From 90e3c6f54fddf8762f7349414211a8c0f94bf594 Mon Sep 17 00:00:00 2001
From: Wohlstand <admin@wohlnet.ru>
Date: Sat, 10 Oct 2015 01:51:33 +0300
Subject: Initial commit

---
 src/midiplay/adlmidiplay.cpp | 202 +++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 202 insertions(+)
 create mode 100644 src/midiplay/adlmidiplay.cpp

(limited to 'src/midiplay/adlmidiplay.cpp')

diff --git a/src/midiplay/adlmidiplay.cpp b/src/midiplay/adlmidiplay.cpp
new file mode 100644
index 0000000..3d58d37
--- /dev/null
+++ b/src/midiplay/adlmidiplay.cpp
@@ -0,0 +1,202 @@
+
+#include <vector>
+#include <string>
+#include <cstdio>
+#include <cstdlib>
+#include <cstring>
+#include <deque>
+#include <SDL2/SDL.h>
+
+#include <adlmidi.h>
+
+class MutexType
+{
+    SDL_mutex* mut;
+public:
+    MutexType() : mut(SDL_CreateMutex()) { }
+    ~MutexType() { SDL_DestroyMutex(mut); }
+    void Lock() { SDL_mutexP(mut); }
+    void Unlock() { SDL_mutexV(mut); }
+};
+
+
+static std::deque<short> AudioBuffer;
+static MutexType AudioBuffer_lock;
+
+static void SDL_AudioCallbackX(void*, Uint8* stream, int len)
+{
+    SDL_LockAudio();
+    short* target = (short*) stream;
+    AudioBuffer_lock.Lock();
+    /*if(len != AudioBuffer.size())
+        fprintf(stderr, "len=%d stereo samples, AudioBuffer has %u stereo samples",
+            len/4, (unsigned) AudioBuffer.size()/2);*/
+    unsigned ate = len/2; // number of shorts
+    if(ate > AudioBuffer.size()) ate = AudioBuffer.size();
+    for(unsigned a=0; a<ate; ++a)
+    {
+        target[a] = AudioBuffer[a];
+    }
+    AudioBuffer.erase(AudioBuffer.begin(), AudioBuffer.begin() + ate);
+    AudioBuffer_lock.Unlock();
+    SDL_UnlockAudio();
+}
+
+
+
+#undef main
+int main(int argc, char** argv)
+{
+
+    if(argc < 2 || std::string(argv[1]) == "--help" || std::string(argv[1]) == "-h")
+    {
+        std::printf(
+            "Usage: adlmidi <midifilename> [ <options> ] [ <banknumber> [ <numcards> [ <numfourops>] ] ]\n"
+            " -p Enables adlib percussion instrument mode\n"
+            " -t Enables tremolo amplification mode\n"
+            " -v Enables vibrato amplification mode\n"
+            " -s Enables scaling of modulator volumes\n"
+            " -nl Quit without looping\n"
+            " -w Write WAV file rather than playing\n"
+        );
+/*
+        for(unsigned a=0; a<sizeof(banknames)/sizeof(*banknames); ++a)
+            std::printf("%10s%2u = %s\n",
+                a?"":"Banks:",
+                a,
+                banknames[a]);
+*/
+        std::printf(
+            "     Use banks 2-5 to play Descent \"q\" soundtracks.\n"
+            "     Look up the relevant bank number from descent.sng.\n"
+            "\n"
+            "     The fourth parameter can be used to specify the number\n"
+            "     of four-op channels to use. Each four-op channel eats\n"
+            "     the room of two regular channels. Use as many as required.\n"
+            "     The Doom & Hexen sets require one or two, while\n"
+            "     Miles four-op set requires the maximum of numcards*6.\n"
+            "\n"
+            );
+        return 0;
+    }
+
+    //const unsigned MaxSamplesAtTime = 512; // 512=dbopl limitation
+    // How long is SDL buffer, in seconds?
+    // The smaller the value, the more often SDL_AudioCallBack()
+    // is called.
+    const double AudioBufferLength = 0.08;
