ppsspp/UI/BackgroundAudio.cpp

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