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|
/* 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/console.h" // for Console
#include "sci/engine/features.h" // for GameFeatures
#include "sci/engine/guest_additions.h" // for GuestAdditions
#include "sci/engine/state.h" // for EngineState
#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 {
class MutableLoopAudioStream : public Audio::AudioStream {
public:
MutableLoopAudioStream(Audio::RewindableAudioStream *stream, const bool loop_, const DisposeAfterUse::Flag dispose = DisposeAfterUse::YES) :
_stream(stream, dispose),
_loop(loop_) {}
virtual int readBuffer(int16 *buffer, int numSamples) override {
int totalSamplesRead = 0;
int samplesRead;
do {
if (_loop && _stream->endOfStream()) {
_stream->rewind();
}
samplesRead = _stream->readBuffer(buffer, numSamples);
totalSamplesRead += samplesRead;
numSamples -= samplesRead;
} while (samplesRead > 0 && _loop && numSamples > 0);
return totalSamplesRead;
}
virtual bool isStereo() const override {
return _stream->isStereo();
}
virtual int getRate() const override {
return _stream->getRate();
}
virtual bool endOfData() const override {
return !_loop && _stream->endOfData();
}
virtual bool endOfStream() const override {
return !_loop && _stream->endOfStream();
}
bool &loop() {
return _loop;
}
bool loop() const {
return _loop;
}
virtual Audio::Timestamp getLength() const {
Audio::SeekableAudioStream *stream = dynamic_cast<Audio::SeekableAudioStream *>(_stream.get());
if (stream == nullptr) {
error("Cannot get length from a non-seekable stream");
}
return stream->getLength();
}
private:
Common::DisposablePtr<Audio::RewindableAudioStream> _stream;
bool _loop;
};
bool detectSolAudio(Common::SeekableReadStream &stream) {
const size_t initialPosition = stream.pos();
byte header[6];
if (stream.read(header, sizeof(header)) != sizeof(header)) {
stream.seek(initialPosition);
return false;
}
stream.seek(initialPosition);
if ((header[0] & 0x7f) != kResourceTypeAudio || READ_BE_UINT32(header + 2) != MKTAG('S', 'O', 'L', 0)) {
return false;
}
return true;
}
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_2_1_EARLY) {
_channels.resize(10);
} else if (getSciVersion() < SCI_VERSION_3) {
_channels.resize(5);
} else {
_channels.resize(8);
}
_useModifiedAttenuation = g_sci->_features->usesModifiedAudioAttenuation();
// The mixer stream type is given as `kSFXSoundType` so that audio from
// Audio32 will be mixed at the same standard volume as the video players
// (which must use `kSFXSoundType` as well).
_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::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) {
const int samplePairsToRead = numSamples >> 1;
const int samplePairsWritten = converter->flow(*sourceStream, targetBuffer, samplePairsToRead, leftVolume, rightVolume);
return samplePairsWritten << 1;
}
int16 Audio32::getNumChannelsToMix() const {
Common::StackLock lock(_mutex);
int16 numChannels = 0;
for (int16 channelIndex = 0; channelIndex < _numActiveChannels; ++channelIndex) {
const AudioChannel &channel = getChannel(channelIndex);
if (channelShouldMix(channel)) {
++numChannels;
}
}
return numChannels;
}
bool Audio32::channelShouldMix(const AudioChannel &channel) const {
if (channel.pausedAtTick ||
(channel.robot && (_robotAudioPaused || channel.stream->endOfStream()))) {
return false;
}
if (channel.fadeStartTick) {
const uint32 fadeElapsed = g_sci->getTickCount() - channel.fadeStartTick;
if (fadeElapsed > channel.fadeDuration && channel.stopChannelOnFade) {
return false;
}
}
return true;
}
// 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 *const 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();
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.
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).
int8 attenuationAmount;
int8 attenuationStepAmount;
if (_useModifiedAttenuation) {
// Divides samples in target buffer by 4, and samples in source buffer
// by 0, when adding each channel to the output buffer.
// 1 channel: 0 >>0
// 2 channels: 0 >>2, 1 >>0
// 3 channels: 0 >>4, 1 >>2, 2 >>0
// 4 channels: 0 >>6, 1 >>4, 2 >>2, 3 >>0 ...
// Attenuation amounts are shift values.
attenuationAmount = (getNumChannelsToMix() - 1) * 2;
attenuationStepAmount = 2;
} else {
// Divides samples in both target & source buffers by 2 when adding each
// channel to the output buffer.
