BansheeEngine/Source/BansheeOpenAudio/BsOggVorbisEncoder.cpp

#include "BsOggVorbisEncoder.h"
#include "FileSystem/BsDataStream.h"
#include "Audio/BsAudioUtility.h"

namespace bs
{
// Writes to the internal cached buffer and flushes it if needed
#define WRITE_TO_BUFFER(data, length)					\
	if ((mBufferOffset + length) > BUFFER_SIZE)			\
		flush();										\
														\
	if(length > BUFFER_SIZE)							\
		mWriteCallback(data, length);					\
	else												\
	{													\
		memcpy(mBuffer + mBufferOffset, data, length);	\
		mBufferOffset += length;						\
	}

	OggVorbisEncoder::OggVorbisEncoder()
		:mBufferOffset(0), mNumChannels(0), mBitDepth(0), mClosed(true)
	{ }

	OggVorbisEncoder::~OggVorbisEncoder()
	{
		close();
	}

	bool OggVorbisEncoder::open(std::function<void(UINT8*, UINT32)> writeCallback, UINT32 sampleRate, UINT32 bitDepth, 
		UINT32 numChannels)
	{
		mNumChannels = numChannels;
		mBitDepth = bitDepth;
		mWriteCallback = writeCallback;
		mClosed = false;

		ogg_stream_init(&mOggState, std::rand());
		vorbis_info_init(&mVorbisInfo);

		// Automatic bitrate management with quality 0.4 (~128 kbps for 44 KHz stereo sound)
		INT32 status = vorbis_encode_init_vbr(&mVorbisInfo, numChannels, sampleRate, 0.4f);
		if (status != 0)
		{
			LOGERR("Failed to write Ogg Vorbis file.");
			close();
			return false;
		}

		vorbis_analysis_init(&mVorbisState, &mVorbisInfo);
		vorbis_block_init(&mVorbisState, &mVorbisBlock);

		// Generate header
		vorbis_comment comment;
		vorbis_comment_init(&comment);

		ogg_packet headerPacket, commentPacket, codePacket;
		status = vorbis_analysis_headerout(&mVorbisState, &comment, &headerPacket, &commentPacket, &codePacket);
		vorbis_comment_clear(&comment);

		if (status != 0)
		{
			LOGERR("Failed to write Ogg Vorbis file.");
			close();
			return false;
		}

		// Write header
		ogg_stream_packetin(&mOggState, &headerPacket);
		ogg_stream_packetin(&mOggState, &commentPacket);
		ogg_stream_packetin(&mOggState, &codePacket);

		ogg_page page;
		while (ogg_stream_flush(&mOggState, &page) > 0)
		{
			WRITE_TO_BUFFER(page.header, page.header_len);
			WRITE_TO_BUFFER(page.body, page.body_len);
		}

		return true;
	}

	void OggVorbisEncoder::write(UINT8* samples, UINT32 numSamples)
	{
		static const UINT32 WRITE_LENGTH = 1024;

		UINT32 numFrames = numSamples / mNumChannels;
		while (numFrames > 0)
		{
			UINT32 numFramesToWrite = std::min(numFrames, WRITE_LENGTH);
			float** buffer = vorbis_analysis_buffer(&mVorbisState, numFramesToWrite);

			if (mBitDepth == 8)
			{
				for (UINT32 i = 0; i < numFramesToWrite; i++)
				{
					for (UINT32 j = 0; j < mNumChannels; j++)
					{
						INT8 sample = *(INT8*)samples;
						float encodedSample = sample / 127.0f;
						buffer[j][i] = encodedSample;

						samples++;
					}
				}
			}
			else if (mBitDepth == 16)
			{
				for (UINT32 i = 0; i < numFramesToWrite; i++)
				{
					for (UINT32 j = 0; j < mNumChannels; j++)
					{
						INT16 sample = *(INT16*)samples;
						float encodedSample = sample / 32767.0f;
						buffer[j][i] = encodedSample;

