Add sync audio for emscripten
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ee39e65061
commit
9b531afb0c
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@ -27,7 +27,7 @@
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#define DEF_PLAY_STOP(entity) \
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MRB_FUNCTION(audio_##entity##Play) \
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{ \
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return mrb_nil_value();char *filename; \
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char *filename; \
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mrb_int volume = 100; \
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mrb_int pitch = 100; \
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mrb_get_args(mrb, "z|ii", &filename, &volume, &pitch); \
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@ -36,7 +36,7 @@
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} \
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MRB_FUNCTION(audio_##entity##Stop) \
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{ \
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return mrb_nil_value();MRB_FUN_UNUSED_PARAM; \
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MRB_FUN_UNUSED_PARAM; \
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shState->audio().entity##Stop(); \
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return mrb_nil_value(); \
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}
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@ -44,7 +44,7 @@
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#define DEF_FADE(entity) \
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MRB_FUNCTION(audio_##entity##Fade) \
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{ \
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return mrb_nil_value();mrb_int time; \
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mrb_int time; \
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mrb_get_args(mrb, "i", &time); \
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shState->audio().entity##Fade(time); \
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return mrb_nil_value(); \
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@ -19,6 +19,10 @@
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** along with mkxp. If not, see <http://www.gnu.org/licenses/>.
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*/
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#ifdef __EMSCRIPTEN__
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#include "audio.h"
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#endif
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#include "graphics.h"
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#include "sharedstate.h"
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#include "binding-util.h"
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@ -30,6 +34,10 @@ MRB_FUNCTION(graphicsUpdate)
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shState->graphics().update();
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#ifdef __EMSCRIPTEN__
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shState->audio().update();
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#endif
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return mrb_nil_value();
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}
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139
src/alstream.cpp
139
src/alstream.cpp
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@ -302,8 +302,12 @@ void ALStream::startStream(float offset)
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startOffset = offset;
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procFrames = offset * source->sampleRate();
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#ifdef __EMSCRIPTEN__
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streamData();
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#else
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thread = createSDLThread
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<ALStream, &ALStream::streamData>(this, threadName);
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#endif
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}
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void ALStream::pauseStream()
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@ -362,7 +366,6 @@ void ALStream::streamData()
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{
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/* Fill up queue */
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bool firstBuffer = true;
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ALDataSource::Status status;
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if (threadTermReq)
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return;
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@ -406,74 +409,82 @@ void ALStream::streamData()
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/* Wait for buffers to be consumed, then
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* refill and queue them up again */
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#ifndef __EMSCRIPTEN__
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while (true)
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{
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shState->rtData().syncPoint.passSecondarySync();
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ALint procBufs = AL::Source::getProcBufferCount(alSrc);
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while (procBufs--)
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{
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if (threadTermReq)
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break;
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AL::Buffer::ID buf = AL::Source::unqueueBuffer(alSrc);
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/* If something went wrong, try again later */
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if (buf == AL::Buffer::ID(0))
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break;
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if (buf == lastBuf)
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{
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/* Reset the processed sample count so
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* querying the playback offset returns 0.0 again */
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procFrames = source->loopStartFrames();
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lastBuf = AL::Buffer::ID(0);
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}
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else
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{
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/* Add the frame count contained in this
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* buffer to the total count */
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ALint bits = AL::Buffer::getBits(buf);
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ALint size = AL::Buffer::getSize(buf);
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ALint chan = AL::Buffer::getChannels(buf);
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if (bits != 0 && chan != 0)
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procFrames += ((size / (bits / 8)) / chan);
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}
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if (sourceExhausted)
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continue;
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status = source->fillBuffer(buf);
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if (status == ALDataSource::Error)
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{
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sourceExhausted.set();
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return;
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}
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AL::Source::queueBuffer(alSrc, buf);
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/* In case of buffer underrun,
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* start playing again */
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if (AL::Source::getState(alSrc) == AL_STOPPED)
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AL::Source::play(alSrc);
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/* If this was the last buffer before the data
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* source loop wrapped around again, mark it as
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* such so we can catch it and reset the processed
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* sample count once it gets unqueued */
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if (status == ALDataSource::WrapAround)