// 1 channel: 0 >>0
// 2 channels: 0 >>1, 1 >>1
// 3 channels: 0 >>2, 1 >>2, 2 >>1
// 4 channels: 0 >>3, 1 >>3, 2 >>2, 3 >>1 ...
// Attenuation amounts are shift values.
attenuationAmount = getNumChannelsToMix() - 1;
attenuationStepAmount = 1;
}
int maxSamplesWritten = 0;
bool firstChannelWritten = false;
for (int16 channelIndex = 0; channelIndex < _numActiveChannels; ++channelIndex) {
const AudioChannel &channel = getChannel(channelIndex);
if (channel.pausedAtTick || (channel.robot && _robotAudioPaused)) {
continue;
}
if (channel.robot && channel.stream->endOfStream()) {
stop(channelIndex--);
continue;
}
// Channel finished fading and had the
// stopChannelOnFade flag set, so no longer exists
if (channel.fadeStartTick && processFade(channelIndex)) {
--channelIndex;
continue;
}
Audio::st_volume_t leftVolume, rightVolume;
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.
if (volume > 0 && volume <= 42) {
volume = 15;
} else if (volume > 42 && volume <= 84) {
volume = 31;
} else if (volume > 84 && volume < kMaxVolume) {
volume = 63;
}
}
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;
}
if (!playOnlyMonitoredChannel && _attenuatedMixing) {
assert(attenuationAmount >= 0);
leftVolume >>= attenuationAmount;
rightVolume >>= attenuationAmount;
if (!_useModifiedAttenuation && !firstChannelWritten) {
firstChannelWritten = true;
} else {
attenuationAmount -= attenuationStepAmount;
}
}
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);
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()) {
if (playOnlyMonitoredChannel) {
// 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);
if (channelSamplesWritten > maxSamplesWritten) {
maxSamplesWritten = channelSamplesWritten;
}
}
}
_inAudioThread = false;
return maxSamplesWritten;
}
#pragma mark -
#pragma mark Channel management
uint8 Audio32::getNumUnlockedChannels() const {
Common::StackLock lock(_mutex);
uint8 numChannels = 0;
for (uint i = 0; i < _numActiveChannels; ++i) {
const AudioChannel &channel = getChannel(i);
if (!channel.robot && Common::find(_lockedResourceIds.begin(), _lockedResourceIds.end(), channel.id) == _lockedResourceIds.end()) {
++numChannels;
}
}
return numChannels;
}
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::lockResource(const ResourceId resourceId, const bool lock) {
Common::StackLock slock(_mutex);
LockList::iterator it = Common::find(_lockedResourceIds.begin(), _lockedResourceIds.end(), resourceId);
if (it != _lockedResourceIds.end()) {
if (!lock) {
_lockedResourceIds.erase(it);
}
} else {
if (lock) {
_lockedResourceIds.push_back(resourceId);
}
}
}
void Audio32::freeUnusedChannels() {
Common::StackLock lock(_mutex);
for (int16 channelIndex = 0; channelIndex < _numActiveChannels; ++channelIndex) {
const AudioChannel &channel = getChannel(channelIndex);
if (!channel.robot && channel.stream->endOfStream()) {
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);
// Robots have no corresponding resource to free
if (channel.robot) {
delete channel.stream;
channel.stream = nullptr;
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
if (_inAudioThread) {
_resourcesToUnlock.push_back(channel.resource);
} else {
_resMan->unlockResource(channel.resource);
}
channel.resource = nullptr;
delete channel.stream;
channel.stream = 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 Robot
int16 Audio32::findRobotChannel() const {
Common::StackLock lock(_mutex);
for (int16 i = 0; i < _numActiveChannels; ++i) {
if (_channels[i].robot) {
return i;
}
}
return kNoExistingChannel;
}
bool Audio32::playRobotAudio(const RobotAudioStream::RobotAudioPacket &packet) {
// Stop immediately
if (packet.dataSize == 0) {
warning("Stopping robot stream by zero-length packet");
return stopRobotAudio();
}
// Flush and then stop
if (packet.dataSize == -1) {
warning("Stopping robot stream by negative-length packet");
return finishRobotAudio();
}
Common::StackLock lock(_mutex);
int16 channelIndex = findRobotChannel();
bool isNewChannel = false;
if (channelIndex == kNoExistingChannel) {
if (_numActiveChannels == _channels.size()) {
return false;
}
channelIndex = _numActiveChannels++;
isNewChannel = true;
}
AudioChannel &channel = getChannel(channelIndex);
if (isNewChannel) {
channel.id = ResourceId();