+    // How much do WE buffer, in seconds? The smaller the value,
+    // the more prone to sound chopping we are.
+    const double OurHeadRoomLength = 0.1;
+    // The lag between visual content and audio content equals
+    // the sum of these two buffers.
+    SDL_AudioSpec spec;
+    SDL_AudioSpec obtained;
+
+    spec.freq     = 44100;
+    spec.format   = AUDIO_S16SYS;
+    spec.channels = 2;
+    spec.samples  = spec.freq * AudioBufferLength;
+    spec.callback = SDL_AudioCallbackX;
+
+    // Set up SDL
+    if(SDL_OpenAudio(&spec, &obtained) < 0)
+    {
+        std::fprintf(stderr, "Couldn't open audio: %s\n", SDL_GetError());
+        //return 1;
+    }
+    if(spec.samples != obtained.samples)
+        std::fprintf(stderr, "Wanted (samples=%u,rate=%u,channels=%u); obtained (samples=%u,rate=%u,channels=%u)\n",
+            spec.samples,    spec.freq,    spec.channels,
+            obtained.samples,obtained.freq,obtained.channels);
+
+    ADL_MIDIPlayer* myDevice;
+    myDevice = adl_init(44100);
+    if(myDevice==NULL)
+    {
+        std::fprintf(stderr, "Failed to init MIDI device!\n");
+        return 1;
+    }
+
+
+    while(argc > 2)
+    {
+        bool had_option = false;
+
+        if(!std::strcmp("-p", argv[2]))
+            adl_setPercMode(myDevice, 1);
+        else if(!std::strcmp("-v", argv[2]))
+            adl_setHVibrato(myDevice, 1);
+        else if(!std::strcmp("-t", argv[2]))
+            adl_setHTremolo(myDevice, 1);
+        else if(!std::strcmp("-nl", argv[2]))
+            adl_setLoopEnabled(myDevice, 0);
+        else if(!std::strcmp("-s", argv[2]))
+            adl_setScaleModulators(myDevice, 1);
+        else break;
+
+        std::copy(argv + (had_option ? 4 : 3), argv + argc,
+                  argv+2);
+        argc -= (had_option ? 2 : 1);
+    }
+
+    if(argc >= 3)
+    {
+        int bankno = std::atoi(argv[2]);
+        if(adl_setBank(myDevice, bankno)!=0)
+        {
+            std::fprintf(stderr,"%s", adl_errorString());
+            return 0;
+        }
+    }
+
+    if(argc >= 4)
+    {
+        if(adl_setNumCards(myDevice, std::atoi(argv[3])) != 0)
+        {
+            std::fprintf(stderr, "%s\n", adl_errorString());
+            return 0;
+        }
+    }
+    if(argc >= 5)
+    {
+        if(adl_setNumFourOpsChn(myDevice, std::atoi(argv[4]))!=0)
+        {
+            std::fprintf(stderr, "%s\n", adl_errorString());
+            return 0;
+        }
+    }
+
+    if(adl_openFile(myDevice, argv[1])!=0)
+    {
+        std::fprintf(stderr, "%s\n", adl_errorString());
+        return 2;
+    }
+
+    SDL_PauseAudio(0);
+
+    while(1)
+    {
+        int buff[4096];
+        unsigned long gotten=adl_play(myDevice, 4096, buff);
+        if(gotten<=0) break;
+
+        AudioBuffer_lock.Lock();
+            size_t pos = AudioBuffer.size();
+            AudioBuffer.resize(pos + gotten);
+            for(unsigned long p = 0; p < gotten; ++p)
+               AudioBuffer[pos+p] = buff[p];
+        AudioBuffer_lock.Unlock();
+
+        const SDL_AudioSpec& spec_ = obtained;
+        while(AudioBuffer.size() > spec_.samples + (spec_.freq*2) * OurHeadRoomLength)
+        {
+            SDL_Delay(1);
+        }
+    }
+
+    adl_close(myDevice);
+    SDL_CloseAudio();
+    return 0;
+}
+
-- 
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