mirror of
https://github.com/hrydgard/ppsspp.git
synced 2024-11-27 07:20:49 +00:00
580 lines
16 KiB
C++
580 lines
16 KiB
C++
#include <string>
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#include <mutex>
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#include "ext/minimp3/minimp3_ex.h"
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#include "Common/File/VFS/VFS.h"
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#include "Common/UI/Root.h"
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#include "Common/Data/Text/I18n.h"
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#include "Common/CommonTypes.h"
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#include "Common/Data/Format/RIFF.h"
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#include "Common/Log.h"
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#include "Common/System/System.h"
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#include "Common/System/OSD.h"
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#include "Common/Serialize/SerializeFuncs.h"
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#include "Common/TimeUtil.h"
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#include "Common/Data/Collections/FixedSizeQueue.h"
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#include "Core/HW/SimpleAudioDec.h"
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#include "Core/HLE/__sceAudio.h"
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#include "Core/System.h"
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#include "Core/Config.h"
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#include "UI/GameInfoCache.h"
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#include "UI/BackgroundAudio.h"
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struct WavData {
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int num_channels = -1;
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int sample_rate = -1;
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int numFrames = -1;
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int samplesPerSec = -1;
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int avgBytesPerSec = -1;
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int raw_offset_loop_start = 0;
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int raw_offset_loop_end = 0;
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int loop_start_offset = 0;
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int loop_end_offset = 0;
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int codec = 0;
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int raw_bytes_per_frame = 0;
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uint8_t *raw_data = nullptr;
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int raw_data_size = 0;
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u8 at3_extradata[16];
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bool Read(RIFFReader &riff);
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~WavData() {
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free(raw_data);
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raw_data = nullptr;
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}
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bool IsSimpleWAV() const {
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bool isBad = raw_bytes_per_frame > sizeof(int16_t) * num_channels;
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return !isBad && num_channels > 0 && sample_rate >= 8000 && codec == 0;
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}
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};
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bool WavData::Read(RIFFReader &file_) {
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// If we have no loop start info, we'll just loop the entire audio.
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raw_offset_loop_start = 0;
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raw_offset_loop_end = 0;
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if (file_.Descend('RIFF')) {
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file_.ReadInt(); //get past 'WAVE'
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if (file_.Descend('fmt ')) { //enter the format chunk
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int temp = file_.ReadInt();
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int format = temp & 0xFFFF;
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switch (format) {
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case 0xFFFE:
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codec = PSP_CODEC_AT3PLUS;
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break;
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case 0x270:
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codec = PSP_CODEC_AT3;
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break;
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case 1:
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// Raw wave data, no codec
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codec = 0;
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break;
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default:
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ERROR_LOG(SCEAUDIO, "Unexpected wave format %04x", format);
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return false;
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}
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num_channels = temp >> 16;
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samplesPerSec = file_.ReadInt();
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/*avgBytesPerSec =*/ file_.ReadInt();
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temp = file_.ReadInt();
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raw_bytes_per_frame = temp & 0xFFFF;
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if (codec == PSP_CODEC_AT3) {
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// The first two bytes are actually not a useful part of the extradata.
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// We already read 16 bytes, so make sure there's enough left.
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if (file_.GetCurrentChunkSize() >= 32) {
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file_.ReadData(at3_extradata, 16);
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} else {
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memset(at3_extradata, 0, sizeof(at3_extradata));
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}
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}
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file_.Ascend();
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// INFO_LOG(AUDIO, "got fmt data: %i", samplesPerSec);
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} else {
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ERROR_LOG(AUDIO, "Error - no format chunk in wav");
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file_.Ascend();
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return false;
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}
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if (file_.Descend('smpl')) {
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std::vector<u8> smplData;
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smplData.resize(file_.GetCurrentChunkSize());
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file_.ReadData(&smplData[0], (int)smplData.size());
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int numLoops = *(int *)&smplData[28];
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struct AtracLoopInfo {
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int cuePointID;
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int type;
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int startSample;
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int endSample;
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int fraction;
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int playCount;
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};
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if (numLoops > 0 && smplData.size() >= 36 + sizeof(AtracLoopInfo) * numLoops) {
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AtracLoopInfo *loops = (AtracLoopInfo *)&smplData[36];
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int samplesPerFrame = codec == PSP_CODEC_AT3PLUS ? 2048 : 1024;
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for (int i = 0; i < numLoops; ++i) {
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// Only seen forward loops, so let's ignore others.