channel.resource = nullptr;
channel.robot = true;
channel.fadeStartTick = 0;
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);
// The RobotAudioStream buffer size is
// ((bytesPerSample * channels * sampleRate * 2000ms) / 1000ms) & ~3
// where bytesPerSample = 2, channels = 1, and sampleRate = 22050
channel.stream = new RobotAudioStream(88200);
_robotAudioPaused = false;
if (_numActiveChannels == 1) {
_startedAtTick = g_sci->getTickCount();
}
}
return static_cast<RobotAudioStream *>(channel.stream)->addPacket(packet);
}
bool Audio32::queryRobotAudio(RobotAudioStream::StreamState &status) const {
Common::StackLock lock(_mutex);
const int16 channelIndex = findRobotChannel();
if (channelIndex == kNoExistingChannel) {
status.bytesPlaying = 0;
return false;
}
status = static_cast<RobotAudioStream *>(getChannel(channelIndex).stream)->getStatus();
return true;
}
bool Audio32::finishRobotAudio() {
Common::StackLock lock(_mutex);
const int16 channelIndex = findRobotChannel();
if (channelIndex == kNoExistingChannel) {
return false;
}
static_cast<RobotAudioStream *>(getChannel(channelIndex).stream)->finish();
return true;
}
bool Audio32::stopRobotAudio() {
Common::StackLock lock(_mutex);
const int16 channelIndex = findRobotChannel();
if (channelIndex == kNoExistingChannel) {
return false;
}
stop(channelIndex);
return true;
}
#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);
MutableLoopAudioStream *stream = dynamic_cast<MutableLoopAudioStream *>(channel.stream);
if (stream == nullptr) {
error("[Audio32::play]: Unable to cast stream for resource %s", resourceId.toString().c_str());
}
if (channel.pausedAtTick) {
resume(channelIndex);
return MIN(65534, 1 + stream->getLength().msecs() * 60 / 1000);
}
warning("Tried to resume channel %s that was not paused", channel.id.toString().c_str());
return MIN(65534, 1 + 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) {
warning("[Audio32::play]: %s could not be found", resourceId.toString().c_str());
return 0;
}
channelIndex = _numActiveChannels++;
AudioChannel &channel = getChannel(channelIndex);
channel.id = resourceId;
channel.resource = resource;
channel.robot = 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::SeekableReadStream *dataStream = resource->makeStream();
Audio::RewindableAudioStream *audioStream;
if (detectSolAudio(*dataStream)) {
audioStream = makeSOLStream(dataStream, DisposeAfterUse::YES);
} else if (detectWaveAudio(*dataStream)) {
audioStream = Audio::makeWAVStream(dataStream, DisposeAfterUse::YES);
} 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;
}
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);
// 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.
MutableLoopAudioStream *stream = dynamic_cast<MutableLoopAudioStream *>(channel.stream);
if (stream == nullptr) {
error("[Audio32::play]: Unable to cast stream for resource %s", resourceId.toString().c_str());
}
channel.duration = /* round up */ 1 + (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;
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;
}
uint16 Audio32::restart(const ResourceId resourceId, const bool autoPlay, const bool loop, const int16 volume, const reg_t soundNode, const bool monitor) {
Common::StackLock lock(_mutex);
stop(resourceId, soundNode);
return play(kNoExistingChannel, resourceId, autoPlay, loop, volume, soundNode, monitor);
}
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);
MutableLoopAudioStream *stream = dynamic_cast<MutableLoopAudioStream *>(channel.stream);
assert(stream);
stream->loop() = loop;
}
#pragma mark -
#pragma mark Effects
int16 Audio32::getVolume(const int16 channelIndex) const {
if (channelIndex < 0 || channelIndex >= _numActiveChannels) {
return (_mixer->getVolumeForSoundType(Audio::Mixer::kSFXSoundType) + 1) * kMaxVolume / Audio::Mixer::kMaxMixerVolume;
}
Common::StackLock lock(_mutex);
return getChannel(channelIndex).volume;
}
void Audio32::setVolume(const int16 channelIndex, int16 volume) {
volume = MIN<int16>(kMaxVolume, volume);
if (channelIndex == kAllChannels) {
if (!g_sci->_guestAdditions->audio32SetVolumeHook(channelIndex, volume)) {
setMasterVolume(volume);
}
} 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;
}
#pragma mark -
#pragma mark Kernel
