNonInterleaved;
_stereoGraphStreamFormat.mBytesPerPacket = four_bytes_per_float;
_stereoGraphStreamFormat.mFramesPerPacket = 1;
_stereoGraphStreamFormat.mBytesPerFrame = four_bytes_per_float;
_stereoGraphStreamFormat.mChannelsPerFrame = 2;
_stereoGraphStreamFormat.mBitsPerChannel = eight_bits_per_byte * four_bytes_per_float;
_stereoGraphStreamFormat.mSampleRate = 44100;
to the following format:
interleavedAudioDescription.mFormatID = kAudioFormatLinearPCM;
interleavedAudioDescription.mFormatFlags = kAudioFormatFlagIsSignedInteger;
interleavedAudioDescription.mChannelsPerFrame = 2;
interleavedAudioDescription.mBytesPerPacket = sizeof(SInt16)*interleavedAudioDescription.mChannelsPerFrame;
interleavedAudioDescription.mFramesPerPacket = 1;
interleavedAudioDescription.mBytesPerFrame = sizeof(SInt16)*interleavedAudioDescription.mChannelsPerFrame;
interleavedAudioDescription.mBitsPerChannel = 8 * sizeof(SInt16);
interleavedAudioDescription.mSampleRate = 44100;
Using the following code:
int32_t availableBytes = 0;
void* tailL = TPCircularBufferTail(inputBufferL(), &availableBytes);
void* tailR = TPCircularBufferTail(inputBufferR(), &availableBytes);
// If we have no data in the buffer, we simply return
if (availableBytes <= 0)
{
return;
}
// ========== Non-Interleaved to Interleaved (Plus Samplerate Conversion) =========
// Get the number of frames available
UInt32 frames = availableBytes / this->mInputFormat.mBytesPerFrame;
pcmOutputBuffer->mBuffers[0].mDataByteSize = frames * interleavedAudioDescription.mBytesPerFrame;
struct complexInputDataProc_t data = (struct complexInputDataProc_t) { .self = this, .sourceL = tailL, .sourceR = tailR, .byteLength = availableBytes };
// Do the conversion
OSStatus result = AudioConverterFillComplexBuffer(interleavedAudioConverter,
complexInputDataProc,
&data,
&frames,
pcmOutputBuffer,
NULL);
// Tell the buffers how much data we consumed during the conversion so that it can be removed
TPCircularBufferConsume(inputBufferL(), availableBytes);
TPCircularBufferConsume(inputBufferR(), availableBytes);
// ========== Buffering Of Interleaved Samples =========
// If we got converted frames back from the converter, we want to add it to a separate buffer
if (frames > 0)
{
// Make sure we have enough space in the buffer to store the new data
TPCircularBufferHead(&pcmCircularBuffer, &availableBytes);
if (availableBytes > pcmOutputBuffer->mBuffers[0].mDataByteSize)
{
// Add the newly converted data to the buffer
TPCircularBufferProduceBytes(&pcmCircularBuffer, pcmOutputBuffer->mBuffers[0].mData, frames * interleavedAudioDescription.mBytesPerFrame);
}
else
{
printf("No Space in Buffer\n");
}
}
However I am getting the following output:
It should be a perfect sine wave, however as you can see it is not.
I have been working on this for days now and just can’t seem to find where it is going wrong.
Can anyone see something that I might be missing?
