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-rw-r--r--engines/sci/engine/ksound.cpp24
-rw-r--r--engines/sci/sound/audio32.cpp281
-rw-r--r--engines/sci/sound/audio32.h231
-rw-r--r--engines/sci/sound/soundcmd.cpp8
4 files changed, 242 insertions, 302 deletions
diff --git a/engines/sci/engine/ksound.cpp b/engines/sci/engine/ksound.cpp
index bbfadf1c53..4d171bdd1c 100644
--- a/engines/sci/engine/ksound.cpp
+++ b/engines/sci/engine/ksound.cpp
@@ -384,10 +384,6 @@ reg_t kDoAudioPosition(EngineState *s, int argc, reg_t *argv) {
}
reg_t kDoAudioRate(EngineState *s, int argc, reg_t *argv) {
- // NOTE: In the original engine this would set the hardware
- // DSP sampling rate; ScummVM mixer does not need this, so
- // we only store the value to satisfy engine compatibility.
-
if (argc > 0) {
const uint16 sampleRate = argv[0].toUint16();
if (sampleRate != 0) {
@@ -407,10 +403,6 @@ reg_t kDoAudioGetCapability(EngineState *s, int argc, reg_t *argv) {
}
reg_t kDoAudioBitDepth(EngineState *s, int argc, reg_t *argv) {
- // NOTE: In the original engine this would set the hardware
- // DSP bit depth; ScummVM mixer does not need this, so
- // we only store the value to satisfy engine compatibility.
-
if (argc > 0) {
const uint16 bitDepth = argv[0].toUint16();
if (bitDepth != 0) {
@@ -426,10 +418,6 @@ reg_t kDoAudioMixing(EngineState *s, int argc, reg_t *argv) {
}
reg_t kDoAudioChannels(EngineState *s, int argc, reg_t *argv) {
- // NOTE: In the original engine this would set the hardware
- // DSP stereo output; ScummVM mixer does not need this, so
- // we only store the value to satisfy engine compatibility.
-
if (argc > 0) {
const int16 numChannels = argv[0].toSint16();
if (numChannels != 0) {
@@ -441,11 +429,6 @@ reg_t kDoAudioChannels(EngineState *s, int argc, reg_t *argv) {
}
reg_t kDoAudioPreload(EngineState *s, int argc, reg_t *argv) {
- // NOTE: In the original engine this would cause audio
- // data for new channels to be preloaded to memory when
- // the channel was initialized; we do not need this, so
- // we only store the value to satisfy engine compatibility.
-
if (argc > 0) {
g_sci->_audio32->setPreload(argv[0].toUint16());
}
@@ -477,11 +460,8 @@ reg_t kDoAudioPanOff(EngineState *s, int argc, reg_t *argv) {
}
reg_t kSetLanguage(EngineState *s, int argc, reg_t *argv) {
- // This is used by script 90 of MUMG Deluxe from the main menu to toggle
- // the audio language between English and Spanish.
- // Basically, it instructs the interpreter to switch the audio resources
- // (resource.aud and associated map files) and load them from the "Spanish"
- // subdirectory instead.
+ // Used by script 90 of MUMG Deluxe from the main menu to toggle between
+ // English and Spanish.
const Common::String audioDirectory = s->_segMan->getString(argv[0]);
g_sci->getResMan()->changeAudioDirectory(audioDirectory);
return s->r_acc;
diff --git a/engines/sci/sound/audio32.cpp b/engines/sci/sound/audio32.cpp
index 59ebc86bf3..f9a0140323 100644
--- a/engines/sci/sound/audio32.cpp
+++ b/engines/sci/sound/audio32.cpp
@@ -153,6 +153,7 @@ Audio32::Audio32(ResourceManager *resMan) :
_handle(),
_mutex(),
+ _channels(getSciVersion() < SCI_VERSION_2_1_EARLY ? 10 : getSciVersion() < SCI_VERSION_3 ? 5 : 8),
_numActiveChannels(0),
_inAudioThread(false),
@@ -170,22 +171,10 @@ Audio32::Audio32(ResourceManager *resMan) :
_startedAtTick(0),
_attenuatedMixing(true),
+ _useModifiedAttenuation(g_sci->_features->usesModifiedAudioAttenuation()),
_monitoredChannelIndex(-1),
- _monitoredBuffer(nullptr),
- _monitoredBufferSize(0),
_numMonitoredSamples(0) {
-
- if (getSciVersion() < SCI_VERSION_2_1_EARLY) {
- _channels.resize(10);
- } else if (getSciVersion() < SCI_VERSION_3) {
- _channels.resize(5);
- } else {
- _channels.resize(8);
- }
-
- _useModifiedAttenuation = g_sci->_features->usesModifiedAudioAttenuation();
-
// In games where scripts premultiply master audio volumes into the volumes
// of the individual audio channels sent to the mixer, Audio32 needs to use
// the kPlainSoundType so that the master SFX volume is not applied twice.
@@ -202,15 +191,14 @@ Audio32::Audio32(ResourceManager *resMan) :
Audio32::~Audio32() {
stop(kAllChannels);
_mixer->stopHandle(_handle);
- free(_monitoredBuffer);
}
#pragma mark -
#pragma mark AudioStream implementation
-int Audio32::writeAudioInternal(Audio::AudioStream *const sourceStream, Audio::RateConverter *const converter, Audio::st_sample_t *targetBuffer, const int numSamples, const Audio::st_volume_t leftVolume, const Audio::st_volume_t rightVolume) {
+int Audio32::writeAudioInternal(Audio::AudioStream &sourceStream, Audio::RateConverter &converter, Audio::st_sample_t *targetBuffer, const int numSamples, const Audio::st_volume_t leftVolume, const Audio::st_volume_t rightVolume) {
const int samplePairsToRead = numSamples >> 1;
- const int samplePairsWritten = converter->flow(*sourceStream, targetBuffer, samplePairsToRead, leftVolume, rightVolume);
+ const int samplePairsWritten = converter.flow(sourceStream, targetBuffer, samplePairsToRead, leftVolume, rightVolume);
return samplePairsWritten << 1;
}
@@ -243,34 +231,28 @@ bool Audio32::channelShouldMix(const AudioChannel &channel) const {
return true;
}
-// In earlier versions of SCI32 engine, audio mixing is
-// split into three different functions.