						samples += 2;
					}
				}
			}
			else if (mBitDepth == 24)
			{
				for (UINT32 i = 0; i < numFramesToWrite; i++)
				{
					for (UINT32 j = 0; j < mNumChannels; j++)
					{
						INT32 sample = AudioUtility::convert24To32Bits(samples);
						float encodedSample = sample / 2147483647.0f;
						buffer[j][i] = encodedSample;

						samples += 3;
					}
				}
			}
			else if (mBitDepth == 32)
			{
				for (UINT32 i = 0; i < numFramesToWrite; i++)
				{
					for (UINT32 j = 0; j < mNumChannels; j++)
					{
						INT32 sample = *(INT32*)samples;
						float encodedSample = sample / 2147483647.0f;
						buffer[j][i] = encodedSample;

						samples += 4;
					}
				}
			}
			else
				assert(false);

			// Signal how many frames were written
			vorbis_analysis_wrote(&mVorbisState, numFramesToWrite);
			writeBlocks();

			numFrames -= numFramesToWrite;
		}
	}

	void OggVorbisEncoder::writeBlocks()
	{
		while (vorbis_analysis_blockout(&mVorbisState, &mVorbisBlock) == 1)
		{
			// Analyze and determine optimal bitrate
			vorbis_analysis(&mVorbisBlock, nullptr);
			vorbis_bitrate_addblock(&mVorbisBlock);

			// Write block into ogg packets
			ogg_packet packet;
			while (vorbis_bitrate_flushpacket(&mVorbisState, &packet))
			{
				ogg_stream_packetin(&mOggState, &packet);

				// If new page, write it to the internal buffer
				ogg_page page;
				while (ogg_stream_flush(&mOggState, &page) > 0)
				{
					WRITE_TO_BUFFER(page.header, page.header_len);
					WRITE_TO_BUFFER(page.body, page.body_len);
				}
			}
		}
	}

	void OggVorbisEncoder::flush()
	{
		if (mBufferOffset > 0 && mWriteCallback != nullptr)
			mWriteCallback(mBuffer, mBufferOffset);

		mBufferOffset = 0;
	}

	void OggVorbisEncoder::close()
	{
		if (mClosed)
			return;

		// Mark end of data and flush any remaining data in the buffers
		vorbis_analysis_wrote(&mVorbisState, 0);
		writeBlocks();
		flush();

		ogg_stream_clear(&mOggState);
		vorbis_block_clear(&mVorbisBlock);
		vorbis_dsp_clear(&mVorbisState);
		vorbis_info_clear(&mVorbisInfo);

		mClosed = true;
	}

	UINT8* OggVorbisEncoder::PCMToOggVorbis(UINT8* samples, const AudioDataInfo& info, UINT32& size)
	{
		struct EncodedBlock
		{
			UINT8* data;
			UINT32 size;
		};

		Vector<EncodedBlock> blocks;
		UINT32 totalEncodedSize = 0;
		auto writeCallback = [&](UINT8* buffer, UINT32 size)
		{
			EncodedBlock newBlock;
			newBlock.data = bs_frame_alloc(size);
			newBlock.size = size;

			memcpy(newBlock.data, buffer, size);
			blocks.push_back(newBlock);
			totalEncodedSize += size;
		};

		bs_frame_mark();

		OggVorbisEncoder writer;
		writer.open(writeCallback, info.sampleRate, info.bitDepth, info.numChannels);

		writer.write(samples, info.numSamples);
		writer.close();

		UINT8* outSampleBuffer = (UINT8*)bs_alloc(totalEncodedSize);
		UINT32 offset = 0;
		for (auto& block : blocks)
		{
			memcpy(outSampleBuffer + offset, block.data, block.size);
			offset += block.size;

			bs_frame_free(block.data);
		}

		bs_frame_clear();
		
		size = totalEncodedSize;
		return outSampleBuffer;
	}

#undef WRITE_TO_BUFFER
}