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This fix reverts the logic so that old sounds don't go through a pcm->float->pcm conversion. #rb Jimmy.Smith, Dan.Thompson #preflight 636ae824dc30a4ce96c34f99 [CL 23069835 by phil popp in ue5-main branch]
579 lines
18 KiB
C++
579 lines
18 KiB
C++
// Copyright Epic Games, Inc. All Rights Reserved.
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#include "AudioFormatOpus.h"
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#include "Audio.h"
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#include "Serialization/MemoryWriter.h"
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#include "Modules/ModuleManager.h"
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#include "Interfaces/IAudioFormat.h"
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#include "Interfaces/IAudioFormatModule.h"
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#include "OpusAudioInfo.h"
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#include "VorbisAudioInfo.h"
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// Need to define this so that resampler.h compiles - probably a way around this somehow
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#define OUTSIDE_SPEEX
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THIRD_PARTY_INCLUDES_START
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#include "opus_multistream.h"
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#include "speex_resampler.h"
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THIRD_PARTY_INCLUDES_END
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/** Use UE memory allocation or Opus */
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#define USE_UE_MEM_ALLOC 1
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#define SAMPLE_SIZE ( ( uint32 )sizeof( short ) )
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static FName NAME_OPUS(TEXT("OPUS"));
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/**
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* IAudioFormat, audio compression abstraction
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**/
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class FAudioFormatOpus : public IAudioFormat
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{
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enum
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{
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/** Version for OPUS format, this becomes part of the DDC key. */
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UE_AUDIO_OPUS_VER = 8,
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};
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public:
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virtual bool AllowParallelBuild() const override
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{
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return false;
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}
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virtual uint16 GetVersion(FName Format) const override
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{
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check(Format == NAME_OPUS);
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return UE_AUDIO_OPUS_VER;
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}
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virtual void GetSupportedFormats(TArray<FName>& OutFormats) const override
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{
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OutFormats.Add(NAME_OPUS);
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}
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virtual bool Cook(FName Format, const TArray<uint8>& SrcBuffer, FSoundQualityInfo& QualityInfo, TArray<uint8>& CompressedDataStore) const override
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{
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TRACE_CPUPROFILER_EVENT_SCOPE(FAudioFormatOpus::Cook);
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check(Format == NAME_OPUS);
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// Get best compatible sample rate
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const uint16 kOpusSampleRate = GetBestOutputSampleRate(QualityInfo.SampleRate);
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// Frame size must be one of 2.5, 5, 10, 20, 40 or 60 ms
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const int32 kOpusFrameSizeMs = 60;
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// Calculate frame size required by Opus
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const int32 kOpusFrameSizeSamples = (kOpusSampleRate * kOpusFrameSizeMs) / 1000;
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const uint32 kSampleStride = SAMPLE_SIZE * QualityInfo.NumChannels;
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const int32 kBytesPerFrame = kOpusFrameSizeSamples * kSampleStride;
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// Check whether source has compatible sample rate
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TArray<uint8> SrcBufferCopy;
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if (QualityInfo.SampleRate != kOpusSampleRate)
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{
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if (!ResamplePCM(QualityInfo.NumChannels, SrcBuffer, QualityInfo.SampleRate, SrcBufferCopy, kOpusSampleRate))
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{
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return false;
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}
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}
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else
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{
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// Take a copy of the source regardless
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SrcBufferCopy = SrcBuffer;
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}
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// Initialise the Opus encoder
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OpusEncoder* Encoder = NULL;
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int32 EncError = 0;
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#if USE_UE_MEM_ALLOC
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int32 EncSize = opus_encoder_get_size(QualityInfo.NumChannels);
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Encoder = (OpusEncoder*)FMemory::Malloc(EncSize);
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EncError = opus_encoder_init(Encoder, kOpusSampleRate, QualityInfo.NumChannels, OPUS_APPLICATION_AUDIO);
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#else
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Encoder = opus_encoder_create(kOpusSampleRate, QualityInfo.NumChannels, OPUS_APPLICATION_AUDIO, &EncError);
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#endif
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if (EncError != OPUS_OK)
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{
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Destroy(Encoder);
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return false;
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}
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int32 BitRate = GetBitRateFromQuality(QualityInfo);
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opus_encoder_ctl(Encoder, OPUS_SET_BITRATE(BitRate));
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// Create a buffer to store compressed data
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CompressedDataStore.Empty();
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FMemoryWriter CompressedData(CompressedDataStore);
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int32 SrcBufferOffset = 0;
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// Calc frame and sample count
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int32 FramesToEncode = SrcBufferCopy.Num() / kBytesPerFrame;
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uint32 TrueSampleCount = SrcBufferCopy.Num() / kSampleStride;
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// Pad the end of data with zeroes if it isn't exactly the size of a frame.
