mirror of https://github.com/OpenRCT2/OpenRCT2.git
Move reading channel data to channel class
This commit is contained in:
Ted John
parent
46b4dbb4cb
commit
8c49bb8590
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@ -354,7 +354,7 @@ private:
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int _group = MIXER_GROUP_SOUND;
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double _rate = 0;
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unsigned long _offset = 0;
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size_t _offset = 0;
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int _loop = 0;
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int _volume = 1;
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@ -428,12 +428,12 @@ public:
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unsigned long GetOffset() const override
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{
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return _offset;
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return (unsigned long)_offset;
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}
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bool SetOffset(unsigned long offset)
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{
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if (_source && offset < _source->Length())
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if (_source != nullptr && offset < _source->Length())
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{
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AudioFormat format = _source->Format();
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int samplesize = format.channels * format.BytesPerSample();
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@ -564,6 +564,53 @@ public:
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_oldvolume_l = _volume_l;
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_oldvolume_r = _volume_r;
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}
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AudioFormat GetFormat() const override
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{
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AudioFormat result = { 0 };
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if (_source != nullptr)
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{
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result = _source->Format();
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}
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return result;
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}
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size_t Read(void * dst, size_t len) override
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{
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size_t bytesRead = 0;
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size_t bytesToRead = len;
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while (bytesToRead > 0 && !_done)
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{
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const uint8 * src = nullptr;
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unsigned long someLen = _source->GetSome((unsigned long)_offset, &src, (unsigned long)bytesToRead);
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if (someLen > 0)
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{
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size_t copyLen = Math::Min((size_t)someLen, bytesToRead);
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memcpy(dst, src, copyLen);
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dst = (void *)((uintptr_t)dst + copyLen);
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bytesToRead -= copyLen;
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bytesRead += copyLen;
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_offset += copyLen;
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}
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if (_offset >= _source->Length())
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{
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if (_loop == 0)
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{
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_done = true;
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}
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else if (_loop == MIXER_LOOP_INFINITE)
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{
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_offset = 0;
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}
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else
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{
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_loop--;
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_offset = 0;
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}
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}
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}
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return bytesRead;
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}
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};
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class AudioMixer : public IAudioMixer
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@ -583,8 +630,11 @@ private:
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Source * css1sources[SOUND_MAXID];
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Source * musicsources[PATH_ID_END];
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void * _channelBuffer = nullptr;
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size_t _channelBufferCapacity = 0;
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public:
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AudioMixer::AudioMixer()
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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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@ -594,6 +644,11 @@ public:
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}
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}
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~AudioMixer()
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{
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Close();
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}
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void Init(const char* device) override
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{
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Close();
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@ -649,6 +704,7 @@ public:
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delete[] effectbuffer;
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effectbuffer = 0;
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}
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free(_channelBuffer);
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}
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void Lock() override
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@ -729,7 +785,12 @@ private:
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while (it != mixer->channels.end())
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{
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IAudioChannel * channel = *it;
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mixer->MixChannel(channel, stream, length);
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int group = channel->GetGroup();
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if (group != MIXER_GROUP_SOUND || gConfigSound.sound_enabled)
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{
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mixer->MixChannel(channel, stream, length);
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}
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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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@ -742,7 +803,7 @@ private:
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}
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}
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void MixChannel(IAudioChannel * channel, uint8* data, int length)
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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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@ -756,197 +817,159 @@ private:
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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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// Do not mix channel if channel is a sound and sound is disabled
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if (channel->GetGroup() == MIXER_GROUP_SOUND && !gConfigSound.sound_enabled) {
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return;
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}
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Source * source = channel->GetSource();
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if (source != nullptr && source->Length() > 0 && !channel->IsDone())
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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 samples = length / samplesize;
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double rate = 1;
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if (format.format == AUDIO_S16SYS)
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{
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AudioFormat streamformat = source->Format();
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int loaded = 0;
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SDL_AudioCVT cvt;
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cvt.len_ratio = 1;
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do
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rate = channel->GetRate();
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}
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int samplestoread = (int)(samples * rate);
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int lengthloaded = 0;
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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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{
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int samplesize = format.channels * format.BytesPerSample();
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int samples = length / samplesize;
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int samplesloaded = loaded / samplesize;
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double rate = 1;
