SoundDevice.cc

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#include "SoundDevice.hh"
#include "MSXMixer.hh"
#include "DeviceConfig.hh"
#include "XMLElement.hh"
#include "WavWriter.hh"
#include "Filename.hh"
#include "StringOp.hh"
#include "MemoryOps.hh"
#include "MemBuffer.hh"
#include "MSXException.hh"
#include "likely.hh"
#include "vla.hh"
#include "memory.hh"
#include <cassert>

using std::string;

namespace openmsx {

static MemBuffer<int, SSE2_ALIGNMENT> mixBuffer;
static unsigned mixBufferSize = 0;

static void allocateMixBuffer(unsigned size)
{
        if (unlikely(mixBufferSize < size)) {
                mixBufferSize = size;
                mixBuffer.resize(mixBufferSize);
        }
}

static string makeUnique(MSXMixer& mixer, string_view name)
{
        string result = name.str();
        if (mixer.findDevice(result)) {
                unsigned n = 0;
                do {
                        result = strCat(name, " (", ++n, ')');
                } while (mixer.findDevice(result));
        }
        return result;
}

void SoundDevice::addFill(int*& buf, int val, unsigned num)
{
        // Note: in the past we tried to optimize this by always producing
        // a multiple of 4 output values. In the general case a sounddevice is
        // allowed to do this, but only at the end of the soundbuffer. This
        // method can also be called in the middle of a buffer (so multiple
        // times per buffer), in such case it does go wrong.
        assert(num > 0);
#ifdef __arm__
        asm volatile (
                "subs   %[num],%[num],#4\n\t"
                "bmi    1f\n"
        "0:\n\t"
                "ldmia  %[buf],{r3-r6}\n\t"
                "add    r3,r3,%[val]\n\t"
                "add    r4,r4,%[val]\n\t"
                "add    r5,r5,%[val]\n\t"
                "add    r6,r6,%[val]\n\t"
                "stmia  %[buf]!,{r3-r6}\n\t"
                "subs   %[num],%[num],#4\n\t"
                "bpl    0b\n"
        "1:\n\t"
                "tst    %[num],#2\n\t"
                "beq    2f\n\t"
                "ldmia  %[buf],{r3-r4}\n\t"
                "add    r3,r3,%[val]\n\t"
                "add    r4,r4,%[val]\n\t"
                "stmia  %[buf]!,{r3-r4}\n"
        "2:\n\t"
                "tst    %[num],#1\n\t"
                "beq    3f\n\t"
                "ldr    r3,[%[buf]]\n\t"
                "add    r3,r3,%[val]\n\t"
                "str    r3,[%[buf]],#4\n"
        "3:\n\t"
                : [buf] "=r"    (buf)
                , [num] "=r"    (num)
                :       "[buf]" (buf)
                , [val] "r"     (val)
                ,       "[num]" (num)
                : "memory", "r3","r4","r5","r6"
        );
        return;
#endif
        do {
                *buf++ += val;
        } while (--num);
}

SoundDevice::SoundDevice(MSXMixer& mixer_, string_view name_,
                         string_view description_,
                         unsigned numChannels_, bool stereo_)
        : mixer(mixer_)
        , name(makeUnique(mixer, name_))
        , description(description_.str())
        , numChannels(numChannels_)
        , stereo(stereo_ ? 2 : 1)
        , numRecordChannels(0)
        , balanceCenter(true)
{
        assert(numChannels <= MAX_CHANNELS);
        assert(stereo == 1 || stereo == 2);

        // initially no channels are muted
        for (unsigned i = 0; i < numChannels; ++i) {
                channelMuted[i] = false;
                channelBalance[i] = 0;
        }
}

