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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/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::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 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()) {
Audio::RewindableAudioStream *rewindableStream = dynamic_cast<Audio::RewindableAudioStream *>(sourceStream);
if (rewindableStream == nullptr) {
error("[Audio32::writeAudioInternal]: Unable to cast stream");
}
rewindableStream->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) {
if (channel.stream->endOfStream()) {
stop(channelIndex--);
} else {
const int channelSamplesWritten = writeAudioInternal(channel.stream, channel.converter, buffer, numSamples, kMaxVolume, kMaxVolume, channel.loop);
if (channelSamplesWritten > maxSamplesWritten) {
maxSamplesWritten = channelSamplesWritten;
}
}
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.robot && channel.stream->endOfStream()) {
if (channel.loop) {
Audio::SeekableAudioStream *stream = dynamic_cast<Audio::SeekableAudioStream *>(channel.stream);
if (stream == nullptr) {
error("[Audio32::freeUnusedChannels]: Unable to cast stream for resource %s", channel.id.toString().c_str());
}
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);
// 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.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 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.loop = false;
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);
Audio::SeekableAudioStream *stream = dynamic_cast<Audio::SeekableAudioStream *>(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) {
return 0;
}
channelIndex = _numActiveChannels++;
AudioChannel &channel = getChannel(channelIndex);
channel.id = resourceId;
channel.resource = resource;
channel.loop = loop;
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::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.
Audio::SeekableAudioStream *stream = dynamic_cast<Audio::SeekableAudioStream *>(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;
}
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
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