Refactor CD-DA flow by removing intermediate buffers and loops
Thanks to @ripsaw8080 for insight into CD-DA channel mapping, @hail-to-the-ryzen for testing and flagging a position-tracking bug, and @dreamer_ for guidance and code review. The CD-DA volume and channel mapping loops were moved to generic mixer calls and no longer require a pre-processing loop: - Application-controlled CD-DA volume adjustment is now applied using an existing mixer volume scalar that was previously unused by the CD-DA code. - Mapping of CD-DA left and right channels is now applied at the tail end of the mixer's sample ingest sequence. The following have been removed: - The CD-DA callback chunk-wise circular buffer - The decode buffers in the Opus and MP3 decoders - The decode buffer and conversion buffers in SDL_Sound These removals and API changes allow the image player's buffer to be passed-through ultimately to the audio codec, skipping multiple intermediate buffers.
This commit is contained in:
krcroft
Patryk Obara
parent
5a9dd2866b
commit
d1a6f373cb
5 changed files with 310 additions and 336 deletions
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@ -22,15 +22,20 @@
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That should call the mixer start from there or something.
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*/
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// #define DEBUG 1
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#include <string.h>
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#include <sys/types.h>
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#include <math.h>
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#include <algorithm>
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#if defined (WIN32)
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//Midi listing
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#ifndef WIN32_LEAN_AND_MEAN
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#define WIN32_LEAN_AND_MEAN
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#endif
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// prevent the Windows header from clobbering std::min and max
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#define NOMINMAX
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#include <windows.h>
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#include <mmsystem.h>
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#endif
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@ -90,16 +95,35 @@ static struct {
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Bit8u MixTemp[MIXER_BUFSIZE];
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MixerChannel * MIXER_AddChannel(MIXER_Handler handler,Bitu freq,const char * name) {
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MixerChannel * chan=new MixerChannel();
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chan->scale = 1.0;
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chan->handler=handler;
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chan->name=name;
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chan->SetFreq(freq);
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MixerChannel::MixerChannel(MIXER_Handler _handler, Bitu _freq, const char * _name) :
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// Public member initialization
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volmain {0, 0},
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next (nullptr),
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name (_name),
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done (0),
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enabled (false),
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// Private member initialization
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handler (_handler),
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freq_add (0),
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freq_counter (0),
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needed (0),
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prev_sample {0, 0},
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next_sample {0, 0},
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volmul {0, 0},
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scale {0, 0},
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channel_map {0, 0},
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interpolate (false) {
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}
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MixerChannel * MIXER_AddChannel(MIXER_Handler handler, Bitu freq, const char * name) {
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MixerChannel * chan=new MixerChannel(handler, freq, name);
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chan->next=mixer.channels;
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chan->SetVolume(1,1);
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chan->enabled=false;
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chan->interpolate = false;
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chan->SetFreq(freq); // also enables 'interpolate' if needed
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chan->SetScale(1.0);
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chan->SetVolume(1, 1);
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chan->MapChannels(0, 1);
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chan->Enable(false);
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mixer.channels=chan;
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return chan;
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}
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@ -128,19 +152,59 @@ void MIXER_DelChannel(MixerChannel* delchan) {
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}
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void MixerChannel::UpdateVolume(void) {
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volmul[0]=(Bits)((1 << MIXER_VOLSHIFT)*scale*volmain[0]*mixer.mastervol[0]);
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volmul[1]=(Bits)((1 << MIXER_VOLSHIFT)*scale*volmain[1]*mixer.mastervol[1]);
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volmul[0]=(Bits)((1 << MIXER_VOLSHIFT)*scale[0]*volmain[0]*mixer.mastervol[0]);
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volmul[1]=(Bits)((1 << MIXER_VOLSHIFT)*scale[1]*volmain[1]*mixer.mastervol[1]);
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}
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template <typename T>
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T clamp(const T& n, const T& lower, const T& upper) {
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return std::max(lower, std::min(n, upper));
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}
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void MixerChannel::SetVolume(float _left,float _right) {
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volmain[0]=_left;
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volmain[1]=_right;
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// Allow unconstrained user-defined values
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volmain[0] = _left;
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volmain[1] = _right;
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UpdateVolume();
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}
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void MixerChannel::SetScale( float f ) {
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scale = f;
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UpdateVolume();
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void MixerChannel::SetScale(float f) {
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SetScale(f, f);
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}
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void MixerChannel::SetScale(float _left, float _right) {
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// Constrain application-defined volume between 0% and 100%
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const float min_volume(0.0);
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const float max_volume(1.0);
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_left = clamp(_left, min_volume, max_volume);
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_right = clamp(_right, min_volume, max_volume);
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if (scale[0] != _left || scale[1] != _right) {