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if (loops[i].type != 0)
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continue;
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// We ignore loop interpolation (fraction) and play count for now.
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raw_offset_loop_start = (loops[i].startSample / samplesPerFrame) * raw_bytes_per_frame;
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loop_start_offset = loops[i].startSample % samplesPerFrame;
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raw_offset_loop_end = (loops[i].endSample / samplesPerFrame) * raw_bytes_per_frame;
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loop_end_offset = loops[i].endSample % samplesPerFrame;
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if (loops[i].playCount == 0) {
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// This was an infinite loop, so ignore the rest.
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// In practice, there's usually only one and it's usually infinite.
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break;
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}
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}
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}
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file_.Ascend();
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}
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// enter the data chunk
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if (file_.Descend('data')) {
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int numBytes = file_.GetCurrentChunkSize();
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numFrames = numBytes / raw_bytes_per_frame; // numFrames
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// It seems the atrac3 codec likes to read a little bit outside.
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const int padding = 32; // 32 is the value FFMPEG uses.
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raw_data = (uint8_t *)malloc(numBytes + padding);
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raw_data_size = numBytes;
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if (num_channels == 1 || num_channels == 2) {
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file_.ReadData(raw_data, numBytes);
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} else {
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ERROR_LOG(AUDIO, "Error - bad blockalign or channels");
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free(raw_data);
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raw_data = nullptr;
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return false;
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}
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file_.Ascend();
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} else {
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ERROR_LOG(AUDIO, "Error - no data chunk in wav");
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file_.Ascend();
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return false;
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}
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file_.Ascend();
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} else {
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ERROR_LOG(AUDIO, "Could not descend into RIFF file.");
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return false;
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}
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sample_rate = samplesPerSec;
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return true;
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}
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// Really simple looping in-memory AT3 player that also takes care of reading the file format.
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// Turns out that AT3 files used for this are modified WAVE files so fairly easy to parse.
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class AT3PlusReader {
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public:
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explicit AT3PlusReader(const std::string &data)
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: file_((const uint8_t *)&data[0], (int32_t)data.size()) {
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// Normally 8k but let's be safe.
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buffer_ = new short[32 * 1024];
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skip_next_samples_ = 0;
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wave_.Read(file_);
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uint8_t *extraData = nullptr;
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size_t extraDataSize = 0;
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size_t blockSize = 0;
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if (wave_.codec == PSP_CODEC_AT3) {
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extraData = &wave_.at3_extradata[2];
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extraDataSize = 14;
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blockSize = wave_.raw_bytes_per_frame;
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} else if (wave_.codec == PSP_CODEC_AT3PLUS) {
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blockSize = wave_.raw_bytes_per_frame;
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}
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decoder_ = CreateAudioDecoder((PSPAudioType)wave_.codec, wave_.sample_rate, wave_.num_channels, blockSize, extraData, extraDataSize);
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INFO_LOG(AUDIO, "read ATRAC, frames: %d, rate %d", wave_.numFrames, wave_.sample_rate);
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}
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~AT3PlusReader() {
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delete[] buffer_;
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buffer_ = nullptr;
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delete decoder_;
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decoder_ = nullptr;
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}
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bool IsOK() { return wave_.raw_data != nullptr; }
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bool Read(int *buffer, int len) {
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if (!wave_.raw_data)
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return false;
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while (bgQueue.size() < (size_t)(len * 2)) {
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int outSamples = 0;
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int inbytesConsumed = 0;
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bool result = decoder_->Decode(wave_.raw_data + raw_offset_, wave_.raw_bytes_per_frame, &inbytesConsumed, 2, (int16_t *)buffer_, &outSamples);
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if (!result || !outSamples)
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return false;
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int outBytes = outSamples * 2 * sizeof(int16_t);
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if (wave_.raw_offset_loop_end != 0 && raw_offset_ == wave_.raw_offset_loop_end) {
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// Only take the remaining bytes, but convert to stereo s16.
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outBytes = std::min(outBytes, wave_.loop_end_offset * 4);
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}
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int start = skip_next_samples_;
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skip_next_samples_ = 0;
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for (int i = start; i < outBytes / 2; i++) {
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bgQueue.push(buffer_[i]);
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}
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if (wave_.raw_offset_loop_end != 0 && raw_offset_ == wave_.raw_offset_loop_end) {
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// Time to loop. Account for the addition below.
