166 lines
4.4 KiB
C++
166 lines
4.4 KiB
C++
/********************************************************************************
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Copyright (C) 2012 Hugh Bailey <obs.jim@gmail.com>
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This program is free software; you can redistribute it and/or modify
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it under the terms of the GNU General Public License as published by
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the Free Software Foundation; either version 2 of the License, or
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(at your option) any later version.
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This program is distributed in the hope that it will be useful,
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but WITHOUT ANY WARRANTY; without even the implied warranty of
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MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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GNU General Public License for more details.
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You should have received a copy of the GNU General Public License
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along with this program; if not, write to the Free Software
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Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307, USA.
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********************************************************************************/
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#include "Main.h"
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#include "../lame/include/lame.h"
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const int audioBlockSize = 4; //output is 2 16bit channels
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struct AACDataPacket
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{
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List<BYTE> Packet;
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inline void FreeData() {Packet.Clear();}
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};
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//lame is not lame.. it's godly.
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class MP3Encoder : public AudioEncoder
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{
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lame_global_flags *lgf;
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List<BYTE> MP3OutputBuffer;
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List<BYTE> header;
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DWORD dwMP3MaxSize;
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bool bFirstPacket;
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UINT outputFrameSize;
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UINT curBitRate;
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List<QWORD> bufferedTimestamps;
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QWORD curEncodeTimestamp;
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DWORD frameCounter;
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bool bFirstFrame;
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public:
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MP3Encoder(UINT bitRate)
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{
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curBitRate = bitRate;
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lgf = lame_init();
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if(!lgf)
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CrashError(TEXT("Unable to open mp3 encoder"));
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lame_set_in_samplerate(lgf, App->GetSampleRateHz());
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lame_set_out_samplerate(lgf, App->GetSampleRateHz());
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lame_set_num_channels(lgf, App->NumAudioChannels());
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lame_set_disable_reservoir(lgf, TRUE); //bit reservoir has to be disabled for seamless streaming
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lame_set_quality(lgf, 2);
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lame_set_VBR(lgf, vbr_off);
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lame_set_brate(lgf, bitRate);
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lame_init_params(lgf);
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outputFrameSize = lame_get_framesize(lgf); //1152 usually
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dwMP3MaxSize = DWORD(1.25*double(outputFrameSize*audioBlockSize) + 7200.0);
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MP3OutputBuffer.SetSize(dwMP3MaxSize+1);
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MP3OutputBuffer[0] = 0x2f;
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bFirstPacket = true;
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Log(TEXT("------------------------------------------"));
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Log(TEXT("%s"), GetInfoString().Array());
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}
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~MP3Encoder()
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{
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lame_close(lgf);
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}
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bool Encode(float *input, UINT numInputFrames, DataPacket &packet, QWORD ×tamp)
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{
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if(bFirstFrame)
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{
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curEncodeTimestamp = timestamp;
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bFirstFrame = false;
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}
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//------------------------------------------------
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UINT lastSampleSize = frameCounter;
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frameCounter += numInputFrames;
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if(frameCounter > outputFrameSize)
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{
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frameCounter -= outputFrameSize;
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bufferedTimestamps << curEncodeTimestamp;
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curEncodeTimestamp = timestamp + ((outputFrameSize-lastSampleSize)*1000/App->GetSampleRateHz());
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}
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int ret = lame_encode_buffer_interleaved_ieee_float(lgf, (float*)input, numInputFrames, MP3OutputBuffer.Array()+1, dwMP3MaxSize);
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if(ret < 0)
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{
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AppWarning(TEXT("MP3 encode failed"));
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return false;
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}
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if(ret > 0)
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{
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if(bFirstPacket)
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{
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header.CopyArray(MP3OutputBuffer.Array(), ret);
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bFirstPacket = false;
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ret = 0;
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}
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else
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{
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packet.lpPacket = MP3OutputBuffer.Array();
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packet.size = ret+1;
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timestamp = bufferedTimestamps[0];
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bufferedTimestamps.Remove(0);
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}
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}
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return ret > 0;
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}
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UINT GetFrameSize() const
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{
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return outputFrameSize;
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}
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void GetHeaders(DataPacket &packet)
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{
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packet.lpPacket = header.Array();
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packet.size = header.Num();
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}
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int GetBitRate() const {return curBitRate;}
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CTSTR GetCodec() const {return TEXT("MP3");}
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String GetInfoString() const
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{
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String strInfo;
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strInfo << TEXT("Audio Encoding: MP3") <<
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TEXT("\r\n bitrate: ") << IntString(curBitRate);
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return strInfo;
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}
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};
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AudioEncoder* CreateMP3Encoder(UINT bitRate)
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{
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return new MP3Encoder(bitRate);
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}
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