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scale[0] = _left;
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scale[1] = _right;
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UpdateVolume();
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#ifdef DEBUG
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LOG_MSG("MIXER %-7s channel: application changed left and right volumes to %3.0f%% and %3.0f%%, respectively",
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name, scale[0] * 100, scale[1] * 100);
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#endif
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}
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}
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void MixerChannel::MapChannels(Bit8u _left, Bit8u _right) {
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// Constrain mapping to the 0 (left) and 1 (right) channel indexes
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const Bit8u min_channel(0);
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const Bit8u max_channel(1);
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_left = clamp(_left, min_channel, max_channel);
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_right = clamp(_right, min_channel, max_channel);
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if (channel_map[0] != _left || channel_map[1] != _right) {
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channel_map[0] = _left;
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channel_map[1] = _right;
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#ifdef DEBUG
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LOG_MSG("MIXER %-7s channel: application changed audio-channel mapping to left=>%s and right=>%s",
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name,
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channel_map[0] == 0 ? "left" : "right",
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channel_map[1] == 0 ? "left" : "right");
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#endif
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}
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}
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void MixerChannel::Enable(bool _yesno) {
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@ -179,8 +243,8 @@ void MixerChannel::AddSilence(void) {
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if (done<needed) {
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done=needed;
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//Make sure the next samples are zero when they get switched to prev
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nextSample[0] = 0;
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nextSample[1] = 0;
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next_sample[0] = 0;
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next_sample[1] = 0;
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//This should trigger an instant request for new samples
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freq_counter = FREQ_NEXT;
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}
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@ -200,24 +264,26 @@ inline void MixerChannel::AddSamples(Bitu len, const Type* data) {
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if (pos >= len)
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return;
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freq_counter -= FREQ_NEXT;
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prevSample[0] = nextSample[0];
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prev_sample[0] = next_sample[0];
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if (stereo) {
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prevSample[1] = nextSample[1];
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prev_sample[1] = next_sample[1];
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}
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if ( sizeof( Type) == 1) {
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if (!signeddata) {
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if (stereo) {
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nextSample[0]=(((Bit8s)(data[pos*2+0] ^ 0x80)) << 8);
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nextSample[1]=(((Bit8s)(data[pos*2+1] ^ 0x80)) << 8);
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next_sample[0]=(((Bit8s)(data[pos*2+0] ^ 0x80)) << 8);
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next_sample[1]=(((Bit8s)(data[pos*2+1] ^ 0x80)) << 8);
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} else {
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nextSample[0]=(((Bit8s)(data[pos] ^ 0x80)) << 8);
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next_sample[0]=(((Bit8s)(data[pos] ^ 0x80)) << 8);
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}
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} else {
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if (stereo) {
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nextSample[0]=(data[pos*2+0] << 8);
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nextSample[1]=(data[pos*2+1] << 8);
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next_sample[0]=(data[pos*2+0] << 8);
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next_sample[1]=(data[pos*2+1] << 8);
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} else {
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nextSample[0]=(data[pos] << 8);
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next_sample[0]=(data[pos] << 8);
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}
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}
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//16bit and 32bit both contain 16bit data internally
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@ -225,50 +291,50 @@ inline void MixerChannel::AddSamples(Bitu len, const Type* data) {
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if (signeddata) {
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if (stereo) {
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if (nativeorder) {
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nextSample[0]=data[pos*2+0];
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nextSample[1]=data[pos*2+1];
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next_sample[0]=data[pos*2+0];
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next_sample[1]=data[pos*2+1];
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} else {
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if ( sizeof( Type) == 2) {
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nextSample[0]=(Bit16s)host_readw((HostPt)&data[pos*2+0]);
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nextSample[1]=(Bit16s)host_readw((HostPt)&data[pos*2+1]);
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next_sample[0]=(Bit16s)host_readw((HostPt)&data[pos*2+0]);
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next_sample[1]=(Bit16s)host_readw((HostPt)&data[pos*2+1]);
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} else {
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nextSample[0]=(Bit32s)host_readd((HostPt)&data[pos*2+0]);
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nextSample[1]=(Bit32s)host_readd((HostPt)&data[pos*2+1]);
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next_sample[0]=(Bit32s)host_readd((HostPt)&data[pos*2+0]);
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next_sample[1]=(Bit32s)host_readd((HostPt)&data[pos*2+1]);
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}
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}
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} else {
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if (nativeorder) {
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nextSample[0] = data[pos];
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next_sample[0] = data[pos];
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} else {
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if ( sizeof( Type) == 2) {
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nextSample[0]=(Bit16s)host_readw((HostPt)&data[pos]);
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next_sample[0]=(Bit16s)host_readw((HostPt)&data[pos]);
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} else {
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nextSample[0]=(Bit32s)host_readd((HostPt)&data[pos]);
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next_sample[0]=(Bit32s)host_readd((HostPt)&data[pos]);