+// In earlier versions of SCI32 engine, audio mixing is split into three
+// different functions.
//
-// The first function is called from the main game thread in
-// AsyncEventCheck; later versions of SSCI also call it when
-// getting the playback position. This function is
-// responsible for cleaning up finished channels and
-// filling active channel buffers with decompressed audio
-// matching the hardware output audio format so they can
-// just be copied into the main DAC buffer directly later.
+// The first function is called from the main game thread in AsyncEventCheck;
+// later versions of SSCI also call it when getting the playback position. This
+// function is responsible for cleaning up finished channels and filling active
+// channel buffers with decompressed audio matching the hardware output audio
+// format so they can just be copied into the main DAC buffer directly later.
//
-// The second function is called by the audio hardware when
-// the DAC buffer needs to be filled, and by `play` when
-// there is only one active sample (so it can just blow away
-// whatever was already in the DAC buffer). It merges all
-// active channels into the DAC buffer and then updates the
-// offset into the DAC buffer.
+// The second function is called by the audio hardware when the DAC buffer needs
+// to be filled, and by `play` when there is only one active sample (so it can
+// just blow away whatever was already in the DAC buffer). It merges all active
+// channels into the DAC buffer and then updates the offset into the DAC buffer.
//
-// Finally, a third function is called by the second
-// function, and it actually puts data into the DAC buffer,
-// performing volume, distortion, and balance adjustments.
+// Finally, a third function is called by the second function, and it actually
+// puts data into the DAC buffer, performing volume, distortion, and balance
+// adjustments.
//
-// Since we only have one callback from the audio thread,
-// and should be able to do all audio processing in
-// real time, and we have streams, and we do not need to
-// completely fill the audio buffer, the functionality of
-// all these original functions is combined here and
-// simplified.
+// Since we only have one callback from the audio thread, and should be able to
+// do all audio processing in real time, and we have streams, and we do not need
+// to completely fill the audio buffer, the functionality of all these original
+// functions is combined here and simplified.
int Audio32::readBuffer(Audio::st_sample_t *const buffer, const int numSamples) {
Common::StackLock lock(_mutex);
@@ -278,20 +260,17 @@ int Audio32::readBuffer(Audio::st_sample_t *const buffer, const int numSamples)
return 0;
}
- // ResourceManager is not thread-safe so we need to
- // avoid calling into it from the audio thread, but at
- // the same time we need to be able to clear out any
- // finished channels on a regular basis
+ // ResourceManager is not thread-safe so we need to avoid calling into it
+ // from the audio thread, but at the same time we need to be able to clear
+ // out any finished channels on a regular basis
_inAudioThread = true;
freeUnusedChannels();
const bool playOnlyMonitoredChannel = getSciVersion() != SCI_VERSION_3 && _monitoredChannelIndex != -1;
- // The caller of `readBuffer` is a rate converter,
- // which reuses (without clearing) an intermediate
- // buffer, so we need to zero the intermediate buffer
- // to prevent mixing into audio data from the last
- // callback.
+ // The caller of `readBuffer` is a rate converter, which reuses (without
+ // clearing) an intermediate buffer, so we need to zero the intermediate
+ // buffer to prevent mixing into audio data from the last callback.
memset(buffer, 0, numSamples * sizeof(Audio::st_sample_t));
// This emulates the attenuated mixing mode of SSCI engine, which reduces
@@ -340,8 +319,8 @@ int Audio32::readBuffer(Audio::st_sample_t *const buffer, const int numSamples)
continue;
}
- // Channel finished fading and had the
- // stopChannelOnFade flag set, so no longer exists
+ // Channel finished fading and had the stopChannelOnFade flag set, so no
+ // longer exists
if (channel.fadeStartTick && processFade(channelIndex)) {
--channelIndex;
continue;
@@ -352,10 +331,10 @@ int Audio32::readBuffer(Audio::st_sample_t *const buffer, const int numSamples)
if (channel.pan == -1 || !isStereo()) {
int volume = channel.volume;
if (getSciVersion() == SCI_VERSION_2) {
- // NOTE: In SSCI, audio is decompressed into a temporary
- // buffer, then the samples in that buffer are looped over,
- // shifting each sample right 3, 2, or 1 bits to reduce the
- // volume.
+ // In SSCI, audio is decompressed into a temporary buffer, then
+ // the samples in that buffer are looped over, shifting each
+ // sample right 3, 2, or 1 bits to reduce the volume within the
+ // ranges given here
if (volume > 0 && volume <= 42) {
volume = 15;
} else if (volume > 42 && volume <= 84) {
@@ -367,20 +346,18 @@ int Audio32::readBuffer(Audio::st_sample_t *const buffer, const int numSamples)
// In SCI3, granularity of the non-maximum volumes is 1/32
volume &= ~4;
- // NOTE: In the SSCI DOS interpreter, non-maximum volumes are
- // divided by 8 which puts them in a range of [0, 16). That
- // reduced volume range gets passed into a volume function which
- // expects values [0, 32). So, effectively, all non-maximum
- // volumes are half-volume in DOS in SCI3. In Windows, volumes
- // [120, 124) are the same as 127 due to a programming bug.
- // We do not emulate either of these incorrect behaviors.
+ // In the SSCI DOS interpreter, non-maximum volumes are divided
+ // by 8 which puts them in a range of [0, 16). That reduced
+ // volume range gets passed into a volume function which expects
+ // values [0, 32). So, effectively, all non-maximum volumes are
+ // half-volume in DOS in SCI3 due to a programming bug. In
+ // Windows, volumes [120, 124) are the same as 127 due to
+ // another programming bug. We do not emulate either of these
+ // incorrect behaviors.