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raw_offset_ = wave_.raw_offset_loop_start - wave_.raw_bytes_per_frame;
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// This time we're counting each stereo sample.
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skip_next_samples_ = wave_.loop_start_offset * 2;
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}
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// Handle loops when there's no loop info.
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raw_offset_ += wave_.raw_bytes_per_frame;
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if (raw_offset_ >= wave_.raw_data_size) {
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raw_offset_ = 0;
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}
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}
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for (int i = 0; i < len * 2; i++) {
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buffer[i] = bgQueue.pop_front();
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}
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return true;
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}
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private:
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RIFFReader file_;
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WavData wave_;
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int raw_offset_ = 0;
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int skip_next_samples_ = 0;
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FixedSizeQueue<s16, 128 * 1024> bgQueue;
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short *buffer_ = nullptr;
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AudioDecoder *decoder_ = nullptr;
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};
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BackgroundAudio g_BackgroundAudio;
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BackgroundAudio::BackgroundAudio() {
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buffer = new int[BUFSIZE]();
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sndLoadPending_.store(false);
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}
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BackgroundAudio::~BackgroundAudio() {
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delete at3Reader_;
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delete[] buffer;
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}
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void BackgroundAudio::Clear(bool hard) {
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if (!hard) {
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fadingOut_ = true;
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volume_ = 1.0f;
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return;
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}
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if (at3Reader_) {
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delete at3Reader_;
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at3Reader_ = nullptr;
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}
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playbackOffset_ = 0;
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sndLoadPending_ = false;
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}
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void BackgroundAudio::SetGame(const Path &path) {
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if (path == bgGamePath_) {
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// Do nothing
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return;
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}
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std::lock_guard<std::mutex> lock(mutex_);
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if (path.empty()) {
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Clear(false);
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sndLoadPending_ = false;
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fadingOut_ = true;
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} else {
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Clear(true);
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gameLastChanged_ = time_now_d();
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sndLoadPending_ = true;
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fadingOut_ = false;
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}
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volume_ = 1.0f;
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bgGamePath_ = path;
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}
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bool BackgroundAudio::Play() {
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std::lock_guard<std::mutex> lock(mutex_);
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// Immediately stop the sound if it is turned off while playing.
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if (!g_Config.bEnableSound) {
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Clear(true);
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System_AudioClear();
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return true;
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}
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double now = time_now_d();
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int sz = 44100 / 60;
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if (lastPlaybackTime_ > 0.0 && lastPlaybackTime_ <= now) {
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sz = (int)((now - lastPlaybackTime_) * 44100);
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}
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sz = std::min(BUFSIZE / 2, sz);
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if (at3Reader_) {
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if (at3Reader_->Read(buffer, sz)) {
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if (fadingOut_) {
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for (int i = 0; i < sz*2; i += 2) {
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buffer[i] *= volume_;
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buffer[i + 1] *= volume_;
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volume_ += delta_;
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}
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}
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}
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} else {
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for (int i = 0; i < sz * 2; i += 2) {
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buffer[i] = 0;
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buffer[i + 1] = 0;
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}
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}
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System_AudioPushSamples(buffer, sz);
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if (at3Reader_ && fadingOut_ && volume_ <= 0.0f) {
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Clear(true);
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fadingOut_ = false;
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gameLastChanged_ = 0;
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}
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lastPlaybackTime_ = now;
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return true;
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}
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void BackgroundAudio::Update() {
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// If there's a game, and some time has passed since the selected game
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// last changed... (to prevent crazy amount of reads when skipping through a list)
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if (sndLoadPending_ && (time_now_d() - gameLastChanged_ > 0.5)) {
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std::lock_guard<std::mutex> lock(mutex_);
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// Already loaded somehow? Or no game info cache?
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if (at3Reader_ || !g_gameInfoCache)
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return;
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// Grab some audio from the current game and play it.
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std::shared_ptr<GameInfo> gameInfo = g_gameInfoCache->GetInfo(nullptr, bgGamePath_, GameInfoFlags::SND);
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if (!gameInfo->Ready(GameInfoFlags::SND)) {
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// Should try again shortly..