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}
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}
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}
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} else {
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if (stereo) {
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if (nativeorder) {
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nextSample[0]=(Bits)data[pos*2+0]-32768;
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nextSample[1]=(Bits)data[pos*2+1]-32768;
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next_sample[0]=(Bits)data[pos*2+0]-32768;
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next_sample[1]=(Bits)data[pos*2+1]-32768;
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} else {
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if ( sizeof( Type) == 2) {
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nextSample[0]=(Bits)host_readw((HostPt)&data[pos*2+0])-32768;
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nextSample[1]=(Bits)host_readw((HostPt)&data[pos*2+1])-32768;
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next_sample[0]=(Bits)host_readw((HostPt)&data[pos*2+0])-32768;
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next_sample[1]=(Bits)host_readw((HostPt)&data[pos*2+1])-32768;
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} else {
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nextSample[0]=(Bits)host_readd((HostPt)&data[pos*2+0])-32768;
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nextSample[1]=(Bits)host_readd((HostPt)&data[pos*2+1])-32768;
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next_sample[0]=(Bits)host_readd((HostPt)&data[pos*2+0])-32768;
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next_sample[1]=(Bits)host_readd((HostPt)&data[pos*2+1])-32768;
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}
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}
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} else {
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if (nativeorder) {
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nextSample[0]=(Bits)data[pos]-32768;
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next_sample[0]=(Bits)data[pos]-32768;
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} else {
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if ( sizeof( Type) == 2) {
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nextSample[0]=(Bits)host_readw((HostPt)&data[pos])-32768;
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next_sample[0]=(Bits)host_readw((HostPt)&data[pos])-32768;
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} else {
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nextSample[0]=(Bits)host_readd((HostPt)&data[pos])-32768;
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next_sample[0]=(Bits)host_readd((HostPt)&data[pos])-32768;
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}
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}
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}
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@ -277,19 +343,28 @@ inline void MixerChannel::AddSamples(Bitu len, const Type* data) {
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//This sample has been handled now, increase position
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pos++;
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}
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//Apply the left and right channel mappers only on write[..]
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//assignments. This ensures the channels are mapped only once
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//(avoiding double-swapping) and also minimizes the places where
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//we use our mapping variables as array indexes.
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//Note that volumes are independent of the channels mapping.
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const Bit8u left_map(channel_map[0]);
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const Bit8u right_map(channel_map[1]);
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//Where to write
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mixpos &= MIXER_BUFMASK;
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Bit32s* write = mixer.work[mixpos];
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if (!interpolate) {
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write[0] += prevSample[0] * volmul[0];
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write[1] += (stereo ? prevSample[1] : prevSample[0]) * volmul[1];
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write[0] += prev_sample[left_map] * volmul[0];
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write[1] += (stereo ? prev_sample[right_map] : prev_sample[left_map]) * volmul[1];
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}
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else {
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Bits diff_mul = freq_counter & FREQ_MASK;
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Bits sample = prevSample[0] + (((nextSample[0] - prevSample[0]) * diff_mul) >> FREQ_SHIFT);
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Bits sample = prev_sample[left_map] + (((next_sample[left_map] - prev_sample[left_map]) * diff_mul) >> FREQ_SHIFT);
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write[0] += sample*volmul[0];
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if (stereo) {
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sample = prevSample[1] + (((nextSample[1] - prevSample[1]) * diff_mul) >> FREQ_SHIFT);
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sample = prev_sample[right_map] + (((next_sample[right_map] - prev_sample[right_map]) * diff_mul) >> FREQ_SHIFT);
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}
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write[1] += sample*volmul[1];
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}
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@ -318,14 +393,14 @@ void MixerChannel::AddStretched(Bitu len,Bit16s * data) {
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if (pos != new_pos) {
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pos = new_pos;
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//Forward the previous sample
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prevSample[0] = data[0];
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prev_sample[0] = data[0];
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data++;
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}
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Bits diff = data[0] - prevSample[0];
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Bits diff = data[0] - prev_sample[0];
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Bits diff_mul = index & FREQ_MASK;
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index += index_add;
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mixpos &= MIXER_BUFMASK;
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Bits sample = prevSample[0] + ((diff * diff_mul) >> FREQ_SHIFT);
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Bits sample = prev_sample[0] + ((diff * diff_mul) >> FREQ_SHIFT);
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mixer.work[mixpos][0] += sample * volmul[0];
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mixer.work[mixpos][1] += sample * volmul[1];
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mixpos++;
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@ -702,7 +777,7 @@ void MIXER_Init(Section* sec) {
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mixer.tick_add=calc_tickadd(mixer.freq);
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TIMER_AddTickHandler(MIXER_Mix_NoSound);
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} else {
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if((mixer.freq != obtained.freq) || (mixer.blocksize != obtained.samples))
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if((static_cast<Bit16s>(mixer.freq) != obtained.freq) || (mixer.blocksize != obtained.samples))
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LOG_MSG("MIXER: Got different values from SDL: freq %d, blocksize %d",obtained.freq,obtained.samples);
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mixer.freq=obtained.freq;
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mixer.blocksize=obtained.samples;
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