Opus for iOS, crashing with 16000 sample rate

I am developing Voip application with Opus for iOS (Objective-C and C++). It works fine with 8000, 12000, 24000 and 48000 sampling rate except with 16000, where the application crashes on opus_encode method. Here is what i am doing: m_oAudioSession = [AVAudioSession sharedInstance]; [m_oAudioSession setCategory:AVAudioSessionCategoryPlayAndRecord error:&m_oError]; [m_oAudioSession setMode:AVAudioSessionModeVoiceChat error:&m_oError]; [m_oAudioSession setPreferredSampleRate:VOIP_AUDIO_DRIVER_DEFAULT_SAMPLE_RATE error:&m_oError]; [m_oAudioSession setPreferredInputNumberOfChannels:VOIP_AUDIO_DRIVER_DEFAULT_INPUT_CHANNELS error:&m_oError]; [m_oAudioSession setPreferredOutputNumberOfChannels:VOIP_AUDIO_DRIVER_DEFAULT_OUTPUT_CHANNELS error:&m_oError]; [m_oAudioSession setPreferredIOBufferDuration:VOIP_AUDIO_DRIVER_DEFAULT_BUFFER_DURATION error:&m_oError]; [m_oAudioSession setActive:YES error:&m_oError]; Constants: VOIP_AUDIO_DRIVER_DEFAULT_SAMPLE_RATE is 16000 VOIP_AUDIO_DRIVER_DEFAULT_INPUT_CHANNELS is 1 VOIP_AUDIO_DRIVER_DEFAULT_OUTPUT_CHANNELS is 1 VOIP_AUDIO_DRIVER_DEFAULT_BUFFER_DURATION is 0.02 VOIP_AUDIO_DRIVER_FRAMES_PER_PACKET is 1 After that i am using a real sampling rate and buffer duration from m_oAudioSession.sampleRate and m_oAudioSession.IOBufferDuration. They are set into m_fSampleRate and m_fBufferDuration variables. The configurations are: //Describes audio component: m_sAudioDescription.componentType = kAudioUnitType_Output; m_sAudioDescription.componentSubType = kAudioUnitSubType_VoiceProcessingIO/*kAudioUnitSubType_RemoteIO*/; m_sAudioDescription.componentFlags = 0; m_sAudioDescription.componentFlagsMask = 0; m_sAudioDescription.componentManufacturer = kAudioUnitManufacturer_Apple; m_sAudioFormat.mSampleRate = m_fSampleRate; m_sAudioFormat.mFormatID = kAudioFormatLinearPCM; m_sAudioFormat.mFormatFlags = kAudioFormatFlagIsSignedInteger | kAudioFormatFlagIsPacked; m_sAudioFormat.mFramesPerPacket = VOIP_AUDIO_DRIVER_FRAMES_PER_PACKET; m_sAudioFormat.mChannelsPerFrame = VOIP_AUDIO_DRIVER_DEFAULT_INPUT_CHANNELS; m_sAudioFormat.mBitsPerChannel = (UInt32)(8 * m_iBytesPerSample); m_sAudioFormat.mBytesPerFrame = (UInt32)((m_sAudioFormat.mBitsPerChannel / 8) * m_sAudioFormat.mChannelsPerFrame); m_sAudioFormat.mBytesPerPacket = m_sAudioFormat.mBytesPerFrame * m_sAudioFormat.mFramesPerPacket; m_sAudioFormat.mReserved = 0; The calculations i make are: m_iBytesPerSample = sizeof(/*AudioSampleType*/SInt16); //Calculating buffer size: int samplesPerFrame = (int)(m_fBufferDuration * m_fSampleRate) + 1; m_iBufferSizeBytes = samplesPerFrame * m_iBytesPerSample; //Allocating input buffer: UInt32 inputBufferListSize = offsetof(AudioBufferList, mBuffers[0]) + (sizeof(AudioBuffer) * m_sAudioFormat.mChannelsPerFrame); m_sInputBuffer = (AudioBufferList *)VoipAlloc(inputBufferListSize); m_sInputBuffer->mNumberBuffers = m_sAudioFormat.mChannelsPerFrame; //Pre-mallocating buffers for AudioBufferLists for(VoipUInt32 tmp_int1 = 0; tmp_int1 < m_sInputBuffer->mNumberBuffers; tmp_int1++) { m_sInputBuffer->mBuffers[tmp_int1].mNumberChannels = VOIP_AUDIO_DRIVER_DEFAULT_INPUT_CHANNELS; m_sInputBuffer->mBuffers[tmp_int1].mDataByteSize = (UInt32)m_iBufferSizeBytes; m_sInputBuffer->mBuffers[tmp_int1].mData = VoipAlloc(m_iBufferSizeBytes); memset(m_sInputBuffer->mBuffers[tmp_int1].mData, 0, m_iBufferSizeBytes); } The reading and writing from audio unit are done using m_sInputBuffer. Here is the Opus creation: m_oEncoder = opus_encoder_create(m_iSampleRate, m_iNumberOfChannels, VOIP_AUDIO_CODECS_OPUS_APPLICATION_TYPE, &_error); if (_error < 0) { fprintf(stderr, "VoipAudioCodecs error: failed to create an encoder: %sn", opus_strerror(_error)); return; } _error = opus_encoder_ctl(m_oEncoder, OPUS_SET_BITRATE(VOIP_AUDIO_CODECS_OPUS_BITRATE)); if (_error < 0) { fprintf(stderr, "VoipAudioCodecs error: failed to set the bitrate: %sn", opus_strerror(_error)); return; } m_oDecoder = opus_decoder_create(m_iSampleRate, m_iNumberOfChannels, &_error); if (_error < 0) { fprintf(stderr, "VoipAudioCodecs error: failed to create a decoder: %sn", opus_strerror(_error)); return; } Opus configurations are: VOIP_AUDIO_CODECS_OPUS_BITRATE is OPUS_BITRATE_MAX //64000 //70400 //84800 //112000 VOIP_AUDIO_CODECS_OPUS_APPLICATION_TYPE is OPUS_APPLICATION_VOIP //OPUS_APPLICATION_AUDIO VOIP_AUDIO_CODECS_OPUS_MAX_FRAME_SIZE is 5760 //Minimum: (120ms; 5760 for 48kHz) VOIP_AUDIO_CODECS_OPUS_BYTES_SIZE is 960 //120, 240, 480, 960, 1920, 2880 When i encode and decode i use these methods: Encode_Opus(VoipInt16* rawSamples, int rawSamplesSize) { unsigned char encodedData[m_iMaxPacketSize]; VoipInt32 bytesEncoded; int frameSize = rawSamplesSize / m_iBytesPerSample; bytesEncoded = opus_encode(m_oEncoder, rawSamples, frameSize, encodedData, m_iMaxPacketSize); if (bytesEncoded < 0) { fprintf(stderr, "VoipAudioCodecs error: encode failed: %sn", opus_strerror(bytesEncoded)); return nullptr; } sVoipAudioCodecOpusEncoded* resultStruct = (sVoipAudioCodecOpusEncoded* )VoipAlloc(sizeof(sVoipAudioCodecOpusEncoded)); resultStruct->m_data = (unsigned char*)VoipAlloc(bytesEncoded); memcpy(resultStruct->m_data, encodedData, bytesEncoded); resultStruct->m_dataSize = bytesEncoded; return resultStruct; } Decode_Opus(void* encodedSamples, VoipInt32 encodedSamplesSize) { VoipInt16 decodedPacket[VOIP_AUDIO_CODECS_OPUS_MAX_FRAME_SIZE]; int _frameSize = opus_decode(m_oDecoder, (const unsigned char*)encodedSamples, encodedSamplesSize, decodedPacket, VOIP_AUDIO_CODECS_OPUS_MAX_FRAME_SIZE, 0); if (_frameSize < 0) { fprintf(stderr, "VoipAudioCodecs error: decoder failed: %sn", opus_strerror(_frameSize)); return nullptr; } size_t frameSize = (size_t)_frameSize; sVoipAudioCodecOpusDecoded* resultStruct = (sVoipAudioCodecOpusDecoded* )VoipAlloc(sizeof(sVoipAudioCodecOpusDecoded)); resultStruct->m_data = (VoipInt16*)VoipAlloc(frameSize * m_iBytesPerSample); memcpy(resultStruct->m_data, decodedPacket, (frameSize * m_iBytesPerSample)); resultStruct->m_dataSize = frameSize * m_iBytesPerSample; return resultStruct; } When the app should send data: VoipUInt32 itemsForProcess = inputAudioQueue->getItemCount(); for (int tmp_queueItems = 0; tmp_queueItems < itemsForProcess; tmp_queueItems++) { sVoipQueue* tmp_samples = inputAudioQueue->popItem(); m_oCircularTempInputBuffer->writeDataToBuffer(tmp_samples->m_pData, tmp_samples->m_iDataSize); while (void* tmp_buffer = m_oCircularTempInputBuffer->readDataFromBuffer(VOIP_AUDIO_CODECS_OPUS_BYTES_SIZE)) { sVoipAudioCodecOpusEncoded* encodedSamples = Encode_Opus((VoipInt16*)tmp_buffer, VOIP_AUDIO_CODECS_OPUS_BYTES_SIZE); //Then packeting and the real sending using tcp socket… } //Rest of the code… } Here is the reading: sVoipAudioCodecOpusDecoded* decodedSamples = Decode_Opus(inputPacket->m_pPacketData, (VoipInt32)inputPacket->m_iPacketSize); if (decodedSamples != nullptr) { m_oCircularTempOutputBuffer->writeDataToBuffer(decodedSamples->m_data, decodedSamples->m_dataSize); VoipFree((void**)&decodedSamples->m_data); VoipFree((void**)&decodedSamples); } while (void* tmp_buffer = m_oCircularTempOutputBuffer->readDataFromBuffer(m_iBufferSizeBytes)) { outputAudioQueue->pushItem(tmp_buffer, m_iBufferSizeBytes); } inputAudioQueue is queue with recorded data from my audio unit’s callback. outputAudioQueue is a queue used from my audio unit’s callback to play the sound. m_iMaxPacketSize is the same as m_iBufferSizeBytes. My questions are: I was wondering, are my calculations correct? And if not, how can i improve them? Do you see any mistake into the code? Do you have a suggestion for fixing the crash bug on opus_encode method when the sampling rate is set to 16000? Thank you in advance. PS. I made some tests with sampling rate on 16000 and found this: If i use this formula: frame_duration = frame_size / sample rate, and if i set frame_duration to preferedIOBufferDuration: 120 / 16000 = 0.0075 //AVAudioSession sets 0.008000 —— Crashes 240 / 16000 = 0.015 //AVAudioSession sets 0.016000 —— Crashes 480 / 16000 = 0.03 //AVAudioSession sets 0.032000 —— Crashes 960 / 16000 = 0.06 //AVAudioSession sets 0.064000 —— Crashes 1920 / 16000 = 0.12 //AVAudioSession sets 0.128000 —— Works 2880 / 16000 = 0.18 //AVAudioSession sets 0.128000 —— CrashesThen i found that there is no encoder crash with sampling rate 16000 and preferredIOBufferDuration 0.12(1920), where AVAudioSession sets 0.128000. So it works only in this case. Any ideas ?