mirror of https://github.com/OpenRCT2/OpenRCT2.git
Add underscores to class fields
This commit is contained in:
Ted John
parent
324dea94bc
commit
c70c80c8f4
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@ -616,18 +616,18 @@ public:
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class AudioMixer : public IAudioMixer
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{
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private:
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SDL_AudioDeviceID deviceid = 0;
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AudioFormat format = { 0 };
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std::list<IAudioChannel *> channels;
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Source_Null source_null;
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float volume = 1.0f;
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float adjust_sound_vol = 0.0f;
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float adjust_music_vol = 0.0f;
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uint8 setting_sound_vol = 0xFF;
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uint8 setting_music_vol = 0xFF;
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SDL_AudioDeviceID _deviceid = 0;
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AudioFormat _format = { 0 };
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std::list<IAudioChannel *> _channels;
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Source_Null _source_null;
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float _volume = 1.0f;
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float _adjust_sound_vol = 0.0f;
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float _adjust_music_vol = 0.0f;
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uint8 _setting_sound_vol = 0xFF;
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uint8 _setting_music_vol = 0xFF;
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Source * css1sources[SOUND_MAXID];
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Source * musicsources[PATH_ID_END];
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Source * _css1sources[SOUND_MAXID] = { nullptr };
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Source * _musicsources[PATH_ID_END] = { nullptr };
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void * _channelBuffer = nullptr;
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size_t _channelBufferCapacity = 0;
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@ -639,12 +639,6 @@ private:
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public:
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AudioMixer()
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{
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for (size_t i = 0; i < Util::CountOf(css1sources); i++) {
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css1sources[i] = 0;
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}
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for (size_t i = 0; i < Util::CountOf(musicsources); i++) {
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musicsources[i] = 0;
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}
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}
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~AudioMixer()
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@ -663,45 +657,54 @@ public:
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want.samples = 1024;
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want.callback = Callback;
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want.userdata = this;
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deviceid = SDL_OpenAudioDevice(device, 0, &want, &have, 0);
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format.format = have.format;
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format.channels = have.channels;
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format.freq = have.freq;
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_deviceid = SDL_OpenAudioDevice(device, 0, &want, &have, 0);
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_format.format = have.format;
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_format.channels = have.channels;
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_format.freq = have.freq;
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const char* filename = get_file_path(PATH_ID_CSS1);
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for (int i = 0; i < (int)Util::CountOf(css1sources); i++) {
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for (int i = 0; i < (int)Util::CountOf(_css1sources); i++) {
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Source_Sample* source_sample = new Source_Sample;
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if (source_sample->LoadCSS1(filename, i)) {
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source_sample->Convert(format); // convert to audio output format, saves some cpu usage but requires a bit more memory, optional
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css1sources[i] = source_sample;
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source_sample->Convert(_format); // convert to audio output format, saves some cpu usage but requires a bit more memory, optional
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_css1sources[i] = source_sample;
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} else {
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css1sources[i] = &source_null;
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_css1sources[i] = &_source_null;
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delete source_sample;
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}
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}
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SDL_PauseAudioDevice(deviceid, 0);
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SDL_PauseAudioDevice(_deviceid, 0);
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}
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void Close() override
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{
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// Free channels
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Lock();
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while (channels.begin() != channels.end()) {
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delete *(channels.begin());
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channels.erase(channels.begin());
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for (IAudioChannel * channel : _channels)
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{
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delete channel;
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}
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_channels.clear();
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Unlock();
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SDL_CloseAudioDevice(deviceid);
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for (size_t i = 0; i < Util::CountOf(css1sources); i++) {
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if (css1sources[i] && css1sources[i] != &source_null) {
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delete css1sources[i];
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css1sources[i] = 0;
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SDL_CloseAudioDevice(_deviceid);
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// Free sources
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for (size_t i = 0; i < Util::CountOf(_css1sources); i++)
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{
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if (_css1sources[i] && _css1sources[i] != &_source_null)
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{
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SafeDelete(_css1sources[i]);
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}
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}
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for (size_t i = 0; i < Util::CountOf(musicsources); i++) {
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if (musicsources[i] && musicsources[i] != &source_null) {
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delete musicsources[i];
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musicsources[i] = 0;
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for (size_t i = 0; i < Util::CountOf(_musicsources); i++)
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{
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if (_musicsources[i] && _musicsources[i] != &_source_null)
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{
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SafeDelete(_musicsources[i]);
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}
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}
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// Free buffers
