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Diffstat (limited to 'engines/sci/sound/audio32.cpp')
| -rw-r--r-- | engines/sci/sound/audio32.cpp | 993 |
1 files changed, 993 insertions, 0 deletions
diff --git a/engines/sci/sound/audio32.cpp b/engines/sci/sound/audio32.cpp new file mode 100644 index 0000000000..0cf8e3cb13 --- /dev/null +++ b/engines/sci/sound/audio32.cpp @@ -0,0 +1,993 @@ +/* ScummVM - Graphic Adventure Engine + * + * ScummVM is the legal property of its developers, whose names + * are too numerous to list here. Please refer to the COPYRIGHT + * file distributed with this source distribution. + * + * This program is free software; you can redistribute it and/or + * modify it under the terms of the GNU General Public License + * as published by the Free Software Foundation; either version 2 + * of the License, or (at your option) any later version. + * + * This program is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the + * GNU General Public License for more details. + * + * You should have received a copy of the GNU General Public License + * along with this program; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301, USA. + * + */ + +#include "sci/sound/audio32.h" +#include "audio/audiostream.h" // for SeekableAudioStream +#include "audio/decoders/raw.h" // for makeRawStream, RawFlags::FLAG_16BITS +#include "audio/decoders/wave.h" // for makeWAVStream +#include "audio/rate.h" // for RateConverter, makeRateConverter +#include "audio/timestamp.h" // for Timestamp +#include "common/config-manager.h" // for ConfMan +#include "common/endian.h" // for MKTAG +#include "common/memstream.h" // for MemoryReadStream +#include "common/str.h" // for String +#include "common/stream.h" // for SeekableReadStream +#include "common/system.h" // for OSystem, g_system +#include "common/textconsole.h" // for warning +#include "common/types.h" // for Flag::NO +#include "engine.h" // for Engine, g_engine +#include "sci/engine/vm_types.h" // for reg_t, make_reg, NULL_REG +#include "sci/resource.h" // for ResourceId, ResourceType::kResour... +#include "sci/sci.h" // for SciEngine, g_sci, getSciVersion +#include "sci/sound/decoders/sol.h" // for makeSOLStream + +namespace Sci { + +bool detectSolAudio(Common::SeekableReadStream &stream) { + const size_t initialPosition = stream.pos(); + +// TODO: Resource manager for audio resources reads past the +// header so even though this is the detection algorithm +// in SSCI, ScummVM can't use it +#if 0 + byte header[6]; + if (stream.read(header, sizeof(header)) != sizeof(header)) { + stream.seek(initialPosition); + return false; + } + + stream.seek(initialPosition); + + if (header[0] != 0x8d || READ_BE_UINT32(header + 2) != MKTAG('S', 'O', 'L', 0)) { + return false; + } + + return true; +#else + byte header[4]; + if (stream.read(header, sizeof(header)) != sizeof(header)) { + stream.seek(initialPosition); + return false; + } + + stream.seek(initialPosition); + + if (READ_BE_UINT32(header) != MKTAG('S', 'O', 'L', 0)) { + return false; + } + + return true; +#endif +} + +bool detectWaveAudio(Common::SeekableReadStream &stream) { + const size_t initialPosition = stream.pos(); + + byte blockHeader[8]; + if (stream.read(blockHeader, sizeof(blockHeader)) != sizeof(blockHeader)) { + stream.seek(initialPosition); + return false; + } + + stream.seek(initialPosition); + const uint32 headerType = READ_BE_UINT32(blockHeader); + + if (headerType != MKTAG('R', 'I', 'F', 'F')) { + return false; + } + + return true; +} + +#pragma mark - + +Audio32::Audio32(ResourceManager *resMan) : + _resMan(resMan), + _mixer(g_system->getMixer()), + _handle(), + _mutex(), + + _numActiveChannels(0), + _inAudioThread(false), + + _globalSampleRate(44100), + _maxAllowedSampleRate(44100), + _globalBitDepth(16), + _maxAllowedBitDepth(16), + _globalNumOutputChannels(2), + _maxAllowedOutputChannels(2), + _preload(0), + + _robotAudioPaused(false), + + _pausedAtTick(0), + _startedAtTick(0), + + _attenuatedMixing(true), + + _monitoredChannelIndex(-1), + _monitoredBuffer(nullptr), + _monitoredBufferSize(0), + _numMonitoredSamples(0) { + + if (getSciVersion() < SCI_VERSION_3) { + _channels.resize(5); + } else { + _channels.resize(8); + } + + _useModifiedAttenuation = false; + if (getSciVersion() == SCI_VERSION_2_1_MIDDLE) { + switch (g_sci->getGameId()) { + case GID_MOTHERGOOSEHIRES: + case GID_PQ4: + case GID_QFG4: + case GID_SQ6: + _useModifiedAttenuation = true; + default: + break; + } + } else if (getSciVersion() == SCI_VERSION_2_1_EARLY && g_sci->getGameId() == GID_KQ7) { + // KQ7 1.51 uses the non-standard attenuation, but 2.00b + // does not, which is strange. + _useModifiedAttenuation = true; + } + + _mixer->playStream(Audio::Mixer::kSFXSoundType, &_handle, this, -1, Audio::Mixer::kMaxChannelVolume, 0, DisposeAfterUse::NO, true); +} + +Audio32::~Audio32() { + stop(kAllChannels); + _mixer->stopHandle(_handle); + free(_monitoredBuffer); +} + +#pragma mark - +#pragma mark AudioStream implementation + +int Audio32::writeAudioInternal(Audio::RewindableAudioStream *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, const bool loop) { + int samplesToRead = numSamples; + + // The parent rate converter will request N * 2 + // samples from this `readBuffer` call, because + // we tell it that we send stereo output, but + // the source stream we're mixing in may be + // mono, in which case we need to request half + // as many samples from the mono stream and let + // the converter double them for stereo output + if (!sourceStream->isStereo()) { + samplesToRead >>= 1; + } + + int samplesWritten = 0; + + do { + if (loop && sourceStream->endOfStream()) { + sourceStream->rewind(); + } + + const int loopSamplesWritten = converter->flow(*sourceStream, targetBuffer, samplesToRead, leftVolume, rightVolume); + + if (loopSamplesWritten == 0) { + break; + } + + samplesToRead -= loopSamplesWritten; + samplesWritten += loopSamplesWritten; + targetBuffer += loopSamplesWritten << 1; + } while (loop && samplesToRead > 0); + + if (!sourceStream->isStereo()) { + samplesWritten <<= 1; + } + + return samplesWritten; +} + +// 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 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. +// +// 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 *buffer, const int numSamples) { + Common::StackLock lock(_mutex); + + if (_pausedAtTick != 0 || _numActiveChannels == 0) { + 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 + _inAudioThread = true; + + freeUnusedChannels(); + + // 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 the volume of the target + // buffer when each new channel is mixed in. + // Instead of manipulating the content of the target + // buffer when mixing (which would either require + // modification of RateConverter or an expensive second + // pass against the entire target buffer), we just + // scale the volume for each channel in advance, with + // the earliest (lowest) channel having the highest + // amount of attenuation (lowest volume). + uint8 attenuationAmount; + uint8 attenuationStepAmount; + if (_useModifiedAttenuation) { + // channel | divisor + // 0 | 0 (>> 0) + // 1 | 4 (>> 2) + // 2 | 8... + attenuationAmount = _numActiveChannels * 2; + attenuationStepAmount = 2; + } else { + // channel | divisor + // 0 | 2 (>> 1) + // 1 | 4 (>> 2) + // 2 | 6... + if (_monitoredChannelIndex == -1 && _numActiveChannels > 1) { + attenuationAmount = _numActiveChannels + 1; + attenuationStepAmount = 1; + } else { + attenuationAmount = 0; + attenuationStepAmount = 0; + } + } + + int maxSamplesWritten = 0; + + for (int16 channelIndex = 0; channelIndex < _numActiveChannels; ++channelIndex) { + attenuationAmount -= attenuationStepAmount; + + const AudioChannel &channel = getChannel(channelIndex); + + if (channel.pausedAtTick || (channel.robot && _robotAudioPaused)) { + continue; + } + + // Channel finished fading and had the + // stopChannelOnFade flag set, so no longer exists + if (channel.fadeStartTick && processFade(channelIndex)) { + --channelIndex; + continue; + } + + if (channel.robot) { + // TODO: Robot audio into output buffer + continue; + } + + if (channel.vmd) { + // TODO: VMD audio into output buffer + continue; + } + + Audio::st_volume_t leftVolume, rightVolume; + + if (channel.pan == -1 || !isStereo()) { + leftVolume = rightVolume = channel.