}
leftVolume = rightVolume = volume * Audio::Mixer::kMaxChannelVolume / kMaxVolume;
} else {
- // TODO: This should match the SCI3 algorithm,
- // which seems to halve the volume of each
- // channel when centered; is this intended?
leftVolume = channel.volume * (100 - channel.pan) / 100 * Audio::Mixer::kMaxChannelVolume / kMaxVolume;
rightVolume = channel.volume * channel.pan / 100 * Audio::Mixer::kMaxChannelVolume / kMaxVolume;
}
@@ -397,19 +374,15 @@ int Audio32::readBuffer(Audio::st_sample_t *const buffer, const int numSamples)
}
if (channelIndex == _monitoredChannelIndex) {
- const size_t bufferSize = numSamples * sizeof(Audio::st_sample_t);
- if (_monitoredBufferSize < bufferSize) {
- _monitoredBuffer = (Audio::st_sample_t *)realloc(_monitoredBuffer, bufferSize);
- _monitoredBufferSize = bufferSize;
+ if (numSamples > (int)_monitoredBuffer.size()) {
+ _monitoredBuffer.resize(numSamples);
}
+ memset(_monitoredBuffer.data(), 0, _monitoredBuffer.size() * sizeof(Audio::st_sample_t));
+ _numMonitoredSamples = writeAudioInternal(*channel.stream, *channel.converter, _monitoredBuffer.data(), numSamples, leftVolume, rightVolume);
- memset(_monitoredBuffer, 0, _monitoredBufferSize);
-
- _numMonitoredSamples = writeAudioInternal(channel.stream, channel.converter, _monitoredBuffer, numSamples, leftVolume, rightVolume);
-
- Audio::st_sample_t *sourceBuffer = _monitoredBuffer;
+ Audio::st_sample_t *sourceBuffer = _monitoredBuffer.data();
Audio::st_sample_t *targetBuffer = buffer;
- const Audio::st_sample_t *const end = _monitoredBuffer + _numMonitoredSamples;
+ const Audio::st_sample_t *const end = _monitoredBuffer.data() + _numMonitoredSamples;
while (sourceBuffer != end) {
Audio::clampedAdd(*targetBuffer++, *sourceBuffer++);
}
@@ -428,7 +401,7 @@ int Audio32::readBuffer(Audio::st_sample_t *const buffer, const int numSamples)
leftVolume = rightVolume = 0;
}
- const int channelSamplesWritten = writeAudioInternal(channel.stream, channel.converter, buffer, numSamples, leftVolume, rightVolume);
+ const int channelSamplesWritten = writeAudioInternal(*channel.stream, *channel.converter, buffer, numSamples, leftVolume, rightVolume);
if (channelSamplesWritten > maxSamplesWritten) {
maxSamplesWritten = channelSamplesWritten;
}
@@ -458,10 +431,9 @@ uint8 Audio32::getNumUnlockedChannels() const {
}
int16 Audio32::findChannelByArgs(int argc, const reg_t *argv, const int startIndex, const reg_t soundNode) const {
- // NOTE: argc/argv are already reduced by one in our engine because
- // this call is always made from a subop, so no reduction for the
- // subop is made in this function. SSCI takes extra steps to skip
- // the subop argument.
+ // SSCI takes extra steps to skip the subop argument here, but argc/argv are
+ // already reduced by one in our engine by the kernel since these calls are
+ // always subops so we do not need to do anything extra
argc -= startIndex;
if (argc <= 0) {
@@ -501,10 +473,10 @@ int16 Audio32::findChannelById(const ResourceId resourceId, const reg_t soundNod
if (resourceId.getType() == kResourceTypeAudio) {
for (int16 i = 0; i < _numActiveChannels; ++i) {
- const AudioChannel channel = _channels[i];
+ const AudioChannel &candidate = _channels[i];
if (
- channel.id == resourceId &&
- (soundNode.isNull() || soundNode == channel.soundNode)
+ candidate.id == resourceId &&
+ (soundNode.isNull() || soundNode == candidate.soundNode)
) {
return i;
}
@@ -553,7 +525,7 @@ void Audio32::freeUnusedChannels() {
}
void Audio32::freeChannel(const int16 channelIndex) {
- // The original engine did this:
+ // SSCI did this:
// 1. Unlock memory-cached resource, if one existed
// 2. Close patched audio file descriptor, if one existed
// 3. Free decompression memory buffer, if one existed
@@ -563,14 +535,13 @@ void Audio32::freeChannel(const int16 channelIndex) {
// Robots have no corresponding resource to free
if (channel.robot) {
- delete channel.stream;
- channel.stream = nullptr;
+ channel.stream.reset();
channel.robot = false;
} else {
- // We cannot unlock resources from the audio thread
- // because ResourceManager is not thread-safe; instead,
- // we just record that the resource needs unlocking and
- // unlock it whenever we are on the main thread again
+ // We cannot unlock resources from the audio thread because
+ // ResourceManager is not thread-safe; instead, we just record that the
+ // resource needs unlocking and unlock it whenever we are on the main
+ // thread again
if (_inAudioThread) {
_resourcesToUnlock.push_back(channel.resource);
} else {
@@ -578,12 +549,10 @@ void Audio32::freeChannel(const int16 channelIndex) {
}
channel.resource = nullptr;
- delete channel.stream;
- channel.stream = nullptr;
+ channel.stream.reset();
}
- delete channel.converter;
- channel.converter = nullptr;
+ channel.converter.reset();
if (_monitoredChannelIndex == channelIndex) {
_monitoredChannelIndex = -1;
@@ -677,13 +646,12 @@ bool Audio32::playRobotAudio(const RobotAudioStream::RobotAudioPacket &packet) {
channel.pausedAtTick = 0;
channel.soundNode = NULL_REG;
channel.volume = kMaxVolume;
- // TODO: SCI3 introduces stereo audio
channel.pan = -1;
- channel.converter = Audio::makeRateConverter(RobotAudioStream::kRobotSampleRate, getRate(), false);
