void Mixer::MixChannel(Channel& channel, uint8* data, int length)
{
// Do not mix channel if channel is a sound and sound is disabled
if (channel.group == MIXER_GROUP_SOUND && !gConfigSound.sound_enabled) {
return;
}
if (channel.source && channel.source->Length() > 0 && !channel.done) {
AudioFormat streamformat = channel.source->Format();
int loaded = 0;
SDL_AudioCVT cvt;
cvt.len_ratio = 1;
do {
int samplesize = format.channels * format.BytesPerSample();
int samples = length / samplesize;
int samplesloaded = loaded / samplesize;
double rate = 1;
if (format.format == AUDIO_S16SYS) {
rate = channel.rate;
}
int samplestoread = (int)((samples - samplesloaded) * rate);
int lengthloaded = 0;
if (channel.offset < channel.source->Length()) {
bool mustconvert = false;
if (MustConvert(*channel.source)) {
if (SDL_BuildAudioCVT(&cvt, streamformat.format, streamformat.channels, streamformat.freq, Mixer::format.format, Mixer::format.channels, Mixer::format.freq) == -1) {
break;
}
mustconvert = true;
}
const uint8* datastream = 0;
int toread = (int)(samplestoread / cvt.len_ratio) * samplesize;
int readfromstream = (channel.source->GetSome(channel.offset, &datastream, toread));
if (readfromstream == 0) {
break;
}
uint8* dataconverted = 0;
const uint8* tomix = 0;
if (mustconvert) {
// 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
if (Convert(cvt, datastream, readfromstream, &dataconverted)) {
tomix = dataconverted;
lengthloaded = cvt.len_cvt;
} else {
break;
}
} else {
tomix = datastream;
lengthloaded = readfromstream;
}
bool effectbufferloaded = false;
if (rate != 1 && format.format == AUDIO_S16SYS) {
int in_len = (int)((double)lengthloaded / samplesize);
int out_len = samples;
if (!channel.resampler) {
channel.resampler = speex_resampler_init(format.channels, format.freq, format.freq, 0, 0);
}
if (readfromstream == toread) {
// use buffer lengths for conversion ratio so that it fits exactly
speex_resampler_set_rate(channel.resampler, in_len, samples - samplesloaded);
} else {
// reached end of stream so we cant use buffer length as resampling ratio
speex_resampler_set_rate(channel.resampler, format.freq, (int)(format.freq * (1 / rate)));
}
speex_resampler_process_interleaved_int(channel.resampler, (const spx_int16_t*)tomix, (spx_uint32_t*)&in_len, (spx_int16_t*)effectbuffer, (spx_uint32_t*)&out_len);
effectbufferloaded = true;
tomix = effectbuffer;
lengthloaded = (out_len * samplesize);
}
if (channel.pan != 0.5f && format.channels == 2) {
if (!effectbufferloaded) {
memcpy(effectbuffer, tomix, lengthloaded);
effectbufferloaded = true;
tomix = effectbuffer;
}
switch (format.format) {
case AUDIO_S16SYS:
EffectPanS16(channel, (sint16*)effectbuffer, lengthloaded / samplesize);
break;
case AUDIO_U8:
EffectPanU8(channel, (uint8*)effectbuffer, lengthloaded / samplesize);
break;
}
}
int mixlength = lengthloaded;
if (loaded + mixlength > length) {
mixlength = length - loaded;
}
float volumeadjust = volume;
volumeadjust *= (gConfigSound.master_volume / 100.0f);
switch (channel.group) {
case MIXER_GROUP_SOUND:
volumeadjust *= (gConfigSound.sound_volume / 100.0f);
// Cap sound volume on title screen so music is more audible
if (RCT2_GLOBAL(RCT2_ADDRESS_SCREEN_FLAGS, uint8) & SCREEN_FLAGS_TITLE_DEMO) {
volumeadjust = Math::Min(volumeadjust, 0.75f);
}
break;
case MIXER_GROUP_RIDE_MUSIC:
volumeadjust *= (gConfigSound.ride_music_volume / 100.0f);
break;
}
int startvolume = (int)(channel.oldvolume * volumeadjust);
int endvolume = (int)(channel.volume * volumeadjust);
if (channel.stopping) {
endvolume = 0;
}
int mixvolume = (int)(channel.volume * volumeadjust);
if (startvolume != endvolume) {
// fade between volume levels to smooth out sound and minimize clicks from sudden volume changes
if (!effectbufferloaded) {
memcpy(effectbuffer, tomix, lengthloaded);
effectbufferloaded = true;
tomix = effectbuffer;
}