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lastBuf = buf;
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if (status == ALDataSource::EndOfStream)
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sourceExhausted.set();
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}
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streamDevice();
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if (threadTermReq)
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break;
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SDL_Delay(AUDIO_SLEEP);
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}
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#endif
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}
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void ALStream::update() {
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if (threadTermReq)
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return;
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shState->rtData().syncPoint.passSecondarySync();
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ALint procBufs = AL::Source::getProcBufferCount(alSrc);
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while (procBufs--)
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{
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if (threadTermReq)
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break;
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AL::Buffer::ID buf = AL::Source::unqueueBuffer(alSrc);
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/* If something went wrong, try again later */
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if (buf == AL::Buffer::ID(0))
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break;
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if (buf == lastBuf)
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{
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/* Reset the processed sample count so
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* querying the playback offset returns 0.0 again */
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procFrames = source->loopStartFrames();
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lastBuf = AL::Buffer::ID(0);
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}
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else
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{
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/* Add the frame count contained in this
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* buffer to the total count */
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ALint bits = AL::Buffer::getBits(buf);
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ALint size = AL::Buffer::getSize(buf);
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ALint chan = AL::Buffer::getChannels(buf);
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if (bits != 0 && chan != 0)
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procFrames += ((size / (bits / 8)) / chan);
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}
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if (sourceExhausted)
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continue;
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status = source->fillBuffer(buf);
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if (status == ALDataSource::Error)
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{
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sourceExhausted.set();
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return;
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}
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AL::Source::queueBuffer(alSrc, buf);
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/* In case of buffer underrun,
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* start playing again */
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if (AL::Source::getState(alSrc) == AL_STOPPED)
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AL::Source::play(alSrc);
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/* If this was the last buffer before the data
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* source loop wrapped around again, mark it as
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* such so we can catch it and reset the processed
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* sample count once it gets unqueued */
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if (status == ALDataSource::WrapAround)
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lastBuf = buf;
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if (status == ALDataSource::EndOfStream)
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sourceExhausted.set();
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}
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}
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@ -23,13 +23,12 @@
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#define ALSTREAM_H
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#include "al-util.h"
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#include "aldatasource.h"
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#include "sdl-util.h"
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#include <string>
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#include <SDL_rwops.h>
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struct ALDataSource;
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#define STREAM_BUFS 3
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/* State-machine like audio playback stream.
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@ -104,6 +103,8 @@ struct ALStream
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float queryOffset();
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bool queryNativePitch();
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void update();
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private:
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void closeSource();
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void openSource(const std::string &filename);
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@ -117,6 +118,8 @@ private:
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/* thread func */
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void streamData();
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ALDataSource::Status status;
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};
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#endif // ALSTREAM_H
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@ -71,22 +71,27 @@ struct AudioPrivate
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syncPoint(rtData.syncPoint)
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{
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meWatch.state = MeNotPlaying;
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#ifndef __EMSCRIPTEN__
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meWatch.thread = createSDLThread
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<AudioPrivate, &AudioPrivate::meWatchFun>(this, "audio_mewatch");
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#endif
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}
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~AudioPrivate()
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{
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meWatch.termReq.set();
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#ifndef __EMSCRIPTEN__
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SDL_WaitThread(meWatch.thread, 0);
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#endif
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}
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void meWatchFun()
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{
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const float fadeOutStep = 1.f / (200 / AUDIO_SLEEP);
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const float fadeInStep = 1.f / (1000 / AUDIO_SLEEP);
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#ifndef __EMSCRIPTEN__
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while (true)
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#endif
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{
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syncPoint.passSecondarySync();
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@ -231,8 +236,9 @@ struct AudioPrivate
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break;
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}
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}
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#ifndef __EMSCRIPTEN__
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SDL_Delay(AUDIO_SLEEP);
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#endif