+ channel.converter.reset(Audio::makeRateConverter(RobotAudioStream::kRobotSampleRate, getRate(), false));
// The RobotAudioStream buffer size is
// ((bytesPerSample * channels * sampleRate * 2000ms) / 1000ms) & ~3
// where bytesPerSample = 2, channels = 1, and sampleRate = 22050
- channel.stream = new RobotAudioStream(88200);
+ channel.stream.reset(new RobotAudioStream(88200));
_robotAudioPaused = false;
if (_numActiveChannels == 1) {
@@ -691,7 +659,7 @@ bool Audio32::playRobotAudio(const RobotAudioStream::RobotAudioPacket &packet) {
}
}
- return static_cast<RobotAudioStream *>(channel.stream)->addPacket(packet);
+ return static_cast<RobotAudioStream *>(channel.stream.get())->addPacket(packet);
}
bool Audio32::queryRobotAudio(RobotAudioStream::StreamState &status) const {
@@ -703,7 +671,7 @@ bool Audio32::queryRobotAudio(RobotAudioStream::StreamState &status) const {
return false;
}
- status = static_cast<RobotAudioStream *>(getChannel(channelIndex).stream)->getStatus();
+ status = static_cast<RobotAudioStream *>(getChannel(channelIndex).stream.get())->getStatus();
return true;
}
@@ -715,7 +683,7 @@ bool Audio32::finishRobotAudio() {
return false;
}
- static_cast<RobotAudioStream *>(getChannel(channelIndex).stream)->finish();
+ static_cast<RobotAudioStream *>(getChannel(channelIndex).stream.get())->finish();
return true;
}
@@ -741,7 +709,7 @@ uint16 Audio32::play(int16 channelIndex, const ResourceId resourceId, const bool
if (channelIndex != kNoExistingChannel) {
AudioChannel &channel = getChannel(channelIndex);
- MutableLoopAudioStream *stream = dynamic_cast<MutableLoopAudioStream *>(channel.stream);
+ MutableLoopAudioStream *stream = dynamic_cast<MutableLoopAudioStream *>(channel.stream.get());
if (stream == nullptr) {
error("[Audio32::play]: Unable to cast stream for resource %s", resourceId.toString().c_str());
}
@@ -760,43 +728,42 @@ uint16 Audio32::play(int16 channelIndex, const ResourceId resourceId, const bool
return 0;
}
- // NOTE: SCI engine itself normally searches in this order:
+ // SSCI normally searches in this order:
//
// For Audio36:
//
- // 1. First, request a FD using Audio36 name and use it as the
- // source FD for reading the audio resource data.
- // 2a. If the returned FD is -1, or equals the audio map, or
- // equals the audio bundle, try to get the offset of the
- // data from the audio map, using the Audio36 name.
+ // 1. First, request a FD using Audio36 name and use it as the source FD for
+ // reading the audio resource data.
+ // 2a. If the returned FD is -1, or equals the audio map, or equals the
+ // audio bundle, try to get the offset of the data from the audio map,
+ // using the Audio36 name.
//
- // If the returned offset is -1, this is not a valid resource;
- // return 0. Otherwise, set the read offset for the FD to the
- // returned offset.
- // 2b. Otherwise, use the FD as-is (it is a patch file), with zero
- // offset, and record it separately so it can be closed later.
+ // If the returned offset is -1, this is not a valid resource; return 0.
+ // Otherwise, set the read offset for the FD to the returned offset.
+ // 2b. Otherwise, use the FD as-is (it is a patch file), with zero offset,
+ // and record it separately so it can be closed later.
//
// For plain audio:
//
- // 1. First, request an Audio resource from the resource cache. If
- // one does not exist, make the same request for a Wave resource.
- // 2a. If an audio resource was discovered, record its memory ID
- // and clear the streaming FD
- // 2b. Otherwise, request an Audio FD. If one does not exist, make
- // the same request for a Wave FD. If neither exist, this is not
- // a valid resource; return 0. Otherwise, use the returned FD as
- // the streaming ID and set the memory ID to null.
+ // 1. First, request an Audio resource from the resource cache. If one does
+ // not exist, make the same request for a Wave resource.
+ // 2a. If an audio resource was discovered, record its memory ID and clear
+ // the streaming FD
+ // 2b. Otherwise, request an Audio FD. If one does not exist, make the same
+ // request for a Wave FD. If neither exist, this is not a valid
+ // resource; return 0. Otherwise, use the returned FD as the streaming
+ // ID and set the memory ID to null.
//
// Once these steps are complete, the audio engine either has a file
- // descriptor + offset that it can use to read streamed audio, or it
- // has a memory ID that it can use to read cached audio.
+ // descriptor + offset that it can use to read streamed audio, or it has a
+ // memory ID that it can use to read cached audio.
//
- // Here in ScummVM we just ask the resource manager to give us the
- // resource and we get a seekable stream.
+ // Here in ScummVM we just ask the resource manager to give us the resource
+ // and we get a seekable stream.
- // TODO: This should be fixed to use streaming, which means
- // fixing the resource manager to allow streaming, which means
- // probably rewriting a bunch of the resource manager.
+ // TODO: This should be fixed to use streaming, which means fixing the
+ // resource manager to allow streaming, which means probably rewriting a
+ // bunch of the resource manager.
