diff options
Diffstat (limited to 'Alc/backends')
| -rw-r--r-- | Alc/backends/coreaudio.cpp | 317 |
1 files changed, 136 insertions, 181 deletions
diff --git a/Alc/backends/coreaudio.cpp b/Alc/backends/coreaudio.cpp index 808e7cb8..c0b6601b 100644 --- a/Alc/backends/coreaudio.cpp +++ b/Alc/backends/coreaudio.cpp @@ -29,6 +29,7 @@ #include "alMain.h" #include "alu.h" #include "ringbuffer.h" +#include "converter.h" #include <unistd.h> #include <AudioUnit/AudioUnit.h> @@ -39,10 +40,10 @@ static const ALCchar ca_device[] = "CoreAudio Default"; struct ALCcoreAudioPlayback final : public ALCbackend { - AudioUnit AudioUnit; + AudioUnit mAudioUnit; - ALuint FrameSize; - AudioStreamBasicDescription Format; // This is the OpenAL format as a CoreAudio ASBD + ALuint mFrameSize{0u}; + AudioStreamBasicDescription mFormat{}; // This is the OpenAL format as a CoreAudio ASBD }; static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice *device); @@ -66,15 +67,12 @@ static void ALCcoreAudioPlayback_Construct(ALCcoreAudioPlayback *self, ALCdevice new (self) ALCcoreAudioPlayback{}; ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCcoreAudioPlayback, ALCbackend, self); - - self->FrameSize = 0; - self->Format = AudioStreamBasicDescription{}; } static void ALCcoreAudioPlayback_Destruct(ALCcoreAudioPlayback *self) { - AudioUnitUninitialize(self->AudioUnit); - AudioComponentInstanceDispose(self->AudioUnit); + AudioUnitUninitialize(self->mAudioUnit); + AudioComponentInstanceDispose(self->mAudioUnit); ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); self->~ALCcoreAudioPlayback(); @@ -90,7 +88,7 @@ static OSStatus ALCcoreAudioPlayback_MixerProc(void *inRefCon, ALCcoreAudioPlayback_lock(self); aluMixData(device, ioData->mBuffers[0].mData, - ioData->mBuffers[0].mDataByteSize / self->FrameSize); + ioData->mBuffers[0].mDataByteSize / self->mFrameSize); ALCcoreAudioPlayback_unlock(self); return noErr; @@ -127,7 +125,7 @@ static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCch return ALC_INVALID_VALUE; } - err = AudioComponentInstanceNew(comp, &self->AudioUnit); + err = AudioComponentInstanceNew(comp, &self->mAudioUnit); if(err != noErr) { ERR("AudioComponentInstanceNew failed\n"); @@ -135,11 +133,11 @@ static ALCenum ALCcoreAudioPlayback_open(ALCcoreAudioPlayback *self, const ALCch } /* init and start the default audio unit... */ - err = AudioUnitInitialize(self->AudioUnit); + err = AudioUnitInitialize(self->mAudioUnit); if(err != noErr) { ERR("AudioUnitInitialize failed\n"); - AudioComponentInstanceDispose(self->AudioUnit); + AudioComponentInstanceDispose(self->mAudioUnit); return ALC_INVALID_VALUE; } @@ -155,13 +153,14 @@ static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self) OSStatus err; UInt32 size; - err = AudioUnitUninitialize(self->AudioUnit); + err = AudioUnitUninitialize(self->mAudioUnit); if(err != noErr) ERR("-- AudioUnitUninitialize failed.\n"); /* retrieve default output unit's properties (output side) */ size = sizeof(AudioStreamBasicDescription); - err = AudioUnitGetProperty(self->AudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 0, &streamFormat, &size); + err = AudioUnitGetProperty(self->mAudioUnit, kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Output, 0, &streamFormat, &size); if(err != noErr || size != sizeof(AudioStreamBasicDescription)) { ERR("AudioUnitGetProperty failed\n"); @@ -179,7 +178,8 @@ static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self) #endif /* set default output unit's input side to match output side */ - err = AudioUnitSetProperty(self->AudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, size); + err = AudioUnitSetProperty(self->mAudioUnit, kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, 0, &streamFormat, size); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); @@ -263,7 +263,8 @@ static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self) streamFormat.mFormatFlags |= kAudioFormatFlagsNativeEndian | kLinearPCMFormatFlagIsPacked; - err = AudioUnitSetProperty(self->AudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription)); + err = AudioUnitSetProperty(self->mAudioUnit, kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, 0, &streamFormat, sizeof(AudioStreamBasicDescription)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); @@ -271,11 +272,12 @@ static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self) } /* setup callback */ - self->FrameSize = device->frameSizeFromFmt(); + self->mFrameSize = device->frameSizeFromFmt(); input.inputProc = ALCcoreAudioPlayback_MixerProc; input.inputProcRefCon = self; - err = AudioUnitSetProperty(self->AudioUnit, kAudioUnitProperty_SetRenderCallback, kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct)); + err = AudioUnitSetProperty(self->mAudioUnit, kAudioUnitProperty_SetRenderCallback, + kAudioUnitScope_Input, 0, &input, sizeof(AURenderCallbackStruct)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); @@ -283,7 +285,7 @@ static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self) } /* init the default audio unit... */ - err = AudioUnitInitialize(self->AudioUnit); + err = AudioUnitInitialize(self->mAudioUnit); if(err != noErr) { ERR("AudioUnitInitialize failed\n"); @@ -295,7 +297,7 @@ static ALCboolean ALCcoreAudioPlayback_reset(ALCcoreAudioPlayback *self) static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self) { - OSStatus err = AudioOutputUnitStart(self->AudioUnit); + OSStatus err = AudioOutputUnitStart(self->mAudioUnit); if(err != noErr) { ERR("AudioOutputUnitStart failed\n"); @@ -307,24 +309,21 @@ static ALCboolean ALCcoreAudioPlayback_start(ALCcoreAudioPlayback *self) static void ALCcoreAudioPlayback_stop(ALCcoreAudioPlayback *self) { - OSStatus err = AudioOutputUnitStop(self->AudioUnit); + OSStatus err = AudioOutputUnitStop(self->mAudioUnit); if(err != noErr) ERR("AudioOutputUnitStop failed\n"); } struct ALCcoreAudioCapture final : public ALCbackend { - AudioUnit AudioUnit; + AudioUnit mAudioUnit{0}; - ALuint FrameSize; - ALdouble SampleRateRatio; // Ratio of hardware sample rate / requested sample rate - AudioStreamBasicDescription Format; // This is the OpenAL format as a CoreAudio ASBD + ALuint mFrameSize{0u}; + AudioStreamBasicDescription mFormat{}; // This is the OpenAL format as a CoreAudio ASBD - AudioConverterRef AudioConverter; // Sample rate converter if needed - AudioBufferList *BufferList; // Buffer for data coming from the input device - ALCvoid *ResampleBuffer; // Buffer for returned RingBuffer data when resampling + std::unique_ptr<SampleConverter> mConverter; - RingBufferPtr Ring{nullptr}; + RingBufferPtr mRing{nullptr}; }; static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device); @@ -343,56 +342,18 @@ DECLARE_DEFAULT_ALLOCATORS(ALCcoreAudioCapture) DEFINE_ALCBACKEND_VTABLE(ALCcoreAudioCapture); -static AudioBufferList *allocate_buffer_list(UInt32 channelCount, UInt32 byteSize) -{ - AudioBufferList *list; - - list = static_cast<AudioBufferList*>(calloc(1, - FAM_SIZE(AudioBufferList, mBuffers, 1) + byteSize)); - if(list) - { - list->mNumberBuffers = 1; - - list->mBuffers[0].mNumberChannels = channelCount; - list->mBuffers[0].mDataByteSize = byteSize; - list->mBuffers[0].mData = &list->mBuffers[1]; - } - return list; -} - -static void destroy_buffer_list(AudioBufferList *list) -{ - free(list); -} - - static void ALCcoreAudioCapture_Construct(ALCcoreAudioCapture *self, ALCdevice *device) { new (self) ALCcoreAudioCapture{}; ALCbackend_Construct(STATIC_CAST(ALCbackend, self), device); SET_VTABLE2(ALCcoreAudioCapture, ALCbackend, self); - - self->AudioUnit = 0; - self->AudioConverter = NULL; - self->BufferList = NULL; - self->ResampleBuffer = NULL; } static void ALCcoreAudioCapture_Destruct(ALCcoreAudioCapture *self) { - free(self->ResampleBuffer); - self->ResampleBuffer = NULL; - - destroy_buffer_list(self->BufferList); - self->BufferList = NULL; - - if(self->AudioConverter) - AudioConverterDispose(self->AudioConverter); - self->AudioConverter = NULL; - - if(self->AudioUnit) - AudioComponentInstanceDispose(self->AudioUnit); - self->AudioUnit = 0; + if(self->mAudioUnit) + AudioComponentInstanceDispose(self->mAudioUnit); + self->mAudioUnit = 0; ALCbackend_Destruct(STATIC_CAST(ALCbackend, self)); self->~ALCcoreAudioCapture(); @@ -405,39 +366,47 @@ static OSStatus ALCcoreAudioCapture_RecordProc(void *inRefCon, UInt32 inNumberFrames, AudioBufferList* UNUSED(ioData)) { auto self = static_cast<ALCcoreAudioCapture*>(inRefCon); - RingBuffer *ring{self->Ring.get()}; + RingBuffer *ring{self->mRing.get()}; AudioUnitRenderActionFlags flags = 0; + union { + ALbyte _[sizeof(AudioBufferList) + sizeof(AudioBuffer)]; + AudioBufferList list; + } audiobuf = { { 0 } }; OSStatus err; - // fill the BufferList with data from the input device - err = AudioUnitRender(self->AudioUnit, &flags, inTimeStamp, 1, inNumberFrames, self->BufferList); + auto rec_vec = ring->getWriteVector(); + + // Fill the ringbuffer's first segment with data from the input device + size_t total_read{minz(rec_vec.first.len, inNumberFrames)}; + audiobuf.list.mNumberBuffers = 1; + audiobuf.list.mBuffers[0].mNumberChannels = self->mFormat.mChannelsPerFrame; + audiobuf.list.mBuffers[0].mData = rec_vec.first.buf; + audiobuf.list.mBuffers[0].mDataByteSize = total_read * self->mFormat.mBytesPerFrame; + err = AudioUnitRender(self->mAudioUnit, &flags, inTimeStamp, 1, inNumberFrames, + &audiobuf.list); + if(err == noErr && inNumberFrames > rec_vec.first.len && rec_vec.second.len > 0) + { + /* If there's still more to get and there's space in the ringbuffer's + * second segment, fill that with data too. + */ + const size_t remlen{inNumberFrames - rec_vec.first.len}; + const size_t toread{minz(rec_vec.second.len, remlen)}; + total_read += toread; + + audiobuf.list.mNumberBuffers = 1; + audiobuf.list.mBuffers[0].mNumberChannels = self->mFormat.mChannelsPerFrame; + audiobuf.list.mBuffers[0].mData = rec_vec.second.buf; + audiobuf.list.mBuffers[0].mDataByteSize = toread * self->mFormat.mBytesPerFrame; + err = AudioUnitRender(self->mAudioUnit, &flags, inTimeStamp, 1, inNumberFrames, + &audiobuf.list); + } if(err != noErr) { ERR("AudioUnitRender error: %d\n", err); return err; } - ring->write(self->BufferList->mBuffers[0].mData, inNumberFrames); - return noErr; -} - -static OSStatus ALCcoreAudioCapture_ConvertCallback(AudioConverterRef UNUSED(inAudioConverter), - UInt32 *ioNumberDataPackets, AudioBufferList *ioData, - AudioStreamPacketDescription** UNUSED(outDataPacketDescription), - void *inUserData) -{ - auto self = reinterpret_cast<ALCcoreAudioCapture*>(inUserData); - RingBuffer *ring{self->Ring.get()}; - - // Read from the ring buffer and store temporarily in a large buffer - ring->read(self->ResampleBuffer, *ioNumberDataPackets); - - // Set the input data - ioData->mNumberBuffers = 1; - ioData->mBuffers[0].mNumberChannels = self->Format.mChannelsPerFrame; - ioData->mBuffers[0].mData = self->ResampleBuffer; - ioData->mBuffers[0].mDataByteSize = (*ioNumberDataPackets) * self->Format.mBytesPerFrame; - + ring->writeAdvance(total_read); return noErr; } @@ -481,29 +450,31 @@ static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar } // Open the component - err = AudioComponentInstanceNew(comp, &self->AudioUnit); + err = AudioComponentInstanceNew(comp, &self->mAudioUnit); if(err != noErr) { ERR("AudioComponentInstanceNew failed\n"); - goto error; + return ALC_INVALID_VALUE; } // Turn off AudioUnit output enableIO = 0; - err = AudioUnitSetProperty(self->AudioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint)); + err = AudioUnitSetProperty(self->mAudioUnit, kAudioOutputUnitProperty_EnableIO, + kAudioUnitScope_Output, 0, &enableIO, sizeof(ALuint)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); - goto error; + return ALC_INVALID_VALUE; } // Turn on AudioUnit input enableIO = 1; - err = AudioUnitSetProperty(self->AudioUnit, kAudioOutputUnitProperty_EnableIO, kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint)); + err = AudioUnitSetProperty(self->mAudioUnit, kAudioOutputUnitProperty_EnableIO, + kAudioUnitScope_Input, 1, &enableIO, sizeof(ALuint)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); - goto error; + return ALC_INVALID_VALUE; } #if !TARGET_OS_IOS @@ -520,20 +491,21 @@ static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar if(err != noErr) { ERR("AudioObjectGetPropertyData failed\n"); - goto error; + return ALC_INVALID_VALUE; } if(inputDevice == kAudioDeviceUnknown) { ERR("No input device found\n"); - goto error; + return ALC_INVALID_VALUE; } // Track the input device - err = AudioUnitSetProperty(self->AudioUnit, kAudioOutputUnitProperty_CurrentDevice, kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID)); + err = AudioUnitSetProperty(self->mAudioUnit, kAudioOutputUnitProperty_CurrentDevice, + kAudioUnitScope_Global, 0, &inputDevice, sizeof(AudioDeviceID)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); - goto error; + return ALC_INVALID_VALUE; } } #endif @@ -542,28 +514,30 @@ static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar input.inputProc = ALCcoreAudioCapture_RecordProc; input.inputProcRefCon = self; - err = AudioUnitSetProperty(self->AudioUnit, kAudioOutputUnitProperty_SetInputCallback, kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct)); + err = AudioUnitSetProperty(self->mAudioUnit, kAudioOutputUnitProperty_SetInputCallback, + kAudioUnitScope_Global, 0, &input, sizeof(AURenderCallbackStruct)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); - goto error; + return ALC_INVALID_VALUE; } // Initialize the device - err = AudioUnitInitialize(self->AudioUnit); + err = AudioUnitInitialize(self->mAudioUnit); if(err != noErr) { ERR("AudioUnitInitialize failed\n"); - goto error; + return ALC_INVALID_VALUE; } // Get the hardware format propertySize = sizeof(AudioStreamBasicDescription); - err = AudioUnitGetProperty(self->AudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize); + err = AudioUnitGetProperty(self->mAudioUnit, kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Input, 1, &hardwareFormat, &propertySize); if(err != noErr || propertySize != sizeof(AudioStreamBasicDescription)) { ERR("AudioUnitGetProperty failed\n"); - goto error; + return ALC_INVALID_VALUE; } // Set up the requested format description @@ -589,7 +563,7 @@ static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar case DevFmtUShort: case DevFmtUInt: ERR("%s samples not supported\n", DevFmtTypeString(device->FmtType)); - goto error; + return ALC_INVALID_VALUE; } switch(device->FmtChans) @@ -608,7 +582,7 @@ static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar case DevFmtX71: case DevFmtAmbi3D: ERR("%s not supported\n", DevFmtChannelsString(device->FmtChans)); - goto error; + return ALC_INVALID_VALUE; } requestedFormat.mBytesPerFrame = requestedFormat.mChannelsPerFrame * requestedFormat.mBitsPerChannel / 8; @@ -619,80 +593,61 @@ static ALCenum ALCcoreAudioCapture_open(ALCcoreAudioCapture *self, const ALCchar requestedFormat.mFramesPerPacket = 1; // save requested format description for later use - self->Format = requestedFormat; - self->FrameSize = device->frameSizeFromFmt(); + self->mFormat = requestedFormat; + self->mFrameSize = device->frameSizeFromFmt(); // Use intermediate format for sample rate conversion (outputFormat) // Set sample rate to the same as hardware for resampling later outputFormat = requestedFormat; outputFormat.mSampleRate = hardwareFormat.mSampleRate; - // Determine sample rate ratio for resampling - self->SampleRateRatio = outputFormat.mSampleRate / device->Frequency; - // The output format should be the requested format, but using the hardware sample rate // This is because the AudioUnit will automatically scale other properties, except for sample rate - err = AudioUnitSetProperty(self->AudioUnit, kAudioUnitProperty_StreamFormat, kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat)); + err = AudioUnitSetProperty(self->mAudioUnit, kAudioUnitProperty_StreamFormat, + kAudioUnitScope_Output, 1, (void *)&outputFormat, sizeof(outputFormat)); if(err != noErr) { ERR("AudioUnitSetProperty failed\n"); - goto error; + return ALC_INVALID_VALUE; } // Set the AudioUnit output format frame count - outputFrameCount = device->UpdateSize * self->SampleRateRatio; - err = AudioUnitSetProperty(self->AudioUnit, kAudioUnitProperty_MaximumFramesPerSlice, kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount)); - if(err != noErr) + ALuint64 FrameCount64{device->UpdateSize}; + FrameCount64 = (FrameCount64*outputFormat.mSampleRate + device->Frequency-1) / + device->Frequency; + FrameCount64 += MAX_RESAMPLE_PADDING*2; + if(FrameCount64 > std::numeric_limits<uint32_t>::max()/2) { - ERR("AudioUnitSetProperty failed: %d\n", err); - goto error; + ERR("FrameCount too large\n"); + return ALC_INVALID_VALUE; } - // Set up sample converter - err = AudioConverterNew(&outputFormat, &requestedFormat, &self->AudioConverter); + outputFrameCount = static_cast<uint32_t>(FrameCount64); + err = AudioUnitSetProperty(self->mAudioUnit, kAudioUnitProperty_MaximumFramesPerSlice, + kAudioUnitScope_Output, 0, &outputFrameCount, sizeof(outputFrameCount)); if(err != noErr) { - ERR("AudioConverterNew failed: %d\n", err); - goto error; + ERR("AudioUnitSetProperty failed: %d\n", err); + return ALC_INVALID_VALUE; } - // Create a buffer for use in the resample callback - self->ResampleBuffer = malloc(device->UpdateSize * self->FrameSize * self->SampleRateRatio); + // Set up sample converter if needed + if(outputFormat.mSampleRate != device->Frequency) + self->mConverter.reset(CreateSampleConverter(device->FmtType, device->FmtType, + self->mFormat.mChannelsPerFrame, hardwareFormat.mSampleRate, device->Frequency, + BSinc24Resampler)); - // Allocate buffer for the AudioUnit output - self->BufferList = allocate_buffer_list(outputFormat.mChannelsPerFrame, device->UpdateSize * self->FrameSize * self->SampleRateRatio); - if(self->BufferList == NULL) - goto error; - - self->Ring.reset(ll_ringbuffer_create( - (size_t)ceil(device->UpdateSize*self->SampleRateRatio*device->NumUpdates), - self->FrameSize, false)); - if(!self->Ring) goto error; + self->mRing.reset(ll_ringbuffer_create(outputFrameCount, self->mFrameSize, false)); + if(!self->mRing) return ALC_INVALID_VALUE; device->DeviceName = name; return ALC_NO_ERROR; - -error: - self->Ring = nullptr; - free(self->ResampleBuffer); - self->ResampleBuffer = NULL; - destroy_buffer_list(self->BufferList); - self->BufferList = NULL; - - if(self->AudioConverter) - AudioConverterDispose(self->AudioConverter); - self->AudioConverter = NULL; - if(self->AudioUnit) - AudioComponentInstanceDispose(self->AudioUnit); - self->AudioUnit = 0; - - return ALC_INVALID_VALUE; } static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self) { - OSStatus err = AudioOutputUnitStart(self->AudioUnit); + OSStatus err = AudioOutputUnitStart(self->mAudioUnit); if(err != noErr) { ERR("AudioOutputUnitStart failed\n"); @@ -703,46 +658,46 @@ static ALCboolean ALCcoreAudioCapture_start(ALCcoreAudioCapture *self) static void ALCcoreAudioCapture_stop(ALCcoreAudioCapture *self) { - OSStatus err = AudioOutputUnitStop(self->AudioUnit); + OSStatus err = AudioOutputUnitStop(self->mAudioUnit); if(err != noErr) ERR("AudioOutputUnitStop failed\n"); } static ALCenum ALCcoreAudioCapture_captureSamples(ALCcoreAudioCapture *self, ALCvoid *buffer, ALCuint samples) { - union { - ALbyte _[sizeof(AudioBufferList) + sizeof(AudioBuffer)]; - AudioBufferList list; - } audiobuf = { { 0 } }; - UInt32 frameCount; - OSStatus err; + RingBuffer *ring{self->mRing.get()}; - // If no samples are requested, just return - if(samples == 0) return ALC_NO_ERROR; + if(!self->mConverter) + { + ring->read(buffer, samples); + return ALC_NO_ERROR; + } - // Point the resampling buffer to the capture buffer - audiobuf.list.mNumberBuffers = 1; - audiobuf.list.mBuffers[0].mNumberChannels = self->Format.mChannelsPerFrame; - audiobuf.list.mBuffers[0].mDataByteSize = samples * self->FrameSize; - audiobuf.list.mBuffers[0].mData = buffer; - - // Resample into another AudioBufferList - frameCount = samples; - err = AudioConverterFillComplexBuffer(self->AudioConverter, - ALCcoreAudioCapture_ConvertCallback, self, &frameCount, &audiobuf.list, NULL - ); - if(err != noErr) + auto rec_vec = ring->getReadVector(); + const void *src0{rec_vec.first.buf}; + auto src0len = static_cast<ALsizei>(rec_vec.first.len); + auto got = static_cast<ALuint>(SampleConverterInput(self->mConverter.get(), &src0, &src0len, + buffer, samples)); + size_t total_read{rec_vec.first.len - src0len}; + if(got < samples && !src0len && rec_vec.second.len > 0) { - ERR("AudioConverterFillComplexBuffer error: %d\n", err); - return ALC_INVALID_VALUE; + const void *src1{rec_vec.second.buf}; + auto src1len = static_cast<ALsizei>(rec_vec.second.len); + got += static_cast<ALuint>(SampleConverterInput(self->mConverter.get(), &src1, &src1len, + static_cast<char*>(buffer)+got, samples-got)); + total_read += rec_vec.second.len - src1len; } + + ring->readAdvance(total_read); return ALC_NO_ERROR; } static ALCuint ALCcoreAudioCapture_availableSamples(ALCcoreAudioCapture *self) { - RingBuffer *ring{self->Ring.get()}; - return ring->readSpace() / self->SampleRateRatio; + RingBuffer *ring{self->mRing.get()}; + + if(!self->mConverter) return ring->readSpace(); + return SampleConverterAvailableOut(self->mConverter.get(), ring->readSpace()); } |