mixvolume = SDL_MIX_MAXVOLUME; // set to max since we are adjusting the volume ourselves
int fadelength = mixlength / format.BytesPerSample();
switch (format.format) {
case AUDIO_S16SYS:
EffectFadeS16((sint16*)effectbuffer, fadelength, startvolume, endvolume);
break;
case AUDIO_U8:
EffectFadeU8((uint8*)effectbuffer, fadelength, startvolume, endvolume);
break;
}
}
SDL_MixAudioFormat(&data[loaded], tomix, format.format, mixlength, mixvolume);
if (dataconverted) {
delete[] dataconverted;
}
channel.offset += readfromstream;
}
loaded += lengthloaded;
if (channel.loop != 0 && channel.offset >= channel.source->Length()) {
if (channel.loop != -1) {
channel.loop--;
}
channel.offset = 0;
}
} while(loaded < length && channel.loop != 0 && !channel.stopping);
channel.oldvolume = channel.volume;
channel.oldvolume_l = channel.volume_l;
channel.oldvolume_r = channel.volume_r;
if (channel.loop == 0 && channel.offset >= channel.source->Length()) {
channel.done = true;
}
}
}
void Mixer::MixChannel(Channel& channel, uint8* data, int length)
{
if (channel.stream) {
if (!channel.resampler) {
channel.resampler = speex_resampler_init(format.channels, format.freq, format.freq, 0, 0);
}
AudioFormat channelformat = *channel.stream->Format();
int loaded = 0;
SDL_AudioCVT cvt;
cvt.len_ratio = 1;
do {
int samplesize = format.channels * format.BytesPerSample();
int samples = length / samplesize;
int samplesloaded = loaded / samplesize;
int samplestoread = (int)ceil((samples - samplesloaded) * channel.rate);
int lengthloaded = 0;
if (channel.offset < channel.stream->Length()) {
bool mustconvert = false;
if (MustConvert(*channel.stream)) {
if (SDL_BuildAudioCVT(&cvt, channelformat.format, channelformat.channels, channelformat.freq, Mixer::format.format, Mixer::format.channels, Mixer::format.freq) == -1) {
break;
}
mustconvert = true;
}
const uint8* datastream = 0;
int readfromstream = (channel.stream->GetSome(channel.offset, &datastream, (int)(((samplestoread) * samplesize) / cvt.len_ratio)) / channelformat.BytesPerSample()) * channelformat.BytesPerSample();
if (readfromstream == 0) {
break;
}
int volume = channel.volume;
uint8* dataconverted = 0;
const uint8* tomix = 0;
if (mustconvert) {
if (Convert(cvt, datastream, readfromstream, &dataconverted)) {
tomix = dataconverted;
lengthloaded = (cvt.len_cvt / samplesize) * samplesize;
} else {
break;
}
} else {
tomix = datastream;
lengthloaded = readfromstream;
}
bool effectbufferloaded = false;
if (channel.rate != 1 && format.format == AUDIO_S16SYS) {
int in_len = (int)(ceil((double)lengthloaded / samplesize));
int out_len = samples + 20; // needs some extra, otherwise resampler sometimes doesn't process all the input samples
speex_resampler_set_rate(channel.resampler, format.freq, (int)(format.freq * (1 / channel.rate)));
speex_resampler_process_interleaved_int(channel.resampler, (const spx_int16_t*)tomix, (spx_uint32_t*)&in_len, (spx_int16_t*)effectbuffer, (spx_uint32_t*)&out_len);
effectbufferloaded = true;
tomix = effectbuffer;
lengthloaded = (out_len * samplesize);
}
if (channel.pan != 0.5f && format.channels == 2) {
if (!effectbufferloaded) {
memcpy(effectbuffer, tomix, lengthloaded);
effectbufferloaded = true;
tomix = effectbuffer;
}
switch (format.format) {
case AUDIO_S16SYS:
EffectPanS16(channel, (sint16*)effectbuffer, lengthloaded / samplesize);
break;
case AUDIO_U8:
EffectPanU8(channel, (uint8*)effectbuffer, lengthloaded / samplesize);
break;
}
}
int mixlength = lengthloaded;
if (loaded + mixlength > length) {
mixlength = length - loaded;
}
SDL_MixAudioFormat(&data[loaded], tomix, format.format, mixlength, volume);
if (dataconverted) {
delete[] dataconverted;
}
channel.offset += readfromstream;
}
loaded += lengthloaded;
if (channel.loop != 0 && channel.offset >= channel.stream->Length()) {
if (channel.loop != -1) {
channel.loop--;
}
channel.offset = 0;
}
} while(loaded < length && channel.loop != 0);
}
}