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return;
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}
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const std::string &data = gameInfo->sndFileData;
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if (!data.empty()) {
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at3Reader_ = new AT3PlusReader(data);
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lastPlaybackTime_ = 0.0;
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}
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sndLoadPending_ = false;
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}
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}
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inline int16_t ConvertU8ToI16(uint8_t value) {
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int ivalue = value - 128;
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return ivalue * 255;
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}
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Sample *Sample::Load(const std::string &path) {
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size_t data_size = 0;
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uint8_t *data = g_VFS.ReadFile(path.c_str(), &data_size);
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if (!data || data_size > 100000000) {
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WARN_LOG(AUDIO, "Failed to load sample '%s'", path.c_str());
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return nullptr;
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}
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const char *mp3_magic = "ID3\03";
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const char *wav_magic = "RIFF";
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if (!memcmp(data, wav_magic, 4)) {
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RIFFReader reader(data, (int)data_size);
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WavData wave;
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if (!wave.Read(reader)) {
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delete[] data;
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return nullptr;
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}
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// A wav file.
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delete[] data;
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if (!wave.IsSimpleWAV()) {
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ERROR_LOG(AUDIO, "Wave format not supported for mixer playback. Must be 8-bit or 16-bit raw mono or stereo. '%s'", path.c_str());
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return nullptr;
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}
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int16_t *samples = new int16_t[wave.num_channels * wave.numFrames];
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if (wave.raw_bytes_per_frame == wave.num_channels * 2) {
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// 16-bit
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memcpy(samples, wave.raw_data, wave.numFrames * wave.raw_bytes_per_frame);
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} else if (wave.raw_bytes_per_frame == wave.num_channels) {
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// 8-bit. Convert.
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for (int i = 0; i < wave.num_channels * wave.numFrames; i++) {
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samples[i] = ConvertU8ToI16(wave.raw_data[i]);
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}
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}
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// Protect against bad metadata.
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int actualFrames = std::min(wave.numFrames, wave.raw_data_size / wave.raw_bytes_per_frame);
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return new Sample(samples, wave.num_channels, actualFrames, wave.sample_rate);
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}
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// Something else.
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// Let's see if minimp3 can read it.
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mp3dec_t mp3d;
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mp3dec_init(&mp3d);
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mp3dec_file_info_t mp3_info;
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int samples = mp3dec_load_buf(&mp3d, data, data_size, &mp3_info, nullptr, nullptr);
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// mp3_info contains the decoded data.
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int16_t *sample_data = new int16_t[mp3_info.samples];
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memcpy(sample_data, mp3_info.buffer, mp3_info.samples * sizeof(int16_t));
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Sample *sample = new Sample(sample_data, mp3_info.channels, (int)mp3_info.samples, mp3_info.hz);
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free(mp3_info.buffer);
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delete[] data;
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return sample;
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}
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static inline int16_t Clamp16(int32_t sample) {
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if (sample < -32767) return -32767;
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if (sample > 32767) return 32767;
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return sample;
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}
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void SoundEffectMixer::Mix(int16_t *buffer, int sz, int sampleRateHz) {
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{
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std::lock_guard<std::mutex> guard(mutex_);
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if (!queue_.empty()) {
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for (const auto &entry : queue_) {
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plays_.push_back(entry);
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}
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queue_.clear();
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}
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if (plays_.empty()) {
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return;
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}
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}
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for (std::vector<PlayInstance>::iterator iter = plays_.begin(); iter != plays_.end(); ) {
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auto sample = samples_[(int)iter->sound].get();
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if (!sample) {
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// Remove playback instance if sample invalid.
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iter = plays_.erase(iter);
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continue;
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}
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int64_t rateOfSample = sample->rateInHz_;
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int64_t stride = (rateOfSample << 32) / sampleRateHz;
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for (int i = 0; i < sz * 2; i += 2) {
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if ((iter->offset >> 32) >= sample->length_ - 2) {
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iter->done = true;
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break;
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}
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int wholeOffset = iter->offset >> 32;
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int frac = (iter->offset >> 20) & 0xFFF; // Use a 12 bit fraction to get away with 32-bit multiplies
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if (sample->channels_ == 2) {
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int interpolatedLeft = (sample->data_[wholeOffset * 2] * (0x1000 - frac) + sample->data_[(wholeOffset + 1) * 2] * frac) >> 12;
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int interpolatedRight = (sample->data_[wholeOffset * 2 + 1] * (0x1000 - frac) + sample->data_[(wholeOffset + 1) * 2 + 1] * frac) >> 12;
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// Clamping add on top per sample. Not great, we should be mixing at higher bitrate instead. Oh well.