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; + } + + if (_monitoredChannelIndex == -1 && _attenuatedMixing) { + leftVolume >>= attenuationAmount; + rightVolume >>= attenuationAmount; + } + + 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; + } + + memset(_monitoredBuffer, 0, _monitoredBufferSize); + + _numMonitoredSamples = writeAudioInternal(channel.stream, channel.converter, _monitoredBuffer, numSamples, leftVolume, rightVolume, channel.loop); + + Audio::st_sample_t *sourceBuffer = _monitoredBuffer; + Audio::st_sample_t *targetBuffer = buffer; + const Audio::st_sample_t *const end = _monitoredBuffer + _numMonitoredSamples; + while (sourceBuffer != end) { + Audio::clampedAdd(*targetBuffer++, *sourceBuffer++); + } + + if (_numMonitoredSamples > maxSamplesWritten) { + maxSamplesWritten = _numMonitoredSamples; + } + } else if (!channel.stream->endOfStream() || channel.loop) { + if (_monitoredChannelIndex != -1) { + // Audio that is not on the monitored channel is silent + // when the monitored channel is active, but the stream still + // needs to be read in order to ensure that sound effects sync + // up once the monitored channel is turned off. The easiest + // way to guarantee this is to just do the normal channel read, + // but set the channel volume to zero so nothing is mixed in + leftVolume = rightVolume = 0; + } + + const int channelSamplesWritten = writeAudioInternal(channel.stream, channel.converter, buffer, numSamples, leftVolume, rightVolume, channel.loop); + if (channelSamplesWritten > maxSamplesWritten) { + maxSamplesWritten = channelSamplesWritten; + } + } + } + + _inAudioThread = false; + + return maxSamplesWritten; +} + +#pragma mark - +#pragma mark Channel management + +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. + + argc -= startIndex; + if (argc <= 0) { + return kAllChannels; + } + + Common::StackLock lock(_mutex); + + if (_numActiveChannels == 0) { + return kNoExistingChannel; + } + + ResourceId searchId; + + if (argc < 5) { + searchId = ResourceId(kResourceTypeAudio, argv[startIndex].toUint16()); + } else { + searchId = ResourceId( + kResourceTypeAudio36, + argv[startIndex].toUint16(), + argv[startIndex + 1].toUint16(), + argv[startIndex + 2].toUint16(), + argv[startIndex + 3].toUint16(), + argv[startIndex + 4].toUint16() + ); + } + + return findChannelById(searchId, soundNode); +} + +int16 Audio32::findChannelById(const ResourceId resourceId, const reg_t soundNode) const { + Common::StackLock lock(_mutex); + + if (_numActiveChannels == 0) { + return kNoExistingChannel; + } + + if (resourceId.getType() == kResourceTypeAudio) { + for (int16 i = 0; i < _numActiveChannels; ++i) { + const AudioChannel channel = _channels[i]; + if ( + channel.id == resourceId && + (soundNode.isNull() || soundNode == channel.soundNode) + ) { + return i; + } + } + } else if (resourceId.getType() == kResourceTypeAudio36) { + for (int16 i = 0; i < _numActiveChannels; ++i) { + const AudioChannel &candidate = getChannel(i); + if (!candidate.robot && candidate.id == resourceId) { + return i; + } + } + } else { + error("Audio32::findChannelById: Unknown resource type %d", resourceId.getType()); + } + + return kNoExistingChannel; +} + +void Audio32::freeUnusedChannels() { + Common::StackLock lock(_mutex); + for (int channelIndex = 0; channelIndex < _numActiveChannels; ++channelIndex) { + const AudioChannel &channel = getChannel(channelIndex); + if (channel.stream->endOfStream()) { + if (channel.loop) { + channel.stream->rewind(); + } else { + stop(channelIndex--); + } + } + } + + if (!_inAudioThread) { + unlockResources(); + } +} + +void Audio32::freeChannel(const int16 channelIndex) { + // The original engine 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 + // 4. Clear monitored memory buffer, if one existed + Common::StackLock lock(_mutex); + AudioChannel &channel = getChannel(channelIndex); + + // 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 { + _resMan->unlockResource(channel.resource); + } + + channel.resource = nullptr; + delete channel.stream; + channel.stream = nullptr; + delete channel.resourceStream; + channel.resourceStream = nullptr; + delete channel.converter; + channel.converter = nullptr; + + if (_monitoredChannelIndex == channelIndex) { + _monitoredChannelIndex = -1; + } +} + +void Audio32::unlockResources() { + Common::StackLock lock(_mutex); + assert(!