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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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int samplestoread = (int)((samples - samplesloaded) * rate);
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int lengthloaded = 0;
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if (channel->GetOffset() < source->Length())
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{
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bool mustconvert = false;
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if (MustConvert(*source))
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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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{
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break;
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}
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mustconvert = true;
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}
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const uint8 * datastream = nullptr;
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int toread = (int)(samplestoread / cvt.len_ratio) * samplesize;
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int readfromstream = source->GetSome(channel->GetOffset(), &datastream, toread);
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if (readfromstream == 0)
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{
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break;
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}
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uint8* dataconverted = 0;
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const uint8* tomix = 0;
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if (mustconvert)
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{
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// tofix: there seems to be an issue with converting audio using SDL_ConvertAudio in the callback vs preconverted, can cause pops and static depending on sample rate and channels
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if (Convert(cvt, datastream, readfromstream, &dataconverted))
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{
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tomix = dataconverted;
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lengthloaded = cvt.len_cvt;
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}
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else
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{
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break;
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}
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}
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else
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{
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tomix = datastream;
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lengthloaded = readfromstream;
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}
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bool effectbufferloaded = false;
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if (rate != 1 && format.format == AUDIO_S16SYS)
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{
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int in_len = (int)((double)lengthloaded / samplesize);
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int out_len = samples;
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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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channel->SetResampler(resampler);
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}
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if (readfromstream == toread)
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{
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// use buffer lengths for conversion ratio so that it fits exactly
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speex_resampler_set_rate(resampler, in_len, samples - samplesloaded);
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}
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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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}
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speex_resampler_process_interleaved_int(resampler, (const spx_int16_t*)tomix, (spx_uint32_t*)&in_len, (spx_int16_t*)effectbuffer, (spx_uint32_t*)&out_len);
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effectbufferloaded = true;
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tomix = effectbuffer;
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lengthloaded = (out_len * samplesize);
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}
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if (channel->GetPan() != 0.5f && format.channels == 2)
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{
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if (!effectbufferloaded)
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{
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memcpy(effectbuffer, tomix, lengthloaded);
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effectbufferloaded = true;
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tomix = effectbuffer;
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}
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switch (format.format) {
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case AUDIO_S16SYS:
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EffectPanS16(channel, (sint16*)effectbuffer, lengthloaded / samplesize);
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break;
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case AUDIO_U8:
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EffectPanU8(channel, (uint8*)effectbuffer, lengthloaded / samplesize);
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break;
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}
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}
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int mixlength = lengthloaded;
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if (loaded + mixlength > length)
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{
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mixlength = length - loaded;
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}
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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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// 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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volumeadjust = Math::Min(volumeadjust, 0.75f);
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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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break;
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}
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int startvolume = (int)(channel->GetOldVolume() * volumeadjust);
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int endvolume = (int)(channel->GetVolume() * volumeadjust);
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if (channel->IsStopping())
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{
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endvolume = 0;
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}
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int mixvolume = (int)(channel->GetVolume() * volumeadjust);
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if (startvolume != endvolume)
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{
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// fade between volume levels to smooth out sound and minimize clicks from sudden volume changes
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if (!effectbufferloaded)
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{
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memcpy(effectbuffer, tomix, lengthloaded);
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effectbufferloaded = true;
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tomix = effectbuffer;
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}
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mixvolume = SDL_MIX_MAXVOLUME; // set to max since we are adjusting the volume ourselves
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int fadelength = mixlength / format.BytesPerSample();
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switch (format.format) {
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case AUDIO_S16SYS:
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EffectFadeS16((sint16*)effectbuffer, fadelength, startvolume, endvolume);
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break;
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case AUDIO_U8:
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EffectFadeU8((uint8*)effectbuffer, fadelength, startvolume, endvolume);
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break;
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}
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}
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SDL_MixAudioFormat(&data[loaded], tomix, format.format, mixlength, mixvolume);
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if (dataconverted)
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{
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delete[] dataconverted;
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}
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channel->SetOffset(channel->GetOffset() + readfromstream);
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}
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loaded += lengthloaded;
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int loop = channel->GetLoop();
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if (loop != 0 && channel->GetOffset() >= source->Length())
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{
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if (loop != -1)
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{
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channel->SetLoop(loop - 1);
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}
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channel->SetOffset(0);
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}
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// Unable to convert channel data
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return;
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}