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}
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}
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};
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@ -247,6 +253,7 @@ void Audio::bgmPlay(const char *filename,
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int pitch,
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float pos)
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{
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printf("PLAYING BGM %s\n", filename);
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p->bgm.play(filename, volume, pitch, pos);
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}
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@ -333,4 +340,14 @@ void Audio::reset()
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p->se.stop();
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}
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void Audio::update()
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{
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#ifdef __EMSCRIPTEN__
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p->bgm.update();
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p->bgm.update();
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p->me.update();
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p->meWatchFun();
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#endif
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}
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Audio::~Audio() { delete p; }
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@ -69,6 +69,8 @@ public:
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void reset();
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void update();
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private:
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Audio(RGSSThreadData &rtData);
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~Audio();
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@ -203,12 +203,14 @@ void AudioStream::fadeOut(int duration)
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return;
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}
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#ifdef __EMSCRIPTEN__
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if (fade.thread)
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{
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fade.reqFini.set();
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SDL_WaitThread(fade.thread, 0);
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fade.thread = 0;
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}
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#endif
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fade.active.set();
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fade.msStep = 1.0f / duration;
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@ -216,8 +218,10 @@ void AudioStream::fadeOut(int duration)
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fade.reqTerm.clear();
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fade.startTicks = SDL_GetTicks();
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#ifndef __EMSCRIPTEN__
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fade.thread = createSDLThread
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<AudioStream, &AudioStream::fadeOutThread>(this, fade.threadName);
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#endif
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unlockStream();
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}
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@ -280,24 +284,38 @@ void AudioStream::finiFadeOutInt()
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void AudioStream::startFadeIn()
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{
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#ifdef __EMSCRIPTEN__
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/* Previous fadein should always be terminated in play() */
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assert(!fadeIn.thread);
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#endif
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fadeIn.rqFini.clear();
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fadeIn.rqTerm.clear();
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fadeIn.startTicks = SDL_GetTicks();
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#ifdef __EMSCRIPTEN__
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fadeInThread();
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#else
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fadeIn.thread = createSDLThread
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<AudioStream, &AudioStream::fadeInThread>(this, fadeIn.threadName);
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#endif
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}
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void AudioStream::fadeOutThread()
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{
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#ifndef __EMSCRIPTEN__
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while (true)
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#else
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if (fade.active)
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#endif
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{
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/* Just immediately terminate on request */
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if (fade.reqTerm)
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#ifdef __EMSCRIPTEN__
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return;
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#else
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break;
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#endif
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lockStream();
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setVolume(FadeOut, 1.0f);
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unlockStream();
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#ifdef __EMSCRIPTEN__
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fade.active.clear();
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return;
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#else
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break;
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#endif
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}
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setVolume(FadeOut, resVol);
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unlockStream();
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#ifndef __EMSCRIPTEN__
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SDL_Delay(AUDIO_SLEEP);
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#endif
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}
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#ifndef __EMSCRIPTEN__
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fade.active.clear();
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#endif
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}
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void AudioStream::fadeInThread()
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SDL_Delay(AUDIO_SLEEP);
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}
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}
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void AudioStream::update()
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{
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fadeOutThread();
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stream.update();
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}
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@ -140,6 +140,8 @@ struct AudioStream
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float playingOffset();
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void update();
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private:
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float volumes[VolumeTypeCount];
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void updateVolume();
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@ -46,6 +46,7 @@ struct SDLSoundSource : ALDataSource
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if (!sample)
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{
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SDL_RWclose(&ops);
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printf("ERROR SDL_sound: %s", Sound_GetError());
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throw Exception(Exception::SDLError, "SDL_sound: %s", Sound_GetError());
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}
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