Resource *resource = _resMan->findResource(resourceId, true);
if (resource == nullptr) {
warning("[Audio32::play]: %s could not be found", resourceId.toString().c_str());
@@ -812,7 +779,6 @@ uint16 Audio32::play(int16 channelIndex, const ResourceId resourceId, const bool
channel.fadeStartTick = 0;
channel.soundNode = soundNode;
channel.volume = volume < 0 || volume > kMaxVolume ? (int)kMaxVolume : volume;
- // TODO: SCI3 introduces stereo audio
channel.pan = -1;
if (monitor) {
@@ -842,18 +808,17 @@ uint16 Audio32::play(int16 channelIndex, const ResourceId resourceId, const bool
audioStream = Audio::makeRawStream(dataStream, _globalSampleRate, flags, DisposeAfterUse::YES);
}
- channel.stream = new MutableLoopAudioStream(audioStream, loop);
- channel.converter = Audio::makeRateConverter(channel.stream->getRate(), getRate(), channel.stream->isStereo(), false);
+ channel.stream.reset(new MutableLoopAudioStream(audioStream, loop));
+ channel.converter.reset(Audio::makeRateConverter(channel.stream->getRate(), getRate(), channel.stream->isStereo(), false));
- // NOTE: SCI engine sets up a decompression buffer here for the audio
- // stream, plus writes information about the sample to the channel to
- // convert to the correct hardware output format, and allocates the
- // monitoring buffer to match the bitrate/samplerate/channels of the
- // original stream. We do not need to do any of these things since we
- // use audio streams, and allocate and fill the monitoring buffer
- // when reading audio data from the stream.
+ // SSCI sets up a decompression buffer here for the audio stream, plus
+ // writes information about the sample to the channel to convert to the
+ // correct hardware output format, and allocates the monitoring buffer to
+ // match the bitrate/samplerate/channels of the original stream. We do not
+ // need to do any of these things since we use audio streams, and allocate
+ // and fill the monitoring buffer when reading audio data from the stream.
- MutableLoopAudioStream *stream = dynamic_cast<MutableLoopAudioStream *>(channel.stream);
+ MutableLoopAudioStream *stream = dynamic_cast<MutableLoopAudioStream *>(channel.stream.get());
if (stream == nullptr) {
error("[Audio32::play]: Unable to cast stream for resource %s", resourceId.toString().c_str());
}
@@ -958,8 +923,8 @@ bool Audio32::pause(const int16 channelIndex) {
}
}
- // NOTE: The actual engine returns false here regardless of whether
- // or not channels were paused
+ // SSCI returns false here regardless of whether or not channels were
+ // paused, so we emulate this behaviour
} else {
AudioChannel &channel = getChannel(channelIndex);
@@ -996,9 +961,10 @@ int16 Audio32::stop(const int16 channelIndex) {
}
}
- // NOTE: SSCI stops the DSP interrupt and frees the
- // global decompression buffer here if there are no
- // more active channels
+ // SSCI stops the DSP interrupt and frees the global decompression buffer
+ // here if there are no more active channels, which we do not need to do
+ // since the system manages audio callbacks and we have no static
+ // decompression buffer
return oldNumChannels;
}
@@ -1015,13 +981,14 @@ int16 Audio32::getPosition(const int16 channelIndex) const {
return -1;
}
- // NOTE: SSCI treats this as an unsigned short except for
- // when the value is 65535, then it treats it as signed
+ // SSCI treats this as an unsigned short except for when the value is 65535,
+ // then it treats it as signed
int position = -1;
const uint32 now = g_sci->getTickCount();
- // NOTE: The original engine also queried the audio driver to see whether
- // it thought that there was audio playback occurring via driver opcode 9
+ // SSCI also queried the audio driver to see whether it thought that there
+ // was audio playback occurring via driver opcode 9, but we have no analogue
+ // to query
if (channelIndex == kAllChannels) {
if (_pausedAtTick) {
position = _pausedAtTick - _startedAtTick;
@@ -1052,7 +1019,7 @@ void Audio32::setLoop(const int16 channelIndex, const bool loop) {
AudioChannel &channel = getChannel(channelIndex);
- MutableLoopAudioStream *stream = dynamic_cast<MutableLoopAudioStream *>(channel.stream);
+ MutableLoopAudioStream *stream = dynamic_cast<MutableLoopAudioStream *>(channel.stream.get());
assert(stream);
stream->loop() = loop;
}
@@ -1161,8 +1128,8 @@ bool Audio32::hasSignal() const {
return false;
}
- const Audio::st_sample_t *buffer = _monitoredBuffer;
- const Audio::st_sample_t *const end = _monitoredBuffer + _numMonitoredSamples;
+ const Audio::st_sample_t *buffer = _monitoredBuffer.data();
+ const Audio::st_sample_t *const end = _monitoredBuffer.data() + _numMonitoredSamples;
while (buffer != end) {
const Audio::st_sample_t sample = *buffer++;
@@ -1193,9 +1160,9 @@ reg_t Audio32::kernelPlay(const bool autoPlay, const int argc, const reg_t *cons
if (argc < 6 || argv[5].toSint16() == 1) {
loop = false;
} else {
- // NOTE: Uses -1 for infinite loop. Presumably the
- // engine was supposed to allow counter loops at one
- // point, but ended up only using loop as a boolean.
+ // SSCI uses -1 for infinite loop. Presumably the engine was
+ // supposed to allow counter loops at one point, but ended up only
+ // using loop as a boolean.
loop = (bool)argv[5].toSint16();
}
@@ -1311,8 +1278,10 @@ reg_t Audio32::kernelFade(const int argc, const reg_t *const argv) {
Common::StackLock lock(_mutex);
- // NOTE: In SSCI, this call to find the channel is hacked up; argc is
- // set to 2 before the call, and then restored after the call.