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if (SrcBufferCopy.Num() % kBytesPerFrame != 0)
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{
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int32 FrameDiff = kBytesPerFrame - (SrcBufferCopy.Num() % kBytesPerFrame);
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SrcBufferCopy.AddZeroed(FrameDiff);
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FramesToEncode++;
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}
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check(QualityInfo.NumChannels <= MAX_uint8);
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check(FramesToEncode <= MAX_uint16);
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SerializeHeaderData(CompressedData, kOpusSampleRate, TrueSampleCount, QualityInfo.NumChannels, FramesToEncode);
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// Temporary storage with more than enough to store any compressed frame
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TArray<uint8> TempCompressedData;
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TempCompressedData.AddUninitialized(kBytesPerFrame);
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while (SrcBufferOffset < SrcBufferCopy.Num())
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{
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int32 CompressedLength = opus_encode(Encoder, (const opus_int16*)(SrcBufferCopy.GetData() + SrcBufferOffset), kOpusFrameSizeSamples, TempCompressedData.GetData(), TempCompressedData.Num());
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if (CompressedLength < 0)
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{
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const char* ErrorStr = opus_strerror(CompressedLength);
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UE_LOG(LogAudio, Warning, TEXT("Failed to encode: [%d] %s"), CompressedLength, ANSI_TO_TCHAR(ErrorStr));
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Destroy(Encoder);
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CompressedDataStore.Empty();
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return false;
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}
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else
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{
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// Store frame length and copy compressed data before incrementing pointers
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check(CompressedLength < MAX_uint16);
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SerialiseFrameData(CompressedData, TempCompressedData.GetData(), CompressedLength);
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SrcBufferOffset += kBytesPerFrame;
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}
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}
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Destroy(Encoder);
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return CompressedDataStore.Num() > 0;
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}
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virtual bool CookSurround(FName Format, const TArray<TArray<uint8> >& SrcBuffers, FSoundQualityInfo& QualityInfo, TArray<uint8>& CompressedDataStore) const override
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{
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TRACE_CPUPROFILER_EVENT_SCOPE(FAudioFormatOpus::CookSurround);
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check(Format == NAME_OPUS);
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// Get best compatible sample rate
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const uint16 kOpusSampleRate = GetBestOutputSampleRate(QualityInfo.SampleRate);
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// Frame size must be one of 2.5, 5, 10, 20, 40 or 60 ms
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const int32 kOpusFrameSizeMs = 60;
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// Calculate frame size required by Opus
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const int32 kOpusFrameSizeSamples = (kOpusSampleRate * kOpusFrameSizeMs) / 1000;
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const uint32 kSampleStride = SAMPLE_SIZE * QualityInfo.NumChannels;
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const int32 kBytesPerFrame = kOpusFrameSizeSamples * kSampleStride;
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// Check whether source has compatible sample rate
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TArray<TArray<uint8>> SrcBufferCopies;
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if (QualityInfo.SampleRate != kOpusSampleRate)
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{
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for (int32 Index = 0; Index < SrcBuffers.Num(); Index++)
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{
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TArray<uint8>& NewCopy = *new (SrcBufferCopies) TArray<uint8>;
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if (!ResamplePCM(1, SrcBuffers[Index], QualityInfo.SampleRate, NewCopy, kOpusSampleRate))
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{
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return false;
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}
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}
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}
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else
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{
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SrcBufferCopies.Reset();
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SrcBufferCopies.AddDefaulted(SrcBuffers.Num());
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// Take a copy of the source regardless
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for (int32 Index = 0; Index < SrcBuffers.Num(); Index++)
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{
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SrcBufferCopies[Index] = SrcBuffers[Index];
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}
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}
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// Ensure that all channels are the same length
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int32 SourceSize = -1;
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for (int32 Index = 0; Index < SrcBufferCopies.Num(); Index++)