Edit:
Buffer initialisation:
TPCircularBuffer pcmCircularBuffer;
static SInt16 pcmOutputBuf[BUFFER_SIZE];
pcmOutputBuffer = (AudioBufferList*)malloc(sizeof(AudioBufferList));
pcmOutputBuffer->mNumberBuffers = 1;
pcmOutputBuffer->mBuffers[0].mNumberChannels = 2;
pcmOutputBuffer->mBuffers[0].mData = pcmOutputBuf;
TPCircularBufferInit(&pcmCircularBuffer, BUFFER_SIZE);
Complex input data proc:
static OSStatus complexInputDataProc(AudioConverterRef inAudioConverter,
UInt32 *ioNumberDataPackets,
AudioBufferList *ioData,
AudioStreamPacketDescription **outDataPacketDescription,
void *inUserData) {
struct complexInputDataProc_t *arg = (struct complexInputDataProc_t*)inUserData;
BroadcastingServices::MP3Encoder *self = (BroadcastingServices::MP3Encoder*)arg->self;
if ( arg->byteLength <= 0 )
{
*ioNumberDataPackets = 0;
return 100; //kNoMoreDataErr;
}
UInt32 framesAvailable = arg->byteLength / self->interleavedAudioDescription.mBytesPerFrame;
if (*ioNumberDataPackets > framesAvailable)
{
*ioNumberDataPackets = framesAvailable;
}
ioData->mBuffers[0].mData = arg->sourceL;
ioData->mBuffers[0].mDataByteSize = arg->byteLength;
ioData->mBuffers[1].mData = arg->sourceR;
ioData->mBuffers[1].mDataByteSize = arg->byteLength;
arg->byteLength = 0;
return noErr;
}
I see a few things that raise a red flag.
1) as mentioned in a comment above, the fact that you are overwriting availableBytes for the left input with that from the right:
void* tailL = TPCircularBufferTail(inputBufferL(), &availableBytes);
void* tailR = TPCircularBufferTail(inputBufferR(), &availableBytes);
If the two input streams are changing asynchronously to this code then most certainly you have a race condition.
2) Truncation errors: availableBytes is not necessarily a multiple of bytes per frame. If not then the following bit of code could cause you to consume more bytes than you actually converted.
void* tailL = TPCircularBufferTail(inputBufferL(), &availableBytes);
void* tailR = TPCircularBufferTail(inputBufferR(), &availableBytes);
...
UInt32 frames = availableBytes / this->mInputFormat.mBytesPerFrame;
...
TPCircularBufferConsume(inputBufferL(), availableBytes);
TPCircularBufferConsume(inputBufferR(), availableBytes);
3) If the output buffer is not ready to consume all of the input you just throw the input buffer away. That happens in this code.
if (availableBytes > pcmOutputBuffer->mBuffers[0].mDataByteSize)
{
...
}
else
{
printf("No Space in Buffer\n");
}
I'd be really curious if your seeing the print output.
Here's is how I would suggest doing it. It's going to be pseudo-codeish since I don't have anything necessary to compile and test it.
int32_t availableBytesInL = 0;
int32_t availableBytesInR = 0;
int32_t availableBytesOut = 0;
// figure out how many bytes are available in each buffer.
void* tailL = TPCircularBufferTail(inputBufferL(), &availableBytesInL);
void* tailR = TPCircularBufferTail(inputBufferR(), &availableBytesInR);
TPCircularBufferHead(&pcmCircularBuffer, &availableBytesOut);
// figure out how many full frames are available
UInt32 framesInL = availableBytesInL / mInputFormat.mBytesPerFrame;
UInt32 framesInR = availableBytesInR / mInputFormat.mBytesPerFrame;
UInt32 framesOut = availableBytesOut / interleavedAudioDescription.mBytesPerFrame;
// figure out how many frames to process this time.