_inAudioThread); + + for (UnlockList::const_iterator it = _resourcesToUnlock.begin(); it != _resourcesToUnlock.end(); ++it) { + _resMan->unlockResource(*it); + } + _resourcesToUnlock.clear(); +} + +#pragma mark - +#pragma mark Script compatibility + +void Audio32::setSampleRate(uint16 rate) { + if (rate > _maxAllowedSampleRate) { + rate = _maxAllowedSampleRate; + } + + _globalSampleRate = rate; +} + +void Audio32::setBitDepth(uint8 depth) { + if (depth > _maxAllowedBitDepth) { + depth = _maxAllowedBitDepth; + } + + _globalBitDepth = depth; +} + +void Audio32::setNumOutputChannels(int16 numChannels) { + if (numChannels > _maxAllowedOutputChannels) { + numChannels = _maxAllowedOutputChannels; + } + + _globalNumOutputChannels = numChannels; +} + +#pragma mark - +#pragma mark Playback + +uint16 Audio32::play(int16 channelIndex, const ResourceId resourceId, const bool autoPlay, const bool loop, const int16 volume, const reg_t soundNode, const bool monitor) { + Common::StackLock lock(_mutex); + + freeUnusedChannels(); + + if (channelIndex != kNoExistingChannel) { + AudioChannel &channel = getChannel(channelIndex); + + if (channel.pausedAtTick) { + resume(channelIndex); + return MIN(65534, 1 + channel.stream->getLength().msecs() * 60 / 1000); + } + + warning("Tried to resume channel %s that was not paused", channel.id.toString().c_str()); + return MIN(65534, 1 + channel.stream->getLength().msecs() * 60 / 1000); + } + + if (_numActiveChannels == _channels.size()) { + warning("Audio mixer is full when trying to play %s", resourceId.toString().c_str()); + return 0; + } + + // NOTE: SCI engine itself 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. + // + // 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. + // + // 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. + // + // 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. + Resource *resource = _resMan->findResource(resourceId, true); + if (resource == nullptr) { + return 0; + } + + channelIndex = _numActiveChannels++; + + AudioChannel &channel = getChannel(channelIndex); + channel.id = resourceId; + channel.resource = resource; + channel.loop = loop; + channel.robot = false; + channel.vmd = false; + 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) { + _monitoredChannelIndex = channelIndex; + } + + Common::MemoryReadStream headerStream(resource->_header, resource->_headerSize, DisposeAfterUse::NO); + Common::SeekableReadStream *dataStream = channel.resourceStream = resource->makeStream(); + + if (detectSolAudio(headerStream)) { + channel.stream = makeSOLStream(&headerStream, dataStream, DisposeAfterUse::NO); + } else if (detectWaveAudio(*dataStream)) { + channel.stream = Audio::makeWAVStream(dataStream, DisposeAfterUse::NO); + } else { + byte flags = Audio::FLAG_LITTLE_ENDIAN; + if (_globalBitDepth == 16) { + flags |= Audio::FLAG_16BITS; + } else { + flags |= Audio::FLAG_UNSIGNED; + } + + if (_globalNumOutputChannels == 2) { + flags |= Audio::FLAG_STEREO; + } + + channel.stream = Audio::makeRawStream(dataStream, _globalSampleRate, flags, DisposeAfterUse::NO); + } + + channel.converter = 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. + + channel.duration = /* round up */ 1 + (channel.stream->getLength().msecs() * 60 / 1000); + + const uint32 now = g_sci->getTickCount(); + channel.pausedAtTick = autoPlay ? 