reg_t Audio32::kernelPlay(const bool autoPlay, const int argc, const reg_t *const argv) {
Common::StackLock lock(_mutex);
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 (getSciVersion() == SCI_VERSION_3) {
if (argc < 7) {
volume = Audio32::kMaxVolume;
} else {
volume = argv[6].toSint16() & Audio32::kMaxVolume;
monitor = argv[6].toSint16() & Audio32::kMonitorAudioFlagSci3;
}
} else {
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();
}
if (getSciVersion() == SCI_VERSION_3) {
if (argc < 3) {
volume = Audio32::kMaxVolume;
} else {
volume = argv[2].toSint16() & Audio32::kMaxVolume;
monitor = argv[2].toSint16() & Audio32::kMonitorAudioFlagSci3;
}
} else {
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));
}
reg_t Audio32::kernelStop(const int argc, const reg_t *const argv) {
Common::StackLock lock(_mutex);
const int16 channelIndex = findChannelByArgs(argc, argv, 0, argc > 1 ? argv[1] : NULL_REG);
return make_reg(0, stop(channelIndex));
}
reg_t Audio32::kernelPause(const int argc, const reg_t *const argv) {
Common::StackLock lock(_mutex);
const int16 channelIndex = findChannelByArgs(argc, argv, 0, argc > 1 ? argv[1] : NULL_REG);
return make_reg(0, pause(channelIndex));
}
reg_t Audio32::kernelResume(const int argc, const reg_t *const argv) {
Common::StackLock lock(_mutex);
const int16 channelIndex = findChannelByArgs(argc, argv, 0, argc > 1 ? argv[1] : NULL_REG);
return make_reg(0, resume(channelIndex));
}
reg_t Audio32::kernelPosition(const int argc, const reg_t *const argv) {
Common::StackLock lock(_mutex);
const int16 channelIndex = findChannelByArgs(argc, argv, 0, argc > 1 ? argv[1] : NULL_REG);
return make_reg(0, getPosition(channelIndex));
}
reg_t Audio32::kernelVolume(const int argc, const reg_t *const argv) {
Common::StackLock lock(_mutex);
const int16 volume = argc > 0 ? argv[0].toSint16() : -1;
const int16 channelIndex = findChannelByArgs(argc, argv, 1, argc > 2 ? argv[2] : NULL_REG);
if (volume != -1) {
setVolume(channelIndex, volume);
}
return make_reg(0, getVolume(channelIndex));
}
reg_t Audio32::kernelMixing(const int argc, const reg_t *const argv) {
Common::StackLock lock(_mutex);
if (argc > 0) {
setAttenuatedMixing(argv[0].toUint16());
}
return make_reg(0, getAttenuatedMixing());
}
reg_t Audio32::kernelFade(const int argc, const reg_t *const argv) {
if (argc < 4) {
return make_reg(0, 0);
}
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.
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();
const int16 steps = argv[3].toSint16();
const bool stopAfterFade = argc > 4 ? (bool)argv[4].toUint16() : false;
return make_reg(0, fadeChannel(channelIndex, volume, speed, steps, stopAfterFade));
}
void Audio32::kernelLoop(const int argc, const reg_t *const argv) {
Common::StackLock lock(_mutex);
const int16 channelIndex = findChannelByArgs(argc, argv, 0, argc == 3 ? argv[2] : NULL_REG);
const bool loop = argv[0].toSint16() != 0 && argv[0].toSint16() != 1;
setLoop(channelIndex, loop);
}
#pragma mark -
#pragma mark Debugging
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);
con->debugPrintf(" %d[%04x:%04x]: %s, started at %d, pos %d/%d, vol %d, pan %d%s%s\n",
i,
PRINT_REG(channel.soundNode),
channel.robot ? "robot" : channel.resource->name().c_str(),
channel.startedAtTick,
(g_sci->getTickCount() - channel.startedAtTick) % channel.duration,
channel.duration,
channel.volume,
channel.pan,
stream && stream->loop() ? ", looping" : "",
channel.pausedAtTick ? ", paused" : "");
if (channel.fadeStartTick) {
con->debugPrintf(" fade: vol %d -> %d, started at %d, pos %d/%d%s\n",
channel.fadeStartVolume,
channel.fadeTargetVolume,
channel.fadeStartTick,
(g_sci->getTickCount() - channel.fadeStartTick) % channel.duration,
channel.fadeDuration,
channel.stopChannelOnFade ? ", stopping" : "");
}
}
if (g_sci->_features->hasSci3Audio()) {
con->debugPrintf("\nLocks: ");
if (_lockedResourceIds.size()) {
const char *separator = "";
for (LockList::const_iterator it = _lockedResourceIds.begin(); it != _lockedResourceIds.end(); ++it) {
con->debugPrintf("%s%s", separator, it->toString().c_str());
separator = ", ";
}
} else {
con->debugPrintf("none");
}
con->debugPrintf("\n");
}
}
} // End of namespace Sci
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