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-rw-r--r--engines/sci/sound/audio32.cpp281
1 files changed, 125 insertions, 156 deletions
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),
generated by cgit v1.2.3 (git 2.25.1) at 2024-07-06 17:35:14 +0000