SoundDevice::~SoundDevice() = default;

bool SoundDevice::isStereo() const
{
        return stereo == 2 || !balanceCenter;
}

int SoundDevice::getAmplificationFactorImpl() const
{
        return 1;
}

void SoundDevice::registerSound(const DeviceConfig& config)
{
        const XMLElement& soundConfig = config.getChild("sound");
        float volume = soundConfig.getChildDataAsInt("volume") / 32767.0f;
        int devBalance = 0;
        string_view mode = soundConfig.getChildData("mode", "mono");
        if (mode == "mono") {
                devBalance = 0;
        } else if (mode == "left") {
                devBalance = -100;
        } else if (mode == "right") {
                devBalance = 100;
        } else {
                throw MSXException("balance \"", mode, "\" illegal");
        }

        for (auto& b : soundConfig.getChildren("balance")) {
                int balance = StringOp::stringToInt(b->getData());

                if (!b->hasAttribute("channel")) {
                        devBalance = balance;
                        continue;
                }

                // TODO Support other balances
                if (balance != 0 && balance != -100 && balance != 100) {
                        throw MSXException("balance ", balance, " illegal");
                }
                if (balance != 0) {
                        balanceCenter = false;
                }

                const string& range = b->getAttribute("channel");
                for (unsigned c : StringOp::parseRange(range, 1, numChannels)) {
                        channelBalance[c - 1] = balance;
                }
        }

        mixer.registerSound(*this, volume, devBalance, numChannels);
}

void SoundDevice::unregisterSound()
{
        mixer.unregisterSound(*this);
}

void SoundDevice::updateStream(EmuTime::param time)
{
        mixer.updateStream(time);
}

void SoundDevice::setSoftwareVolume(VolumeType volume, EmuTime::param time)
{
        setSoftwareVolume(volume, volume, time);
}

void SoundDevice::setSoftwareVolume(VolumeType left, VolumeType right, EmuTime::param time)
{
        updateStream(time);
        softwareVolumeLeft  = left;
        softwareVolumeRight = right;
        mixer.updateSoftwareVolume(*this);
}

void SoundDevice::recordChannel(unsigned channel, const Filename& filename)
{
        assert(channel < numChannels);
        bool wasRecording = writer[channel] != nullptr;
        if (!filename.empty()) {
                writer[channel] = make_unique<Wav16Writer>(
                        filename, stereo, inputSampleRate);
        } else {
                writer[channel].reset();
        }
        bool recording = writer[channel] != nullptr;
        if (recording != wasRecording) {
                if (recording) {
                        if (numRecordChannels == 0) {
                                mixer.setSynchronousMode(true);
                        }
                        ++numRecordChannels;
                        assert(numRecordChannels <= numChannels);
                } else {
                        assert(numRecordChannels > 0);
                        --numRecordChannels;
                        if (numRecordChannels == 0) {
                                mixer.setSynchronousMode(false);
                        }
                }
        }
}

void SoundDevice::muteChannel(unsigned channel, bool muted)
{
        assert(channel < numChannels);
        channelMuted[channel] = muted;
}

bool SoundDevice::mixChannels(int* dataOut, unsigned samples)
{
#ifdef __SSE2__
        assert((uintptr_t(dataOut) & 15) == 0); // must be 16-byte aligned
#endif
        if (samples == 0) return true;
        unsigned outputStereo = isStereo() ? 2 : 1;

        MemoryOps::MemSet<unsigned> mset;
        if (numChannels != 1) {
                // The generateChannels() method of SoundDevices with more than
                // one channel will _add_ the generated channel data in the
                // provided buffers. Those with only one channel will directly
                // replace the content of the buffer. For the former we must
                // start from a buffer containing all zeros.
                mset(reinterpret_cast<unsigned*>(dataOut), outputStereo * samples, 0);
        }