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while (loaded < length && channel->GetLoop() != 0 && !channel->IsStopping());
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mustconvert = true;
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}
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channel->UpdateOldVolume();
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if (channel->GetLoop() == 0 && channel->GetOffset() >= source->Length())
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// Read raw PCM from channel
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size_t toread = (size_t)(samplestoread / cvt.len_ratio) * samplesize;
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if (_channelBuffer == nullptr || _channelBufferCapacity < toread)
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{
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_channelBuffer = realloc(_channelBuffer, toread);
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_channelBufferCapacity = toread;
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}
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size_t bytesRead = channel->Read(_channelBuffer, toread);
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// Convert data to required format if necessary
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uint8 * convertedBuffer = nullptr;
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const uint8 * buffer = 0;
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if (mustconvert)
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{
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// tofix: there seems to be an issue with converting audio using SDL_ConvertAudio in the callback vs preconverted, can cause pops and static depending on sample rate and channels
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if (Convert(cvt, (const uint8 *)_channelBuffer, (unsigned long)bytesRead, &convertedBuffer))
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{
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channel->SetDone(true);
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buffer = convertedBuffer;
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lengthloaded = cvt.len_cvt;
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}
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else
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{
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return;
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}
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}
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else
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{
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buffer = (const uint8 *)_channelBuffer;
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lengthloaded = (int)bytesRead;
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}
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// Apply effects
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bool effectbufferloaded = false;
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if (rate != 1 && format.format == AUDIO_S16SYS)
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{
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int in_len = (int)((double)lengthloaded / samplesize);
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int out_len = samples;
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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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channel->SetResampler(resampler);
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}
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if (bytesRead == toread)
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{
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// use buffer lengths for conversion ratio so that it fits exactly
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speex_resampler_set_rate(resampler, in_len, samples);
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}
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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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}
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speex_resampler_process_interleaved_int(resampler, (const spx_int16_t*)buffer, (spx_uint32_t*)&in_len, (spx_int16_t*)effectbuffer, (spx_uint32_t*)&out_len);
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effectbufferloaded = true;
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buffer = effectbuffer;
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lengthloaded = (out_len * samplesize);
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}
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// Pan
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if (channel->GetPan() != 0.5f && format.channels == 2)
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{
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if (!effectbufferloaded)
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{
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memcpy(effectbuffer, buffer, lengthloaded);
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effectbufferloaded = true;
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buffer = effectbuffer;
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}
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switch (format.format) {
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case AUDIO_S16SYS:
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EffectPanS16(channel, (sint16*)effectbuffer, lengthloaded / samplesize);
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break;
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case AUDIO_U8:
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EffectPanU8(channel, (uint8*)effectbuffer, lengthloaded / samplesize);
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break;
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}
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}
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int mixlength = Math::Min(lengthloaded, length);
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// 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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// 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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volumeadjust = Math::Min(volumeadjust, 0.75f);
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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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break;
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}
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int startvolume = (int)(channel->GetOldVolume() * volumeadjust);
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int endvolume = (int)(channel->GetVolume() * volumeadjust);
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if (channel->IsStopping())
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{
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endvolume = 0;
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}
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int mixvolume = (int)(channel->GetVolume() * volumeadjust);
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if (startvolume != endvolume)
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{
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// fade between volume levels to smooth out sound and minimize clicks from sudden volume changes
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if (!effectbufferloaded)
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{
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memcpy(effectbuffer, buffer, lengthloaded);
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effectbufferloaded = true;
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buffer = effectbuffer;
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}
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mixvolume = SDL_MIX_MAXVOLUME; // set to max since we are adjusting the volume ourselves
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int fadelength = mixlength / format.BytesPerSample();
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switch (format.format) {
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case AUDIO_S16SYS:
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EffectFadeS16((sint16*)effectbuffer, fadelength, startvolume, endvolume);
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break;
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case AUDIO_U8:
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EffectFadeU8((uint8*)effectbuffer, fadelength, startvolume, endvolume);
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break;
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}
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}
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SDL_MixAudioFormat(data, buffer, format.format, mixlength, mixvolume);
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delete[] convertedBuffer;
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channel->UpdateOldVolume();
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}
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static void EffectPanS16(const IAudioChannel * channel, sint16 * data, int length)
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|
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@ -1003,12 +1026,11 @@ private:
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}
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}
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bool MustConvert(Source& source)
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bool MustConvert(const AudioFormat * sourceFormat)
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{
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const AudioFormat sourceformat = source.Format();
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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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||||
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|
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@ -170,6 +170,9 @@ interface IAudioChannel
|
|||
|
||||
virtual void Play(Source& source, int loop = MIXER_LOOP_NONE) abstract;
|
||||
virtual void UpdateOldVolume() abstract;
|
||||
|
||||
virtual AudioFormat GetFormat() const abstract;
|
||||
virtual size_t Read(void * dst, size_t len) abstract;
|
||||
};
|
||||
|
||||
interface IAudioMixer
|
||||
|
|
|
|||
Loading…
Reference in New Issue