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{
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if (!Index)
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{
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SourceSize = SrcBufferCopies[Index].Num();
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}
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else
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{
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if (SourceSize != SrcBufferCopies[Index].Num())
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{
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return false;
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}
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}
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}
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if (SourceSize <= 0)
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{
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return false;
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}
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// Initialise the Opus multistream encoder
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OpusMSEncoder* Encoder = NULL;
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int32 EncError = 0;
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int32 streams = 0;
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int32 coupled_streams = 0;
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// mapping_family not documented but figured out: 0 = 1 or 2 channels, 1 = 1 to 8 channel surround sound, 255 = up to 255 channels with no surround processing
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int32 mapping_family = 1;
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TArray<uint8> mapping;
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mapping.AddUninitialized(QualityInfo.NumChannels);
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#if USE_UE_MEM_ALLOC
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int32 EncSize = opus_multistream_surround_encoder_get_size(QualityInfo.NumChannels, mapping_family);
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Encoder = (OpusMSEncoder*)FMemory::Malloc(EncSize);
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EncError = opus_multistream_surround_encoder_init(Encoder, kOpusSampleRate, QualityInfo.NumChannels, mapping_family, &streams, &coupled_streams, mapping.GetData(), OPUS_APPLICATION_AUDIO);
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#else
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Encoder = opus_multistream_surround_encoder_create(kOpusSampleRate, QualityInfo.NumChannels, mapping_family, &streams, &coupled_streams, mapping.GetData(), OPUS_APPLICATION_AUDIO, &EncError);
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#endif
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if (EncError != OPUS_OK)
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{
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Destroy(Encoder);
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return false;
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}
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int32 BitRate = GetBitRateFromQuality(QualityInfo);
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opus_multistream_encoder_ctl(Encoder, OPUS_SET_BITRATE(BitRate));
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// Create a buffer to store compressed data
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CompressedDataStore.Empty();
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FMemoryWriter CompressedData(CompressedDataStore);
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int32 SrcBufferOffset = 0;
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// Calc frame and sample count
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int32 FramesToEncode = SourceSize / (kOpusFrameSizeSamples * SAMPLE_SIZE);
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uint32 TrueSampleCount = SourceSize / SAMPLE_SIZE;
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// Add another frame if Source does not divide into an equal number of frames
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if (SourceSize % (kOpusFrameSizeSamples * SAMPLE_SIZE) != 0)
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{
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FramesToEncode++;
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}
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check(QualityInfo.NumChannels <= MAX_uint8);
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check(FramesToEncode <= MAX_uint16);
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SerializeHeaderData(CompressedData, kOpusSampleRate, TrueSampleCount, QualityInfo.NumChannels, FramesToEncode);
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// Temporary storage for source data in an interleaved format
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TArray<uint8> TempInterleavedSrc;
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TempInterleavedSrc.AddUninitialized(kBytesPerFrame);
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// Temporary storage with more than enough to store any compressed frame
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TArray<uint8> TempCompressedData;
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TempCompressedData.AddUninitialized(kBytesPerFrame);
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while (SrcBufferOffset < SourceSize)
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{
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// Read a frames worth of data from the source and pack it into interleaved temporary storage
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for (int32 SampleIndex = 0; SampleIndex < kOpusFrameSizeSamples; ++SampleIndex)
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{
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int32 CurrSrcOffset = SrcBufferOffset + SampleIndex*SAMPLE_SIZE;
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int32 CurrInterleavedOffset = SampleIndex*kSampleStride;
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if (CurrSrcOffset < SourceSize)
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{
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check(QualityInfo.NumChannels <= 8); // Static analysis fix: warning C6385: Reading invalid data from 'Order': the readable size is '256' bytes, but '8160' bytes may be read.