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int left = Clamp16(buffer[i] + (interpolatedLeft * iter->volume >> 8));
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int right = Clamp16(buffer[i + 1] + (interpolatedRight * iter->volume >> 8));
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buffer[i] = left;
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buffer[i + 1] = right;
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} else if (sample->channels_ == 1) {
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int interpolated = (sample->data_[wholeOffset] * (0x1000 - frac) + sample->data_[wholeOffset + 1] * frac) >> 12;
|
|
|
|
// Clamping add on top per sample. Not great, we should be mixing at higher bitrate instead. Oh well.
|
|
int value = Clamp16(buffer[i] + (interpolated * iter->volume >> 8));
|
|
|
|
buffer[i] = value;
|
|
buffer[i + 1] = value;
|
|
}
|
|
|
|
iter->offset += stride;
|
|
}
|
|
|
|
if (iter->done) {
|
|
iter = plays_.erase(iter);
|
|
} else {
|
|
iter++;
|
|
}
|
|
}
|
|
}
|
|
|
|
void SoundEffectMixer::Play(UI::UISound sfx, float volume) {
|
|
std::lock_guard<std::mutex> guard(mutex_);
|
|
queue_.push_back(PlayInstance{ sfx, 0, (int)(255.0f * volume), false });
|
|
}
|
|
|
|
void SoundEffectMixer::UpdateSample(UI::UISound sound, Sample *sample) {
|
|
if (sample) {
|
|
std::lock_guard<std::mutex> guard(mutex_);
|
|
samples_[(size_t)sound] = std::unique_ptr<Sample>(sample);
|
|
} else {
|
|
LoadDefaultSample(sound);
|
|
}
|
|
}
|
|
|
|
void SoundEffectMixer::LoadDefaultSample(UI::UISound sound) {
|
|
const char *filename = nullptr;
|
|
switch (sound) {
|
|
case UI::UISound::BACK: filename = "sfx_back.wav"; break;
|
|
case UI::UISound::SELECT: filename = "sfx_select.wav"; break;
|
|
case UI::UISound::CONFIRM: filename = "sfx_confirm.wav"; break;
|
|
case UI::UISound::TOGGLE_ON: filename = "sfx_toggle_on.wav"; break;
|
|
case UI::UISound::TOGGLE_OFF: filename = "sfx_toggle_off.wav"; break;
|
|
case UI::UISound::ACHIEVEMENT_UNLOCKED: filename = "sfx_achievement_unlocked.wav"; break;
|
|
case UI::UISound::LEADERBOARD_SUBMITTED: filename = "sfx_leaderbord_submitted.wav"; break;
|
|
default:
|
|
return;
|
|
}
|
|
Sample *sample = Sample::Load(filename);
|
|
if (!sample) {
|
|
ERROR_LOG(SYSTEM, "Failed to load the default sample for UI sound %d", (int)sound);
|
|
}
|
|
std::lock_guard<std::mutex> guard(mutex_);
|
|
samples_[(size_t)sound] = std::unique_ptr<Sample>(sample);
|
|
}
|
|
|
|
void SoundEffectMixer::LoadSamples() {
|
|
samples_.resize((size_t)UI::UISound::COUNT);
|
|
LoadDefaultSample(UI::UISound::BACK);
|
|
LoadDefaultSample(UI::UISound::SELECT);
|
|
LoadDefaultSample(UI::UISound::CONFIRM);
|
|
LoadDefaultSample(UI::UISound::TOGGLE_ON);
|
|
LoadDefaultSample(UI::UISound::TOGGLE_OFF);
|
|
|
|
if (!g_Config.sAchievementsUnlockAudioFile.empty()) {
|
|
UpdateSample(UI::UISound::ACHIEVEMENT_UNLOCKED, Sample::Load(g_Config.sAchievementsUnlockAudioFile));
|
|
} else {
|
|
LoadDefaultSample(UI::UISound::ACHIEVEMENT_UNLOCKED);
|
|
}
|
|
if (!g_Config.sAchievementsLeaderboardSubmitAudioFile.empty()) {
|
|
UpdateSample(UI::UISound::LEADERBOARD_SUBMITTED, Sample::Load(g_Config.sAchievementsLeaderboardSubmitAudioFile));
|
|
} else {
|
|
LoadDefaultSample(UI::UISound::LEADERBOARD_SUBMITTED);
|
|
}
|
|
|
|
UI::SetSoundCallback([](UI::UISound sound, float volume) {
|
|
g_BackgroundAudio.SFX().Play(sound, volume);
|
|
});
|
|
}
|