audio changes

assets/disk0/home/playrawpcm.js

@@ -4,7 +4,7 @@ music.pread(samples, 65534)
 
 audio.setPcmMode(0)
 audio.setMasterVolume(0, 255)
-audio.putPcmDataByPtr(samples, 65534, 0)
+audio.putPcmDataByPtr(0, samples, 65534, 0)
 audio.setLoopPoint(0, 65534)
 audio.play(0)*/
 
@@ -127,7 +127,7 @@ while (sampleSize > 0) {
             let readLength = (sampleSize < BLOCK_SIZE) ? sampleSize : BLOCK_SIZE
             readBytes(readLength, decodePtr)
 
-            audio.putPcmDataByPtr(decodePtr, readLength, 0)
+            audio.putPcmDataByPtr(0, decodePtr, readLength, 0)
             audio.setSampleUploadLength(0, readLength)
             audio.startSampleUpload(0)
 

assets/disk0/tvdos/bin/hopper.js

@@ -0,0 +1,5 @@
+/**
+ * Hopper is a package manager for TSVM
+ * Created by CuriousTorvald on 2026-04-16
+ */
+

assets/disk0/tvdos/bin/playmv1.js

@@ -326,7 +326,7 @@ while (!stopPlay && seqread.getReadCount() < FILE_LENGTH) {
                     // RAW PCM packets (decode on the fly)
                     else if (packetType == 0x1000 || packetType == 0x1001) {
                         let frame = seqread.readBytes(readLength)
-                        audio.putPcmDataByPtr(frame, readLength, 0)
+                        audio.putPcmDataByPtr(0, frame, readLength, 0)
                         audio.setSampleUploadLength(0, readLength)
                         audio.startSampleUpload(0)
                         sys.free(frame)

assets/disk0/tvdos/bin/playpcm.js

@@ -162,7 +162,7 @@ while (!stopPlay && seqread.getReadCount() < FILE_SIZE && readLength > 0) {
 
             seqread.readBytes(readLength, readPtr)
 
-            audio.putPcmDataByPtr(readPtr, readLength, 0)
+            audio.putPcmDataByPtr(0, readPtr, readLength, 0)
             audio.setSampleUploadLength(0, readLength)
             audio.startSampleUpload(0)
 

assets/disk0/tvdos/bin/playtav.js

@@ -18,6 +18,7 @@ const ADDRESSING_INTERNAL = 0x02
 const SND_BASE_ADDR = audio.getBaseAddr()
 const SND_MEM_ADDR = audio.getMemAddr()
 const pcm = require("pcm")
+const AUDIO_DEVICE = 3
 const MP2_FRAME_SIZE = [144,216,252,288,360,432,504,576,720,864,1008,1152,1440,1728]
 const TAV_TEMPORAL_LEVELS = 2
 
@@ -152,10 +153,10 @@ graphics.clearPixels4(0)
 const gpuGraphicsMode = graphics.getGraphicsMode()
 
 // Initialize audio
-audio.resetParams(0)
-audio.purgeQueue(0)
-audio.setPcmMode(0)
-audio.setMasterVolume(0, 255)
+audio.resetParams(AUDIO_DEVICE)
+audio.purgeQueue(AUDIO_DEVICE)
+audio.setPcmMode(AUDIO_DEVICE)
+audio.setMasterVolume(AUDIO_DEVICE, 255)
 
 // set colour zero as half-opaque black
 graphics.setPalette(0, 0, 0, 0, 7)
@@ -1152,10 +1153,10 @@ try {
                 else if (keyCode == 62) { // SPACE - pause/resume
                     paused = !paused
                     if (paused) {
-                        audio.stop(0)
+                        audio.stop(AUDIO_DEVICE)
                         serial.println(`Paused at frame ${frameCount}`)
                     } else {
-                        audio.play(0)
+                        audio.play(AUDIO_DEVICE)
                         serial.println(`Resumed`)
                     }
                 }
@@ -1176,10 +1177,10 @@ try {
                         baseTimecodeFrameCount = 0
                         currentTimecodeNs = 0
                         nextSubtitleEventIndex = 0  // Reset subtitle event processing
-                        audio.purgeQueue(0)
+                        audio.purgeQueue(AUDIO_DEVICE)
                         if (paused) {
-                            audio.play(0)
-                            audio.stop(0)
+                            audio.play(AUDIO_DEVICE)
+                            audio.stop(AUDIO_DEVICE)
                         }
                         skipped = true
                     }
@@ -1201,10 +1202,10 @@ try {
                         baseTimecodeFrameCount = 0
                         currentTimecodeNs = 0
                         nextSubtitleEventIndex = 0  // Reset subtitle event processing
-                        audio.purgeQueue(0)
+                        audio.purgeQueue(AUDIO_DEVICE)
                         if (paused) {
-                            audio.play(0)
-                            audio.stop(0)
+                            audio.play(AUDIO_DEVICE)
+                            audio.stop(AUDIO_DEVICE)
                         }
                         skipped = true
                     }
@@ -1232,10 +1233,10 @@ try {
                                 break
                             }
                         }
-                        audio.purgeQueue(0)
+                        audio.purgeQueue(AUDIO_DEVICE)
                         if (paused) {
-                            audio.play(0)
-                            audio.stop(0)
+                            audio.play(AUDIO_DEVICE)
+                            audio.stop(AUDIO_DEVICE)
                         }
                         skipped = true
                     }
@@ -1271,10 +1272,10 @@ try {
                                 break
                             }
                         }
-                        audio.purgeQueue(0)
+                        audio.purgeQueue(AUDIO_DEVICE)
                         if (paused) {
-                            audio.play(0)
-                            audio.stop(0)
+                            audio.play(AUDIO_DEVICE)
+                            audio.stop(AUDIO_DEVICE)
                         }
                         skipped = true
                     } else if (!seekTarget) {
@@ -1313,7 +1314,7 @@ try {
                     baseTimecodeFrameCount = 0
                     currentTimecodeNs = 0
                     nextSubtitleEventIndex = 0  // Reset subtitle event processing
-                    audio.purgeQueue(0)
+                    audio.purgeQueue(AUDIO_DEVICE)
                     currentFileIndex++
                     if (skipped) {
                         skipped = false
@@ -1737,7 +1738,7 @@ try {
 