+ // In SSCI, this call to find the channel is hacked up; argc is set to 2
+ // before the call, and then restored after the call. We just implemented
+ // findChannelByArgs in a manner that allows us to pass this information
+ // without messing with argc/argv instead
const int16 channelIndex = findChannelByArgs(2, argv, 0, argc > 5 ? argv[5] : NULL_REG);
const int16 volume = argv[1].toSint16();
const int16 speed = argv[2].toSint16();
@@ -1359,7 +1328,7 @@ void Audio32::printAudioList(Console *con) const {
Common::StackLock lock(_mutex);
for (int i = 0; i < _numActiveChannels; ++i) {
const AudioChannel &channel = _channels[i];
- const MutableLoopAudioStream *stream = dynamic_cast<MutableLoopAudioStream *>(channel.stream);
+ const MutableLoopAudioStream *stream = dynamic_cast<MutableLoopAudioStream *>(channel.stream.get());
con->debugPrintf(" %d[%04x:%04x]: %s, started at %d, pos %d/%d, vol %d, pan %d%s%s\n",
i,
PRINT_REG(channel.soundNode),
diff --git a/engines/sci/sound/audio32.h b/engines/sci/sound/audio32.h
index 71f5883541..a0167f3d2d 100644
--- a/engines/sci/sound/audio32.h
+++ b/engines/sci/sound/audio32.h
@@ -50,22 +50,23 @@ struct AudioChannel {
ResourceId id;
/**
- * The resource loaded into this channel.
+ * The resource loaded into this channel. The resource is owned by
+ * ResourceManager.
*/
Resource *resource;
/**
- * The audio stream loaded into this channel. Can cast
- * to `SeekableAudioStream` for normal channels and
- * `RobotAudioStream` for robot channels.
+ * The audio stream loaded into this channel. Can cast to
+ * `SeekableAudioStream` for normal channels and `RobotAudioStream` for
+ * robot channels.
*/
- Audio::AudioStream *stream;
+ Common::ScopedPtr<Audio::AudioStream> stream;
/**
- * The converter used to transform and merge the input
- * stream into the mixer's output buffer.
+ * The converter used to transform and merge the input stream into the
+ * mixer's output buffer.
*/
- Audio::RateConverter *converter;
+ Common::ScopedPtr<Audio::RateConverter> converter;
/**
* Duration of the channel, in ticks.
@@ -83,8 +84,8 @@ struct AudioChannel {
uint32 pausedAtTick;
/**
- * The time, in ticks, that the channel fade began.
- * If 0, the channel is not being faded.
+ * The time, in ticks, that the channel fade began. If 0, the channel is not
+ * being faded.
*/
uint32 fadeStartTick;
@@ -104,20 +105,19 @@ struct AudioChannel {
int fadeTargetVolume;
/**
- * Whether or not the channel should be stopped and
- * freed when the fade is complete.
+ * Whether or not the channel should be stopped and freed when the fade is
+ * complete.
*/
bool stopChannelOnFade;
/**
- * Whether or not this channel contains a Robot
- * audio block.
+ * Whether or not this channel contains a Robot audio block.
*/
bool robot;
/**
- * For digital sound effects, the related VM
- * Sound::nodePtr object for the sound.
+ * For digital sound effects, the related VM Sound::nodePtr object for the
+ * sound.
*/
reg_t soundNode;
@@ -127,17 +127,37 @@ struct AudioChannel {
int volume;
/**
- * The amount to pan to the right, from 0 to 100.
- * 50 is centered, -1 is not panned.
+ * The amount to pan to the right, from 0 to 100. 50 is centered, -1 is not
+ * panned.
*/
int pan;
+
+ AudioChannel &operator=(AudioChannel &other) {
+ id = other.id;
+ resource = other.resource;
+ stream.reset(other.stream.release());
+ converter.reset(other.converter.release());
+ duration = other.duration;
+ startedAtTick = other.startedAtTick;
+ pausedAtTick = other.pausedAtTick;
+ fadeStartTick = other.fadeStartTick;
+ fadeStartVolume = other.fadeStartVolume;
+ fadeDuration = other.fadeDuration;
+ fadeTargetVolume = other.fadeTargetVolume;
+ stopChannelOnFade = other.stopChannelOnFade;
+ robot = other.robot;
+ soundNode = other.soundNode;
+ volume = other.volume;
+ pan = other.pan;
+ return *this;
+ }
};
#pragma mark -
/**
- * Special audio channel indexes used to select a channel
- * for digital audio playback.
+ * Special audio channel indexes used to select a channel for digital audio
+ * playback.
*/
enum AudioChannelIndex {
kRobotChannel = -3,
@@ -146,10 +166,9 @@ enum AudioChannelIndex {
};
/**
- * Audio32 acts as a permanent audio stream into the system
- * mixer and provides digital audio services for the SCI32
- * engine, since the system mixer does not support all the
- * features of SCI.
+ * Audio32 acts as a permanent audio stream into the system mixer and provides
+ * digital audio services for the SCI32 engine, since the system mixer does not
+ * support all the features of SCI.
*/
class Audio32 : public Audio::AudioStream, public Common::Serializable {
public:
@@ -196,10 +215,10 @@ private:
bool channelShouldMix(const AudioChannel &channel) const;
/**
- * Mixes audio from the given source stream into the
- * target buffer using the given rate converter.
+ * Mixes audio from the given source stream into the target buffer using the
+ * given rate converter.
*/
- int writeAudioInternal(Audio::AudioStream *const sourceStream, Audio::RateConverter *const converter, Audio::st_sample_t *targetBuffer, const int numSamples, const Audio::st_volume_t leftVolume, const Audio::st_volume_t rightVolume);
+ int writeAudioInternal(Audio::AudioStream &sourceStream, Audio::RateConverter &converter, Audio::st_sample_t *targetBuffer, const int numSamples, const Audio::st_volume_t leftVolume, const Audio::st_volume_t rightVolume);
#pragma mark -
#pragma mark Channel management
@@ -226,17 +245,15 @@ public:
uint8 getNumUnlockedChannels() const;
/**
- * Finds a channel that is already configured for the
- * given audio sample.
+ * Finds a channel that is already configured for the given audio sample.
*
- * @param startIndex The location of the audio resource
- * information in the arguments list.