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for (uint32 ChannelIndex = 0; ChannelIndex < QualityInfo.NumChannels; ++ChannelIndex)
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{
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// Interleave the channels in the Vorbis format, so that the correct channel is used for LFE
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int32 OrderedChannelIndex = VorbisChannelInfo::Order[QualityInfo.NumChannels - 1][ChannelIndex];
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int32 CurrInterleavedIndex = CurrInterleavedOffset + ChannelIndex*SAMPLE_SIZE;
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// Copy both bytes that make up a single sample
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TempInterleavedSrc[CurrInterleavedIndex] = SrcBufferCopies[OrderedChannelIndex][CurrSrcOffset];
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TempInterleavedSrc[CurrInterleavedIndex + 1] = SrcBufferCopies[OrderedChannelIndex][CurrSrcOffset + 1];
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}
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}
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else
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{
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// Zero the rest of the temp buffer to make it an exact frame
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FMemory::Memzero(TempInterleavedSrc.GetData() + CurrInterleavedOffset, kBytesPerFrame - CurrInterleavedOffset);
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SampleIndex = kOpusFrameSizeSamples;
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}
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}
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int32 CompressedLength = opus_multistream_encode(Encoder, (const opus_int16*)(TempInterleavedSrc.GetData()), kOpusFrameSizeSamples, TempCompressedData.GetData(), TempCompressedData.Num());
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if (CompressedLength < 0)
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{
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const char* ErrorStr = opus_strerror(CompressedLength);
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UE_LOG(LogAudio, Warning, TEXT("Failed to encode: [%d] %s"), CompressedLength, ANSI_TO_TCHAR(ErrorStr));
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Destroy(Encoder);
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CompressedDataStore.Empty();
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return false;
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}
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else
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{
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// Store frame length and copy compressed data before incrementing pointers
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check(CompressedLength < MAX_uint16);
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SerialiseFrameData(CompressedData, TempCompressedData.GetData(), CompressedLength);
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SrcBufferOffset += kOpusFrameSizeSamples * SAMPLE_SIZE;
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}
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}
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Destroy(Encoder);
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return CompressedDataStore.Num() > 0;
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}
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virtual int32 Recompress(FName Format, const TArray<uint8>& SrcBuffer, FSoundQualityInfo& QualityInfo, TArray<uint8>& OutBuffer) const override
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{
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check(Format == NAME_OPUS);
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FOpusAudioInfo AudioInfo;
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// Cannot quality preview multichannel sounds
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if( QualityInfo.NumChannels > 2 )
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{
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return 0;
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}
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TArray<uint8> CompressedDataStore;
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if( !Cook( Format, SrcBuffer, QualityInfo, CompressedDataStore ) )
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{
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return 0;
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}
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// Parse the opus header for the relevant information
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if( !AudioInfo.ReadCompressedInfo( CompressedDataStore.GetData(), CompressedDataStore.Num(), &QualityInfo ) )
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{
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return 0;
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}
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// Decompress all the sample data
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OutBuffer.Empty(QualityInfo.SampleDataSize);
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OutBuffer.AddZeroed(QualityInfo.SampleDataSize);
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AudioInfo.ExpandFile( OutBuffer.GetData(), &QualityInfo );
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return CompressedDataStore.Num();
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}
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virtual int32 GetMinimumSizeForInitialChunk(FName Format, const TArray<uint8>& SrcBuffer) const override
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{
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// Since UE uses it's own version of the header, we hardcode the size of that here. See SplitDataForStreaming below to see our initial info.