                 seqread.readBytes(audioLen, SND_BASE_ADDR - 2368)
                 audio.mp2Decode()
-                audio.mp2UploadDecoded(0)
+                audio.mp2UploadDecoded(AUDIO_DEVICE)
 
             }
             else if (packetType === TAV_PACKET_AUDIO_TAD) {
@@ -1750,7 +1751,7 @@ try {
 
                 seqread.readBytes(payloadLen, SND_MEM_ADDR - 262144)
                 audio.tadDecode()
-                audio.tadUploadDecoded(0, sampleLen)
+                audio.tadUploadDecoded(AUDIO_DEVICE, sampleLen)
             }
             else if (packetType === TAV_PACKET_AUDIO_NATIVE) {
                 // PCM length must not exceed 65536 bytes!
@@ -1762,10 +1763,10 @@ try {
                 let pcmLen = gzip.decompFromTo(zstdPtr, zstdLen, pcmPtr) // <- segfaults!
                 if (pcmLen > 65536) throw Error(`PCM data too long -- got ${pcmLen} bytes`)
 
-                audio.putPcmDataByPtr(pcmPtr, pcmLen, 0)
+                audio.putPcmDataByPtr(AUDIO_DEVICE, pcmPtr, pcmLen, 0)
 
-                audio.setSampleUploadLength(0, pcmLen)
-                audio.startSampleUpload(0)
+                audio.setSampleUploadLength(AUDIO_DEVICE, pcmLen)
+                audio.startSampleUpload(AUDIO_DEVICE)
                 sys.free(zstdPtr)
 
                 sys.free(pcmPtr)
@@ -2049,7 +2050,7 @@ try {
 
                 // Fire audio on first frame
                 if (!audioFired) {
-                    audio.play(0)
+                    audio.play(AUDIO_DEVICE)
                     audioFired = true
                 }
 
@@ -2137,7 +2138,7 @@ try {
 
                 // Fire audio on first frame
                 if (!audioFired) {
-                    audio.play(0)
+                    audio.play(AUDIO_DEVICE)
                     audioFired = true
                 }
 
@@ -2173,8 +2174,8 @@ try {
                     sys.memcpy(predecodedPcmBuffer + predecodedPcmOffset, SND_BASE_ADDR, uploadSize)
 
                     // Set upload parameters and trigger upload to queue
-                    audio.setSampleUploadLength(0, uploadSize)
-                    audio.startSampleUpload(0)
+                    audio.setSampleUploadLength(AUDIO_DEVICE, uploadSize)
+                    audio.startSampleUpload(AUDIO_DEVICE)
 
                     predecodedPcmOffset += uploadSize
                 }
@@ -2458,8 +2459,8 @@ finally {
     sys.poke(-1299460, 20)
     sys.poke(-1299460, 21)
 
-    audio.stop(0)
-    audio.purgeQueue(0)
+    audio.stop(AUDIO_DEVICE)
+    audio.purgeQueue(AUDIO_DEVICE)
 }
 
 graphics.setPalette(0, 0, 0, 0, 0)

assets/disk0/tvdos/bin/playwav.js

@@ -289,7 +289,7 @@ while (!stopPlay && seqread.getReadCount() < FILE_SIZE - 8) {
                     let decodedSampleLength = decodeInfilePcm(readPtr, decodePtr, readLength)
                     printdbg(`        decodedSampleLength: ${decodedSampleLength}`)
 