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SafeFree(_channelBuffer);
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SafeFree(_convertBuffer);
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SafeFree(_effectBuffer);
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@ -709,12 +712,12 @@ public:
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void Lock() override
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{
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SDL_LockAudioDevice(deviceid);
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SDL_LockAudioDevice(_deviceid);
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}
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void Unlock() override
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{
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SDL_UnlockAudioDevice(deviceid);
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SDL_UnlockAudioDevice(_deviceid);
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}
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IAudioChannel * Play(Source& source, int loop, bool deleteondone, bool deletesourceondone) override
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@ -726,7 +729,7 @@ public:
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newchannel->Play(source, loop);
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newchannel->SetDeleteOnDone(deleteondone);
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newchannel->SetDeleteSourceOnDone(deletesourceondone);
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channels.push_back(newchannel);
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_channels.push_back(newchannel);
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}
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Unlock();
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return newchannel;
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@ -741,38 +744,45 @@ public:
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bool LoadMusic(size_t pathId) override
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{
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if (pathId >= Util::CountOf(musicsources)) {
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if (pathId >= Util::CountOf(_musicsources))
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{
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return false;
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}
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if (!musicsources[pathId]) {
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if (!_musicsources[pathId])
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{
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const char* filename = get_file_path((int)pathId);
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Source_Sample* source_sample = new Source_Sample;
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if (source_sample->LoadWAV(filename)) {
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musicsources[pathId] = source_sample;
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if (source_sample->LoadWAV(filename))
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{
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_musicsources[pathId] = source_sample;
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return true;
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} else {
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}
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else
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{
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delete source_sample;
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musicsources[pathId] = &source_null;
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_musicsources[pathId] = &_source_null;
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return false;
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}
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} else {
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}
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else
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{
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return true;
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}
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}
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void SetVolume(float volume) override
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{
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this->volume = volume;
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_volume = volume;
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}
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Source * GetSoundSource(int id) override
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{
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return css1sources[id];
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return _css1sources[id];
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}
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Source * GetMusicSource(int id) override
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{
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return musicsources[id];
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return _musicsources[id];
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}
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private:
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@ -781,8 +791,8 @@ private:
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auto mixer = static_cast<AudioMixer *>(arg);
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memset(stream, 0, length);
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std::list<IAudioChannel *>::iterator it = mixer->channels.begin();
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while (it != mixer->channels.end())
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std::list<IAudioChannel *>::iterator it = mixer->_channels.begin();
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while (it != mixer->_channels.end())
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{
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IAudioChannel * channel = *it;
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@ -794,7 +804,7 @@ private:
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if ((channel->IsDone() && channel->DeleteOnDone()) || channel->IsStopping())
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{
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delete channel;
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it = mixer->channels.erase(it);
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it = mixer->_channels.erase(it);
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}
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else
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{
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@ -806,24 +816,24 @@ private:
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void MixChannel(IAudioChannel * channel, uint8 * data, int length)
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{
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// Did the volume level get changed? Recalculate level in this case.
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if (setting_sound_vol != gConfigSound.sound_volume)
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if (_setting_sound_vol != gConfigSound.sound_volume)
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{
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setting_sound_vol = gConfigSound.sound_volume;
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adjust_sound_vol = powf(setting_sound_vol / 100.f, 10.f / 6.f);
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_setting_sound_vol = gConfigSound.sound_volume;
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_adjust_sound_vol = powf(_setting_sound_vol / 100.f, 10.f / 6.f);
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}
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if (setting_music_vol != gConfigSound.ride_music_volume)
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if (_setting_music_vol != gConfigSound.ride_music_volume)
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{
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setting_music_vol = gConfigSound.ride_music_volume;
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adjust_music_vol = powf(setting_music_vol / 100.f, 10.f / 6.f);
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_setting_music_vol = gConfigSound.ride_music_volume;
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_adjust_music_vol = powf(_setting_music_vol / 100.f, 10.f / 6.f);
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}
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AudioFormat streamformat = channel->GetFormat();
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SDL_AudioCVT cvt;