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return FCStringAnsi::Strlen(OPUS_ID_STRING) + 1 // Format identifier
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+ sizeof(uint16) // Sample Rate
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+ sizeof(uint32) // True Sample Count
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+ sizeof(uint8) // Number of Channels
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+ sizeof(uint16); // Serialized Frames
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}
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virtual bool SplitDataForStreaming(const TArray<uint8>& SrcBuffer, TArray<TArray<uint8>>& OutBuffers, const int32 MaxInitialChunkSize, const int32 MaxChunkSize) const override
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{
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if (SrcBuffer.Num() == 0)
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{
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return false;
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}
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uint32 ReadOffset = 0;
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uint32 WriteOffset = 0;
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uint16 ProcessedFrames = 0;
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const uint8* LockedSrc = SrcBuffer.GetData();
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// Read Identifier, True Sample Count, Number of channels and Frames to Encode first
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if (FCStringAnsi::Strcmp((char*)LockedSrc, OPUS_ID_STRING) != 0)
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{
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return false;
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}
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ReadOffset += FCStringAnsi::Strlen(OPUS_ID_STRING) + 1;
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uint16 SampleRate = *((uint16*)(LockedSrc + ReadOffset));
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ReadOffset += sizeof(uint16);
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uint32 TrueSampleCount = *((uint32*)(LockedSrc + ReadOffset));
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ReadOffset += sizeof(uint32);
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uint8 NumChannels = *(LockedSrc + ReadOffset);
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ReadOffset += sizeof(uint8);
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uint16 SerializedFrames = *((uint16*)(LockedSrc + ReadOffset));
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ReadOffset += sizeof(uint16);
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// Should always be able to store basic info in a single chunk
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check(ReadOffset - WriteOffset <= (uint32)MaxInitialChunkSize);
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int32 ChunkSize = MaxInitialChunkSize;
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while (ProcessedFrames < SerializedFrames)
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{
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uint16 FrameSize = *((uint16*)(LockedSrc + ReadOffset));
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if ( (ReadOffset + sizeof(uint16) + FrameSize) - WriteOffset >= ChunkSize)
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{
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WriteOffset += AddDataChunk(OutBuffers, LockedSrc + WriteOffset, ReadOffset - WriteOffset);
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}
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ReadOffset += sizeof(uint16) + FrameSize;
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ProcessedFrames++;
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ChunkSize = MaxChunkSize;
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}
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if (WriteOffset < ReadOffset)
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{
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WriteOffset += AddDataChunk(OutBuffers, LockedSrc + WriteOffset, ReadOffset - WriteOffset);
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}
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return true;
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}
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/**
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* Calculate the best sample rate for the output opus data
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*/
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static uint16 GetBestOutputSampleRate(int32 SampleRate)
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{
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static const uint16 ValidSampleRates[] =
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{
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0, // not really valid, but simplifies logic below
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8000,
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12000,
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16000,
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24000,
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48000,
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};
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// look for the next highest valid rate
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for (int32 Index = UE_ARRAY_COUNT(ValidSampleRates) - 2; Index >= 0; Index--)
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{
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if (SampleRate > ValidSampleRates[Index])
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{
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return ValidSampleRates[Index + 1];
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}
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}
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// this should never get here!