+ * @param startIndex The location of the audio resource information in the
+ * arguments list.
*/
int16 findChannelByArgs(int argc, const reg_t *argv, const int startIndex, const reg_t soundNode) const;
/**
- * Finds a channel that is already configured for the
- * given audio sample.
+ * Finds a channel that is already configured for the given audio sample.
*/
int16 findChannelById(const ResourceId resourceId, const reg_t soundNode = NULL_REG) const;
@@ -255,26 +272,24 @@ private:
Common::Array<AudioChannel> _channels;
/**
- * The number of active audio channels in the mixer.
- * Being active is not the same as playing; active
- * channels may be paused.
+ * The number of active audio channels in the mixer. Being active is not the
+ * same as playing; active channels may be paused.
*/
uint8 _numActiveChannels;
/**
* Whether or not we are in the audio thread.
*
- * This flag is used instead of passing a parameter to
- * `freeUnusedChannels` because a parameter would
- * require forwarding through the public method `stop`,
- * and there is not currently any reason for this
- * implementation detail to be exposed.
+ * This flag is used instead of passing a parameter to `freeUnusedChannels`
+ * because a parameter would require forwarding through the public method
+ * `stop`, and there is not currently any reason for this implementation
+ * detail to be exposed.
*/
bool _inAudioThread;
/**
- * The list of resources from freed channels that need
- * to be unlocked from the main thread.
+ * The list of resources from freed channels that need to be unlocked from
+ * the main thread.
*/
UnlockList _resourcesToUnlock;
@@ -302,8 +317,7 @@ private:
}
/**
- * Frees all non-looping channels that have reached the
- * end of their stream.
+ * Frees all non-looping channels that have reached the end of their stream.
*/
void freeUnusedChannels();
@@ -313,8 +327,7 @@ private:
void freeChannel(const int16 channelIndex);
/**
- * Unlocks all resources that were freed by the audio
- * thread.
+ * Unlocks all resources that were freed by the audio thread.
*/
void unlockResources();
@@ -322,58 +335,58 @@ private:
#pragma mark Script compatibility
public:
/**
- * Gets the (fake) sample rate of the hardware DAC.
- * For script compatibility only.
+ * Gets the (fake) sample rate of the hardware DAC. For script compatibility
+ * only.
*/
inline uint16 getSampleRate() const {
return _globalSampleRate;
}
/**
- * Sets the (fake) sample rate of the hardware DAC.
- * For script compatibility only.
+ * Sets the (fake) sample rate of the hardware DAC. For script compatibility
+ * only.
*/
void setSampleRate(uint16 rate);
/**
- * Gets the (fake) bit depth of the hardware DAC.
- * For script compatibility only.
+ * Gets the (fake) bit depth of the hardware DAC. For script compatibility
+ * only.
*/
inline uint8 getBitDepth() const {
return _globalBitDepth;
}
/**
- * Sets the (fake) sample rate of the hardware DAC.
- * For script compatibility only.
+ * Sets the (fake) sample rate of the hardware DAC. For script compatibility
+ * only.
*/
void setBitDepth(uint8 depth);
/**
- * Gets the (fake) number of output (speaker) channels
- * of the hardware DAC. For script compatibility only.
+ * Gets the (fake) number of output (speaker) channels of the hardware DAC.
+ * For script compatibility only.
*/
inline uint8 getNumOutputChannels() const {
return _globalNumOutputChannels;
}
/**
- * Sets the (fake) number of output (speaker) channels
- * of the hardware DAC. For script compatibility only.
+ * Sets the (fake) number of output (speaker) channels of the hardware DAC.
+ * For script compatibility only.
*/
void setNumOutputChannels(int16 numChannels);
/**
- * Gets the (fake) number of preloaded channels.
- * For script compatibility only.
+ * Gets the (fake) number of preloaded channels. For script compatibility
+ * only.
*/
inline uint8 getPreload() const {
return _preload;
}
/**
- * Sets the (fake) number of preloaded channels.
- * For script compatibility only.
+ * Sets the (fake) number of preloaded channels. For script compatibility
+ * only.
*/
inline void setPreload(uint8 preload) {
_preload = preload;
@@ -381,49 +394,43 @@ public:
private:
/**
- * The hardware DAC sample rate. Stored only for script
- * compatibility.
+ * The hardware DAC sample rate. Stored only for script compatibility.
*/
uint16 _globalSampleRate;
/**
- * The maximum allowed sample rate of the system mixer.
- * Stored only for script compatibility.
+ * The maximum allowed sample rate of the system mixer. Stored only for
+ * script compatibility.
*/
uint16 _maxAllowedSampleRate;
/**
- * The hardware DAC bit depth. Stored only for script
- * compatibility.
+ * The hardware DAC bit depth. Stored only for script compatibility.
*/
uint8 _globalBitDepth;
/**
- * The maximum allowed bit depth of the system mixer.
- * Stored only for script compatibility.
+ * The maximum allowed bit depth of the system mixer. Stored only for script
+ * compatibility.
*/
uint8 _maxAllowedBitDepth;
/**
- * The hardware DAC output (speaker) channel
- * configuration. Stored only for script compatibility.
+ * The hardware DAC output (speaker) channel configuration. Stored only for
+ * script compatibility.
*/
uint8 _globalNumOutputChannels;
/**
- * The maximum allowed number of output (speaker)
- * channels of the system mixer. Stored only for script
- * compatibility.
+ * The maximum allowed number of output (speaker) channels of the system
+ * mixer. Stored only for script compatibility.
*/
uint8 _maxAllowedOutputChannels;
/**
- * The number of audio channels that should have their
- * data preloaded into memory instead of streaming from
- * disk.
- * 1 = all channels, 2 = 2nd active channel and above,
- * etc.
- * Stored only for script compatibility.
+ * The number of audio channels that should have their data preloaded into
+ * memory instead of streaming from disk. 1 = all channels, 2 = 2nd active
+ * channel and above, etc. Stored only for script compatibility.