        VLA(int*, bufs, numChannels);
        unsigned separateChannels = 0;
        unsigned pitch = (samples * stereo + 3) & ~3; // align for SSE access
        // TODO optimization: All channels with the same balance (according to
        // channelBalance[]) could use the same buffer when balanceCenter is
        // false
        for (unsigned i = 0; i < numChannels; ++i) {
                if (!channelMuted[i] && !writer[i] && balanceCenter) {
                        // no need to keep this channel separate
                        bufs[i] = dataOut;
                } else {
                        // muted or recorded channels must go separate
                        //  cannot yet fill in bufs[i] here
                        ++separateChannels;
                }
        }
        if (separateChannels) {
                allocateMixBuffer(pitch * separateChannels);
                mset(reinterpret_cast<unsigned*>(mixBuffer.data()),
                     pitch * separateChannels, 0);
                // still need to fill in (some) bufs[i] pointers
                unsigned count = 0;
                for (unsigned i = 0; i < numChannels; ++i) {
                        if (!(!channelMuted[i] && !writer[i] && balanceCenter)) {
                                bufs[i] = &mixBuffer[pitch * count++];
                        }
                }
                assert(count == separateChannels);
        }

        generateChannels(bufs, samples);

        if (separateChannels == 0) {
                for (unsigned i = 0; i < numChannels; ++i) {
                        if (bufs[i]) {
                                return true;
                        }
                }
                return false;
        }

        // record channels
        for (unsigned i = 0; i < numChannels; ++i) {
                if (writer[i]) {
                        assert(bufs[i] != dataOut);
                        if (bufs[i]) {
                                auto amp = getAmplificationFactor();
                                writer[i]->write(
                                        bufs[i], stereo, samples,
                                        amp.first.toFloat(),
                                        amp.second.toFloat());
                        } else {
                                writer[i]->writeSilence(stereo, samples);
                        }
                }
        }

        // remove muted channels (explictly by user or by device itself)
        bool anyUnmuted = false;
        unsigned numMix = 0;
        VLA(int, mixBalance, numChannels);
        for (unsigned i = 0; i < numChannels; ++i) {
                if (bufs[i] && !channelMuted[i]) {
                        anyUnmuted = true;
                        if (bufs[i] != dataOut) {
                                bufs[numMix] = bufs[i];
                                mixBalance[numMix] = channelBalance[i];
                                ++numMix;
                        }
                }
        }

        if (numMix == 0) {
                // all extra channels muted
                return anyUnmuted;
        }

        // actually mix channels
        if (!balanceCenter) {
                unsigned i = 0;
                do {
                        int left0  = 0;
                        int right0 = 0;
                        int left1  = 0;
                        int right1 = 0;
                        unsigned j = 0;
                        do {
                                if (mixBalance[j] <= 0) {
                                        left0  += bufs[j][i + 0];
                                        left1  += bufs[j][i + 1];
                                }
                                if (mixBalance[j] >= 0) {
                                        right0 += bufs[j][i + 0];
                                        right1 += bufs[j][i + 1];
                                }
                                j++;
                        } while (j < numMix);
                        dataOut[i * 2 + 0] = left0;
                        dataOut[i * 2 + 1] = right0;
                        dataOut[i * 2 + 2] = left1;
                        dataOut[i * 2 + 3] = right1;
                        i += 2;
                } while (i < samples);

                return true;
        }

        // In the past we had ARM and x86-SSE2 optimized assembly routines for
        // the stuff below. Currently this code is only rarely used anymore
        // (only when recording or muting individual soundchip channels), so
        // it's not worth the extra complexity anymore.
        unsigned num = samples * stereo;
        unsigned i = 0;
        do {
                int out0 = dataOut[i + 0];
                int out1 = dataOut[i + 1];
                int out2 = dataOut[i + 2];
                int out3 = dataOut[i + 3];
                unsigned j = 0;
                do {
                        out0 += bufs[j][i + 0];
                        out1 += bufs[j][i + 1];
                        out2 += bufs[j][i + 2];
                        out3 += bufs[j][i + 3];
                        ++j;
                } while (j < numMix);
                dataOut[i + 0] = out0;
                dataOut[i + 1] = out1;
                dataOut[i + 2] = out2;
                dataOut[i + 3] = out3;
                i += 4;
        } while (i < num);

        return true;
}

const DynamicClock& SoundDevice::getHostSampleClock() const
{
        return mixer.getHostSampleClock();
}
double SoundDevice::getEffectiveSpeed() const
{
        return mixer.getEffectiveSpeed();
}

} // namespace openmsx