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check(0);
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return 0;
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}
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bool ResamplePCM(uint32 NumChannels, const TArray<uint8>& InBuffer, uint32 InSampleRate, TArray<uint8>& OutBuffer, uint32 OutSampleRate) const
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{
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// Initialize resampler to convert to desired rate for Opus
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int32 err = 0;
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SpeexResamplerState* resampler = speex_resampler_init(NumChannels, InSampleRate, OutSampleRate, SPEEX_RESAMPLER_QUALITY_DESKTOP, &err);
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if (err != RESAMPLER_ERR_SUCCESS)
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{
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speex_resampler_destroy(resampler);
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return false;
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}
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// Calculate extra space required for sample rate
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const uint32 SampleStride = SAMPLE_SIZE * NumChannels;
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const float Duration = (float)InBuffer.Num() / (InSampleRate * SampleStride);
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const int32 SafeCopySize = (Duration + 1) * OutSampleRate * SampleStride;
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OutBuffer.Empty(SafeCopySize);
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OutBuffer.AddUninitialized(SafeCopySize);
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uint32 InSamples = InBuffer.Num() / SampleStride;
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uint32 OutSamples = OutBuffer.Num() / SampleStride;
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// Do resampling and check results
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if (NumChannels == 1)
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{
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err = speex_resampler_process_int(resampler, 0, (const short*)(InBuffer.GetData()), &InSamples, (short*)(OutBuffer.GetData()), &OutSamples);
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}
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else
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{
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err = speex_resampler_process_interleaved_int(resampler, (const short*)(InBuffer.GetData()), &InSamples, (short*)(OutBuffer.GetData()), &OutSamples);
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}
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speex_resampler_destroy(resampler);
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|
if (err != RESAMPLER_ERR_SUCCESS)
|
|
{
|
|
return false;
|
|
}
|
|
|
|
// reduce the size of Out Buffer if more space than necessary was allocated
|
|
const int32 WrittenBytes = (int32)(OutSamples * SampleStride);
|
|
if (WrittenBytes < OutBuffer.Num())
|
|
{
|
|
OutBuffer.SetNum(WrittenBytes, true);
|
|
}
|
|
|
|
return true;
|
|
}
|
|
|
|
int32 GetBitRateFromQuality(FSoundQualityInfo& QualityInfo) const
|
|
{
|
|
// There is no perfect way to map Vorbis' Quality setting to an Opus bitrate but this
|
|
// will use it as a multiplier to decide how much smaller than the original the
|
|
// compressed data should be
|
|
int32 OriginalBitRate = QualityInfo.SampleRate * QualityInfo.NumChannels * SAMPLE_SIZE * 8;
|
|
return (float)OriginalBitRate * FMath::GetMappedRangeValueClamped(FVector2f(1, 100), FVector2f(0.04, 0.25), (float)QualityInfo.Quality);
|
|
}
|
|
|
|
void SerializeHeaderData(FMemoryWriter& CompressedData, uint16 SampleRate, uint32 TrueSampleCount, uint8 NumChannels, uint16 NumFrames) const
|
|
{
|
|
const char* OpusIdentifier = OPUS_ID_STRING;
|
|
CompressedData.Serialize((void*)OpusIdentifier, FCStringAnsi::Strlen(OpusIdentifier) + 1);
|
|
CompressedData.Serialize(&SampleRate, sizeof(uint16));
|
|
CompressedData.Serialize(&TrueSampleCount, sizeof(uint32));
|
|
CompressedData.Serialize(&NumChannels, sizeof(uint8));
|
|
CompressedData.Serialize(&NumFrames, sizeof(uint16));
|
|
}
|
|
|
|
void SerialiseFrameData(FMemoryWriter& CompressedData, uint8* FrameData, uint16 FrameSize) const
|
|
{
|
|
CompressedData.Serialize(&FrameSize, sizeof(uint16));
|
|
CompressedData.Serialize(FrameData, FrameSize);
|
|
}
|
|
|
|
void Destroy(OpusEncoder* Encoder) const
|
|
{
|
|
#if USE_UE_MEM_ALLOC
|
|
FMemory::Free(Encoder);
|
|
#else
|
|
opus_encoder_destroy(Encoder);
|
|
#endif
|
|
}
|
|
|
|
void Destroy(OpusMSEncoder* Encoder) const
|
|
{
|
|
#if USE_UE_MEM_ALLOC
|
|
FMemory::Free(Encoder);
|
|
#else
|
|
opus_multistream_encoder_destroy(Encoder);
|
|
#endif
|
|
}
|
|
|
|
/**
|
|
* Adds a new chunk of data to the array
|
|
*
|
|
* @param OutBuffers Array of buffers to add to
|
|
* @param ChunkData Pointer to chunk data
|
|
* @param ChunkSize How much data to write
|
|
* @return How many bytes were written
|
|
*/
|
|
int32 AddDataChunk(TArray<TArray<uint8>>& OutBuffers, const uint8* ChunkData, int32 ChunkSize) const
|
|
{
|
|
TArray<uint8>& NewBuffer = *new (OutBuffers) TArray<uint8>;
|
|
NewBuffer.Empty(ChunkSize);
|
|
NewBuffer.AddUninitialized(ChunkSize);
|
|
FMemory::Memcpy(NewBuffer.GetData(), ChunkData, ChunkSize);
|
|
return ChunkSize;
|
|
}
|
|
};
|
|
|
|
|
|
/**
|
|
* Module for opus audio compression
|
|
*/
|
|
|
|
static IAudioFormat* Singleton = NULL;
|
|
|
|
class FAudioPlatformOpusModule : public IAudioFormatModule
|
|
{
|
|
public:
|
|
virtual ~FAudioPlatformOpusModule()
|
|
{
|
|
delete Singleton;
|
|
Singleton = NULL;
|
|
}
|
|
virtual IAudioFormat* GetAudioFormat()
|
|
{
|
|
if (!Singleton)
|
|
{
|
|
Singleton = new FAudioFormatOpus();
|
|
}
|
|
return Singleton;
|
|
}
|
|
};
|
|
|
|
IMPLEMENT_MODULE( FAudioPlatformOpusModule, AudioFormatOpus);
|