*/
uint8 _preload;
@@ -442,8 +449,7 @@ private:
int16 findRobotChannel() const;
/**
- * When true, channels marked as robot audio will not be
- * played.
+ * When true, channels marked as robot audio will not be played.
*/
bool _robotAudioPaused;
@@ -456,8 +462,8 @@ public:
uint16 play(int16 channelIndex, const ResourceId resourceId, const bool autoPlay, const bool loop, const int16 volume, const reg_t soundNode, const bool monitor);
/**
- * Resumes playback of a paused audio channel, or of
- * the entire audio player.
+ * Resumes playback of a paused audio channel, or of the entire audio
+ * player.
*/
bool resume(const int16 channelIndex);
bool resume(const ResourceId resourceId, const reg_t soundNode = NULL_REG) {
@@ -475,8 +481,7 @@ public:
}
/**
- * Stops and unloads an audio channel, or the entire
- * audio player.
+ * Stops and unloads an audio channel, or the entire audio player.
*/
int16 stop(const int16 channelIndex);
int16 stop(const ResourceId resourceId, const reg_t soundNode = NULL_REG) {
@@ -490,8 +495,7 @@ public:
uint16 restart(const ResourceId resourceId, const bool autoPlay, const bool loop, const int16 volume, const reg_t soundNode, const bool monitor);
/**
- * Returns the playback position for the given channel
- * number, in ticks.
+ * Returns the playback position for the given channel number, in ticks.
*/
int16 getPosition(const int16 channelIndex) const;
int16 getPosition(const ResourceId resourceId, const reg_t soundNode = NULL_REG) {
@@ -531,8 +535,8 @@ private:
#pragma mark Effects
public:
/**
- * Gets the volume for a given channel. Passing
- * `kAllChannels` will get the global volume.
+ * Gets the volume for a given channel. Passing `kAllChannels` will get the
+ * global volume.
*/
int16 getVolume(const int16 channelIndex) const;
int16 getVolume(const ResourceId resourceId, const reg_t soundNode) const {
@@ -541,8 +545,8 @@ public:
}
/**
- * Sets the volume of an audio channel. Passing
- * `kAllChannels` will set the global volume.
+ * Sets the volume of an audio channel. Passing `kAllChannels` will set the
+ * global volume.
*/
void setVolume(const int16 channelIndex, int16 volume);
void setVolume(const ResourceId resourceId, const reg_t soundNode, const int16 volume) {
@@ -583,24 +587,21 @@ public:
private:
/**
- * If true, audio will be mixed by reducing the target
- * buffer by half every time a new channel is mixed in.
- * The final channel is not attenuated.
+ * If true, audio will be mixed by reducing the target buffer by half every
+ * time a new channel is mixed in. The final channel is not attenuated.
*/
bool _attenuatedMixing;
/**
- * When true, a modified attenuation algorithm is used
- * (`A/4 + B`) instead of standard linear attenuation
- * (`A/2 + B/2`).
+ * When true, a modified attenuation algorithm is used (`A/4 + B`) instead
+ * of standard linear attenuation (`A/2 + B/2`).
*/
bool _useModifiedAttenuation;
/**
* Processes an audio fade for the given channel.
*
- * @returns true if the fade was completed and the
- * channel was stopped.
+ * @returns true if the fade was completed and the channel was stopped.
*/
bool processFade(const int16 channelIndex);
@@ -608,35 +609,27 @@ private:
#pragma mark Signal monitoring
public:
/**
- * Returns whether the currently monitored audio channel
- * contains any signal within the next audio frame.
+ * Returns whether the currently monitored audio channel contains any signal
+ * within the next audio frame.
*/
bool hasSignal() const;
private:
/**
- * The index of the channel being monitored for signal,
- * or -1 if no channel is monitored. When a channel is
- * monitored, it also causes the engine to play only the
- * monitored channel.
+ * The index of the channel being monitored for signal, or -1 if no channel
+ * is monitored. When a channel is monitored, it also causes the engine to
+ * play only the monitored channel.
*/
int16 _monitoredChannelIndex;
/**
- * The data buffer holding decompressed audio data for
- * the channel that will be monitored for an audio
- * signal.
- */
- Audio::st_sample_t *_monitoredBuffer;
-
- /**
- * The size of the buffer, in bytes.
+ * The data buffer holding decompressed audio data for the channel that will
+ * be monitored for an audio signal.
*/
- size_t _monitoredBufferSize;
+ Common::Array<Audio::st_sample_t> _monitoredBuffer;
/**
- * The number of valid audio samples in the signal
- * monitoring buffer.
+ * The number of valid audio samples in the signal monitoring buffer.
*/
int _numMonitoredSamples;
diff --git a/engines/sci/sound/soundcmd.cpp b/engines/sci/sound/soundcmd.cpp
index 0bb295d41e..5a1c2c0825 100644
--- a/engines/sci/sound/soundcmd.cpp
+++ b/engines/sci/sound/soundcmd.cpp
@@ -350,11 +350,9 @@ reg_t SoundCommandParser::kDoSoundPause(EngineState *s, int argc, reg_t *argv) {
}
#ifdef ENABLE_SCI32
- // NOTE: The original engine also expected a global
- // "kernel call" flag to be true in order to perform
- // this action, but the architecture of the ScummVM
- // implementation is so different that it doesn't
- // matter here
+ // SSCI also expected a global "kernel call" flag to be true in order to
+ // perform this action, but the architecture of the ScummVM
+ // implementation is so different that it doesn't matter here
if (_soundVersion >= SCI_VERSION_2_1_EARLY && musicSlot->isSample) {
if (shouldPause) {
g_sci->_audio32->pause(ResourceId(kResourceTypeAudio, musicSlot->resourceId), musicSlot->soundObj);
generated by cgit v1.2.3 (git 2.25.1) at 2024-11-26 21:57:23 +0000