UInt32 frames = min(min(framesInL, framesInL), framesOut);
if (frames == 0)
return;
int32_t bytesConsumed = frames * mInputFormat.mBytesPerFrame;
struct complexInputDataProc_t data = (struct complexInputDataProc_t) {
.self = this, .sourceL = tailL, .sourceR = tailR, .byteLength = bytesConsumed };
// Do the conversion
OSStatus result = AudioConverterFillComplexBuffer(interleavedAudioConverter,
complexInputDataProc,
&data,
&frames,
pcmOutputBuffer,
NULL);
int32_t bytesProduced = frames * interleavedAudioDescription.mBytesPerFrame;
// Tell the buffers how much data we consumed during the conversion so that it can be removed
TPCircularBufferConsume(inputBufferL(), bytesConsumed);
TPCircularBufferConsume(inputBufferR(), bytesConsumed);
TPCircularBufferProduceBytes(&pcmCircularBuffer, pcmOutputBuffer->mBuffers[0].mData, bytesProduced);
Basically what I've done here is to figure out up front how many frames should be processed making sure I'm only processing as many frames as the output buffer can handle. If it were me I'd also add some checking for buffer underruns on the output and buffer overruns on the input. Finally, I'm not exactly sure of the semantics of AudioConverterFillComplexBuffer wrt the frame parameter that is passing in and out. It's conceivable that the # frames out would be less or more than the number of frames in. Although, since your not doing sample rate conversion that's probably not going to happen. I've attempted to account for that condition by assigning bytesProduced after the conversion.
Hope this helps. If not you have 2 other clues. One is that the drop outs are periodic and two is that the size of the drop outs looks to be about the same. If you can figure out how many samples each are then you can look for those numbers in your code.
I don't think your output buffer, pcmCircularBuffer, is big enough.
Try replacing
TPCircularBufferInit(&pcmCircularBuffer, BUFFER_SIZE);
with
TPCircularBufferInit(&pcmCircularBuffer, sizeof(pcmOutputBuf));
Even if that is the solution, I think there are some problems with your code. I don't know exactly what you're doing, I guess encoding mp3 (which by itself is an uphill battle on iOS, why not use hardware AAC?), but unless you have realtime demands on both input and output, why use ring buffers at all? Also, I recommend using units to visually catch type frame/byte size mismatches: e.g. BUFFER_SIZE_IN_FRAMES
If it's not the solution, then I want to see the sine generator.
I would like to play audio that consists of random, arbitrary bytes.
I am using an AVAudio player from random bytes like so:
self.player = [[AVAudioPlayer alloc] initWithData:[self create20mbRandomNSData] error:&error];
- (NSData *)create20mbRandomNSData
{
int twentyMb = 20971520;
NSMutableData* theData = [NSMutableData dataWithCapacity:twentyMb];
for( unsigned int i = 0 ; i < twentyMb/4 ; ++i )
{
u_int32_t randomBits = arc4random();
[theData appendBytes:(void*)&randomBits length:4];
}
return theData;
}
However, this code produces an error: OSStatus error 1954115647, which translates to: "unsupported file type."
This I understand. Random bytes != an audio file.
My question: Is there a relatively easy way to interpret, read, or convert arbitrary bytes into a file of a specific type? Ideally, I would like to convert to .mp3 or another audio format supported by CoreAudio. However, this question applies to any and all file types.
Thanks!
Im stuck on an issue on my objective C App.
I'm reading a byte array from a serveur (Socket c#) who send me an PCM encoded sound, and i'm currently looking for a sample code that decode for me this byte array (NSData), and play it.
Does anyone know a solution ? Or how can I read a u-Law audio?
Thanks a lot ! :D
This link has information about mu-law encoding and decoding:
http://dystopiancode.blogspot.com.es/2012/02/pcm-law-and-u-law-companding-algorithms.html
#define MULAW_BIAS 33
/*
* Description:
* Decodes an 8-bit unsigned integer using the mu-Law.
* Parameters:
* number - the number who will be decoded
* Returns:
* The decoded number
*/
int16_t MuLaw_Decode(int8_t number)
{
uint8_t sign = 0, position = 0;
int16_t decoded = 0;
number=~number;
if(number&0x80)
{
number&=~(1<<7);
sign = -1;
}
position = ((number & 0xF0) >>4) + 5;
decoded = ((1<<position)|((number&0x0F)<<(position-4))|(1<<(position-5)))
- MULAW_BIAS;
return (sign==0)?(decoded):(-(decoded));
}
When you have the uncompressed audio you should be able to play it using the Audio Queue API.
Good luck!