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cvt.len_ratio = 1;
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int samplesize = format.channels * format.BytesPerSample();
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int samplesize = _format.channels * _format.BytesPerSample();
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int samples = length / samplesize;
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double rate = 1;
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if (format.format == AUDIO_S16SYS)
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if (_format.format == AUDIO_S16SYS)
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{
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rate = channel->GetRate();
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}
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@ -831,7 +841,7 @@ private:
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bool mustConvert = false;
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if (MustConvert(&streamformat))
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{
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if (SDL_BuildAudioCVT(&cvt, streamformat.format, streamformat.channels, streamformat.freq, format.format, format.channels, format.freq) == -1)
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if (SDL_BuildAudioCVT(&cvt, streamformat.format, streamformat.channels, streamformat.freq, _format.format, _format.channels, _format.freq) == -1)
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{
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// Unable to convert channel data
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return;
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@ -870,7 +880,7 @@ private:
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}
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// Apply effects
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if (rate != 1 && format.format == AUDIO_S16SYS)
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if (rate != 1 && _format.format == AUDIO_S16SYS)
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{
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int in_len = (int)((double)bufferLen / samplesize);
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int out_len = samples;
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@ -878,7 +888,7 @@ private:
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SpeexResamplerState * resampler = channel->GetResampler();
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if (resampler == nullptr)
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{
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resampler = speex_resampler_init(format.channels, format.freq, format.freq, 0, 0);
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resampler = speex_resampler_init(_format.channels, _format.freq, _format.freq, 0, 0);
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channel->SetResampler(resampler);
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}
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if (bytesRead == toread)
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@ -889,7 +899,7 @@ private:
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else
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{
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// reached end of stream so we cant use buffer length as resampling ratio
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speex_resampler_set_rate(resampler, format.freq, (int)(format.freq * (1 / rate)));
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speex_resampler_set_rate(resampler, _format.freq, (int)(_format.freq * (1 / rate)));
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}
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size_t effectBufferReqLen = out_len * samplesize;
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@ -913,16 +923,16 @@ private:
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int mixVolume = ApplyVolume(channel, buffer, bufferLen);
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size_t dstLength = Math::Min((size_t)length, bufferLen);
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SDL_MixAudioFormat(data, (const Uint8 *)buffer, format.format, (Uint32)dstLength, mixVolume);
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SDL_MixAudioFormat(data, (const Uint8 *)buffer, _format.format, (Uint32)dstLength, mixVolume);
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channel->UpdateOldVolume();
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}
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void ApplyPan(const IAudioChannel * channel, void * buffer, size_t len, size_t sampleSize)
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{
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if (channel->GetPan() != 0.5f && format.channels == 2)
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if (channel->GetPan() != 0.5f && _format.channels == 2)
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{
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switch (format.format) {
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switch (_format.format) {
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case AUDIO_S16SYS:
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EffectPanS16(channel, (sint16 *)buffer, (int)(len / sampleSize));
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break;
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@ -935,11 +945,11 @@ private:
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int ApplyVolume(const IAudioChannel * channel, void * buffer, size_t len)
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{
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float volumeAdjust = volume;
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float volumeAdjust = _volume;
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volumeAdjust *= (gConfigSound.master_volume / 100.0f);
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switch (channel->GetGroup()) {
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case MIXER_GROUP_SOUND:
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volumeAdjust *= adjust_sound_vol;
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volumeAdjust *= _adjust_sound_vol;
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// Cap sound volume on title screen so music is more audible
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if (gScreenFlags & SCREEN_FLAGS_TITLE_DEMO)
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@ -948,7 +958,7 @@ private:
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}
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break;
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case MIXER_GROUP_RIDE_MUSIC:
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volumeAdjust *= adjust_music_vol;
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volumeAdjust *= _adjust_music_vol;
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break;
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}
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@ -966,8 +976,8 @@ private:
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mixVolume = SDL_MIX_MAXVOLUME;
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// Fade between volume levels to smooth out sound and minimize clicks from sudden volume changes
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int fadeLength = (int)len / format.BytesPerSample();
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switch (format.format) {
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int fadeLength = (int)len / _format.BytesPerSample();
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switch (_format.format) {
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case AUDIO_S16SYS:
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EffectFadeS16((sint16 *)buffer, fadeLength, startVolume, endVolume);
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break;
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@ -1035,9 +1045,9 @@ private:
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bool MustConvert(const AudioFormat * sourceFormat)
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{
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if (sourceFormat->format != format.format ||
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sourceFormat->channels != format.channels ||
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sourceFormat->freq != format.freq)
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if (sourceFormat->format != _format.format ||
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sourceFormat->channels != _format.channels ||
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sourceFormat->freq != _format.freq)
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{
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return true;
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}
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