0 : now; + channel.startedAtTick = now; + + if (_numActiveChannels == 1) { + _startedAtTick = now; + } + + return channel.duration; +} + +bool Audio32::resume(const int16 channelIndex) { + if (channelIndex == kNoExistingChannel) { + return false; + } + + Common::StackLock lock(_mutex); + const uint32 now = g_sci->getTickCount(); + + if (channelIndex == kAllChannels) { + // Global pause in SSCI is an extra layer over + // individual channel pauses, so only unpause channels + // if there was not a global pause in place + if (_pausedAtTick == 0) { + return false; + } + + for (int i = 0; i < _numActiveChannels; ++i) { + AudioChannel &channel = getChannel(i); + if (!channel.pausedAtTick) { + channel.startedAtTick += now - _pausedAtTick; + } + } + + _startedAtTick += now - _pausedAtTick; + _pausedAtTick = 0; + return true; + } else if (channelIndex == kRobotChannel) { + for (int i = 0; i < _numActiveChannels; ++i) { + AudioChannel &channel = getChannel(i); + if (channel.robot) { + channel.startedAtTick += now - channel.pausedAtTick; + channel.pausedAtTick = 0; + // TODO: Robot + // StartRobot(); + return true; + } + } + } else { + AudioChannel &channel = getChannel(channelIndex); + if (channel.pausedAtTick) { + channel.startedAtTick += now - channel.pausedAtTick; + channel.pausedAtTick = 0; + return true; + } + } + + return false; +} + +bool Audio32::pause(const int16 channelIndex) { + if (channelIndex == kNoExistingChannel) { + return false; + } + + Common::StackLock lock(_mutex); + const uint32 now = g_sci->getTickCount(); + bool didPause = false; + + if (channelIndex == kAllChannels) { + if (_pausedAtTick == 0) { + _pausedAtTick = now; + didPause = true; + } + } else if (channelIndex == kRobotChannel) { + _robotAudioPaused = true; + for (int16 i = 0; i < _numActiveChannels; ++i) { + AudioChannel &channel = getChannel(i); + if (channel.robot) { + channel.pausedAtTick = now; + } + } + + // NOTE: The actual engine returns false here regardless of whether + // or not channels were paused + } else { + AudioChannel &channel = getChannel(channelIndex); + + if (channel.pausedAtTick == 0) { + channel.pausedAtTick = now; + didPause = true; + } + } + + return didPause; +} + +int16 Audio32::stop(const int16 channelIndex) { + Common::StackLock lock(_mutex); + const int16 oldNumChannels = _numActiveChannels; + + if (channelIndex == kNoExistingChannel || oldNumChannels == 0) { + return 0; + } + + if (channelIndex == kAllChannels) { + for (int i = 0; i < oldNumChannels; ++i) { + freeChannel(i); + } + _numActiveChannels = 0; + } else { + freeChannel(channelIndex); + --_numActiveChannels; + for (int i = channelIndex; i < oldNumChannels - 1; ++i) { + _channels[i] = _channels[i + 1]; + if (i + 1 == _monitoredChannelIndex) { + _monitoredChannelIndex = i; + } + } + } + + // NOTE: SSCI stops the DSP interrupt and frees the + // global decompression buffer here if there are no + // more active channels + + return oldNumChannels; +} + +int16 Audio32::getPosition(const int16 channelIndex) const { + Common::StackLock lock(_mutex); + if (channelIndex == kNoExistingChannel || _numActiveChannels == 0) { + return -1; + } + + // NOTE: 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 + if (channelIndex == kAllChannels) { + if (_pausedAtTick) { + position = _pausedAtTick - _startedAtTick; + } else { + position = now - _startedAtTick; + } + } else { + const AudioChannel &channel = getChannel(channelIndex); + + if (channel.pausedAtTick) { + position = channel.pausedAtTick - channel.startedAtTick; + } else if (_pausedAtTick) { + position = _pausedAtTick - channel.startedAtTick; + } else { + position = now - channel.startedAtTick; + } + } + + return MIN(position, 65534); +} + +void Audio32::setLoop(const int16 channelIndex, const bool loop) { + Common::StackLock lock(_mutex); + + if (channelIndex < 0 || channelIndex >= _numActiveChannels) { + return; + } + + AudioChannel &channel = getChannel(channelIndex); + channel.loop = loop; +} + +reg_t Audio32::kernelPlay(const bool autoPlay, const int argc, const reg_t *const argv) { + if (argc == 0) { + return make_reg(0, _numActiveChannels); + } + + const int16 channelIndex = findChannelByArgs(argc, argv, 0, NULL_REG); + ResourceId resourceId; + bool loop; + int16 volume; + bool monitor = false; + reg_t soundNode = NULL_REG; + + if (argc >= 5) { + resourceId = ResourceId(kResourceTypeAudio36, argv[0].toUint16(), argv[1].toUint16(), argv[2].toUint16(), argv[3].toUint16(), argv[4].toUint16()); + + 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. + loop = (bool)argv[5].toSint16(); + } + + if (argc < 7 || argv[6].toSint16() < 0 || argv[6].toSint16() > Audio32::kMaxVolume) { + volume = Audio32::kMaxVolume; + + if (argc >= 7) { + monitor = true; + } + } else { + volume = argv[6].toSint16(); + } + } else { + resourceId = ResourceId(kResourceTypeAudio, argv[0].toUint16()); + + if (argc < 2 || argv[1].toSint16() == 1) { + loop = false; + } else { + loop = (bool)argv[1].toSint16(); + } + + // TODO: SCI3 uses the 0x80 bit as a flag to + // indicate "priority channel", but the volume is clamped + // in this call to 0x7F so that flag never makes it into + // the audio subsystem + if (argc < 3 || argv[2].toSint16() < 0 || argv[2].toSint16() > Audio32::kMaxVolume) { + volume = Audio32::kMaxVolume; + + if (argc >= 3) { + monitor = true; + } + } else { + volume = argv[2].toSint16(); + } + + soundNode = argc == 4 ? argv[3] : NULL_REG; + } + + return make_reg(0, play(channelIndex, resourceId, autoPlay, loop, volume, soundNode, monitor)); +} + +#pragma mark - +#pragma mark Effects + +int16 Audio32::getVolume(const int16 channelIndex) const { + if (channelIndex < 0 || channelIndex >= _numActiveChannels) { + return _mixer->getChannelVolume(_handle) * kMaxVolume / Audio::Mixer::kMaxChannelVolume; + } + + Common::StackLock lock(_mutex); + return getChannel(channelIndex).volume; +} + +void Audio32::setVolume(const int16 channelIndex, int16 volume) { + volume = MIN((int16)kMaxVolume, volume); + if (channelIndex == kAllChannels) { + ConfMan.setInt("sfx_volume", volume * Audio::Mixer::kMaxChannelVolume / kMaxVolume); + ConfMan.setInt("speech_volume", volume * Audio::Mixer::kMaxChannelVolume / kMaxVolume); + _mixer->setChannelVolume(_handle, volume * Audio::Mixer::kMaxChannelVolume / kMaxVolume); + g_engine->syncSoundSettings(); + } else if (channelIndex != kNoExistingChannel) { + Common::StackLock lock(_mutex); + getChannel(channelIndex).volume = volume; + } +} + +bool Audio32::fadeChannel(const int16 channelIndex, const int16 targetVolume, const int16 speed, const int16 steps, const bool stopAfterFade) { + Common::StackLock lock(_mutex); + + if (channelIndex < 0 || channelIndex >= _numActiveChannels) { + return false; + } + + AudioChannel &channel = getChannel(channelIndex); + + if (channel.id.getType() != kResourceTypeAudio || channel.volume == targetVolume) { + return false; + } + + if (steps && speed) { + channel.fadeStartTick = g_sci->getTickCount(); + channel.fadeStartVolume = channel.volume; + channel.fadeTargetVolume = targetVolume; + channel.fadeDuration = speed * steps; + channel.stopChannelOnFade = stopAfterFade; + } else { + setVolume(channelIndex, targetVolume); + } + + return true; +} + +bool Audio32::processFade(const int16 channelIndex) { + Common::StackLock lock(_mutex); + AudioChannel &channel = getChannel(channelIndex); + + if (channel.fadeStartTick) { + const uint32 fadeElapsed = g_sci->getTickCount() - channel.fadeStartTick; + if (fadeElapsed > channel.fadeDuration) { + channel.fadeStartTick = 0; + if (channel.stopChannelOnFade) { + stop(channelIndex); + return true; + } else { + setVolume(channelIndex, channel.fadeTargetVolume); + } + return false; + } + + int volume; + if (channel.fadeStartVolume > channel.fadeTargetVolume) { + volume = channel.fadeStartVolume - fadeElapsed * (channel.fadeStartVolume - channel.fadeTargetVolume) / channel.fadeDuration; + } else { + volume = channel.fadeStartVolume + fadeElapsed * (channel.fadeTargetVolume - channel.fadeStartVolume) / channel.fadeDuration; + } + + setVolume(channelIndex, volume); + return false; + } + + return false; +} + +#pragma mark - +#pragma mark Signal monitoring + +bool Audio32::hasSignal() const { + Common::StackLock lock(_mutex); + + if (_monitoredChannelIndex == -1) { + return false; + } + + const Audio::st_sample_t *buffer = _monitoredBuffer; + const Audio::st_sample_t *const end = _monitoredBuffer + _numMonitoredSamples; + + while (buffer != end) { + const Audio::st_sample_t sample = *buffer++; + if (sample > 1280 || sample < -1280) { + return true; + } + } + + return false; +} + +} // End of namespace Sci |