ppsspp/Core/HW/SimpleAudioDec.h
kaienfr e25bc6908a Fix nonsmooth sound issue and incomplete frame at the end of source and pcm buffers.
using a std::string to save the source buffer, and changing the loading strategy so as we will not get any incomplete frame.
2014-04-12 23:16:38 +02:00

192 lines
4.6 KiB
C++

// Copyright (c) 2013- PPSSPP Project.
// This program is free software: you can redistribute it and/or modify
// it under the terms of the GNU General Public License as published by
// the Free Software Foundation, version 2.0 or later versions.
// This program is distributed in the hope that it will be useful,
// but WITHOUT ANY WARRANTY; without even the implied warranty of
// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
// GNU General Public License 2.0 for more details.
// A copy of the GPL 2.0 should have been included with the program.
// If not, see http://www.gnu.org/licenses/
// Official git repository and contact information can be found at
// https://github.com/hrydgard/ppsspp and http://www.ppsspp.org/.
#pragma once
#include <cmath>
#include "base/basictypes.h"
#include "Core/HW/MediaEngine.h"
#include "Core/HLE/sceAudio.h"
#ifdef USE_FFMPEG
extern "C" {
#include <libavformat/avformat.h>
#include <libswresample/swresample.h>
#include <libavutil/samplefmt.h>
}
#endif // USE_FFMPEG
// Wraps FFMPEG in a nice interface that's drop-in compatible with
// the old one. Decodes packet by packet - does NOT demux. That's done by
// MpegDemux. Only decodes Atrac3+, not regular Atrac3.
// Based on http://ffmpeg.org/doxygen/trunk/doc_2examples_2decoding_encoding_8c-example.html#_a13
// Ideally, Maxim's Atrac3+ decoder would be available as a standalone library
// that we could link, as that would be totally sufficient for the use case here.
// However, it will be maintained as a part of FFMPEG so that's the way we'll go
// for simplicity and sanity.
struct SimpleAudio {
public:
SimpleAudio(int);
SimpleAudio(u32, int);
~SimpleAudio();
bool Decode(void* inbuf, int inbytes, uint8_t *outbuf, int *outbytes);
bool IsOK() const { return codec_ != 0; }
int getOutSamples();
int getSourcePos();
bool ResetCodecCtx(int channels, int samplerate);
void setResampleFrequency(int freq);
u32 ctxPtr;
int audioType;
int outSamples; // output samples per frame
int srcPos; // bytes consumed in source during the last decoding
int wanted_resample_freq; // wanted resampling rate/frequency
private:
#ifdef USE_FFMPEG
AVFrame *frame_;
AVCodec *codec_;
AVCodecContext *codecCtx_;
SwrContext *swrCtx_;
AVCodecID audioCodecId; // AV_CODEC_ID_XXX
bool GetAudioCodecID(int audioType); // Get audioCodecId from audioType
#endif // USE_FFMPEG
};
// audioType
enum {
PSP_CODEC_AT3PLUS = 0x00001000,
PSP_CODEC_AT3 = 0x00001001,
PSP_CODEC_MP3 = 0x00001002,
PSP_CODEC_AAC = 0x00001003,
};
static const char *const codecNames[4] = {
"AT3+", "AT3", "MP3", "AAC",
};
void AudioClose(SimpleAudio **ctx);
static const char *GetCodecName(int codec) {
if (codec >= PSP_CODEC_AT3PLUS && codec <= PSP_CODEC_AAC) {
return codecNames[codec - PSP_CODEC_AT3PLUS];
}
else {
return "(unk)";
}
};
bool isValidCodec(int codec);
class AuCtx{
public:
// Au source informations
u64 startPos;
u64 endPos;
u32 AuBuf;
u32 AuBufSize;
u32 PCMBuf;
u32 PCMBufSize;
int freq;
int BitRate;
int SamplingRate;
int Channels;
int Version;
// audio settings
u32 SumDecodedSamples;
int LoopNum;
u32 MaxOutputSample;
// Au decoder
SimpleAudio *decoder;
// Au type
int audioType;
// buffers informations
int AuBufAvailable; // the available buffer of AuBuf to be able to recharge data
int readPos; // read position in audio source file
int askedReadSize; // the size of data requied to be read from file by the game
int realReadSize; // the really read size from file
std::string sourcebuff; // source buffer
AuCtx() :decoder(NULL){};
~AuCtx(){
if (decoder){
AudioClose(&decoder);
decoder = NULL;
}
};
u32 AuExit();
u32 AuDecode(u32 pcmAddr);
u32 AuGetLoopNum();
u32 AuSetLoopNum(int loop);
int AuCheckStreamDataNeeded();
u32 AuNotifyAddStreamData(int size);
u32 AuGetInfoToAddStreamData(u32 buff, u32 size, u32 srcPos);
u32 AuGetMaxOutputSample();
u32 AuGetSumDecodedSample();
u32 AuResetPlayPosition();
int AuGetChannelNum();
int AuGetBitRate();
int AuGetSamplingRate();
u32 AuResetPlayPositionByFrame(int position);
int AuGetVersion();
void DoState(PointerWrap &p) {
auto s = p.Section("AuContext", 0, 1);
if (!s)
return;
p.Do(startPos);
p.Do(endPos);
p.Do(AuBuf);
p.Do(AuBufSize);
p.Do(PCMBuf);
p.Do(PCMBufSize);
p.Do(freq);
p.Do(SumDecodedSamples);
p.Do(LoopNum);
p.Do(Channels);
p.Do(MaxOutputSample);
p.Do(readPos);
p.Do(audioType);
p.Do(BitRate);
p.Do(SamplingRate);
p.Do(askedReadSize);
p.Do(realReadSize);
if (p.mode == p.MODE_READ){
decoder = new SimpleAudio(audioType);
AuBufAvailable = 0; // reset to read from file at position readPos
}
};
};