-                    audio.putPcmDataByPtr(decodePtr, decodedSampleLength, 0)
+                    audio.putPcmDataByPtr(0, decodePtr, decodedSampleLength, 0)
                     audio.setSampleUploadLength(0, decodedSampleLength)
                     audio.startSampleUpload(0)
 

assets/disk0/tvdos/include/psg.mjs

@@ -119,7 +119,7 @@ function mixInto(buf, lengthSec, offsetSec, op, amp, pan, sampleFn) {
 
 // ── Waveform generators ─────────────────────────────────────────────────────
 
-function makeSquare(buf, length, offset, freq, duty, op, amp, pan) {
+function makeSquare(buf, length, offset, freq, duty, op, amp, pan, phaseOffset) {
     // buffer: [Uint8Array, Uint8Array] or native buffer
     // length: in seconds
     // offset: in seconds
@@ -128,13 +128,16 @@ function makeSquare(buf, length, offset, freq, duty, op, amp, pan) {
     // op: add / mul / sub; default: add
     // amp: 0.0 to 1.0; default: 0.5
     // pan: -1.0 to 1.0; default: 0.0
+    // phaseOffset: optional absolute-time base (seconds) added to phase calc only,
+    //              not to the buffer write position — use to ensure phase continuity
+    //              across successive calls (e.g. frame boundaries).
     if (duty == null) duty = 0.5
     if (op   == null) op   = 'add'
     if (amp  == null) amp  = 0.5
     if (pan  == null) pan  = 0.0
+    const tBase = (phaseOffset || 0) + offset
     mixInto(buf, length, offset, op, amp, pan, function(i) {
-        const t = offset + i / HW_SAMPLING_RATE
-        const phase = (t * freq) % 1
+        const phase = ((tBase + i / HW_SAMPLING_RATE) * freq) % 1
         return (phase < duty) ? 1.0 : -1.0
     })
 }
@@ -225,7 +228,7 @@ function lfsrAdvance(state, steps, mode) {
 const LFSR_PERIOD_LONG  = 32767  // mode 0
 const LFSR_PERIOD_SHORT = 93     // mode 1
 
-function makeNoise(buf, length, offset, freq, type, op, amp, pan) {
+function makeNoise(buf, length, offset, freq, type, op, amp, pan, phaseOffset) {
     // buffer: [Uint8Array, Uint8Array] or native buffer
     // length: in seconds
     // offset: in seconds
@@ -238,20 +241,23 @@ function makeNoise(buf, length, offset, freq, type, op, amp, pan) {
     // op: add / mul / sub; default: add
     // amp: 0.0 to 1.0; default: 0.5
     // pan: -1.0 to 1.0; default: 0.0
+    // phaseOffset: optional absolute-time base (seconds) added to phase/LFSR calc only —
+    //              see makeSquare for details.
     //
-    // LFSR types (1 and 2) are deterministic given (offset, freq): calling with
-    // monotonically advancing offset values produces a seamless noise stream
+    // LFSR types (1 and 2) are deterministic given (phaseOffset+offset, freq): calling
+    // with monotonically advancing phaseOffset+offset produces a seamless noise stream
     // across frames. White noise types (-1, 0) are random per call.
     if (op  == null) op  = 'add'
     if (amp == null) amp = 0.5
     if (pan == null) pan = 0.0
+    const tBase = (phaseOffset || 0) + offset
 
     if (type === -1) {
         // 8-bit white: new random float in [-1, 1] each clock period
         let prevClock = -1
         let noiseVal = 0.0
         mixInto(buf, length, offset, op, amp, pan, function(i) {
-            const currentClock = Math.floor((offset + i / HW_SAMPLING_RATE) * freq) | 0
+            const currentClock = Math.floor((tBase + i / HW_SAMPLING_RATE) * freq) | 0
             if (currentClock !== prevClock) {
                 prevClock = currentClock
                 noiseVal = Math.random() * 2.0 - 1.0
@@ -263,7 +269,7 @@ function makeNoise(buf, length, offset, freq, type, op, amp, pan) {
         let prevClock = -1
         let noiseVal = 1.0
         mixInto(buf, length, offset, op, amp, pan, function(i) {
-            const currentClock = Math.floor((offset + i / HW_SAMPLING_RATE) * freq) | 0
+            const currentClock = Math.floor((tBase + i / HW_SAMPLING_RATE) * freq) | 0
             if (currentClock !== prevClock) {
                 prevClock = currentClock
                 noiseVal = (Math.random() >= 0.5) ? 1.0 : -1.0
@@ -274,13 +280,13 @@ function makeNoise(buf, length, offset, freq, type, op, amp, pan) {
         // LFSR-based noise (types 1 and 2)
         const mode   = (type === 2) ? 1 : 0
         const period = (mode === 0) ? LFSR_PERIOD_LONG : LFSR_PERIOD_SHORT
-        // Advance to deterministic position for this offset so consecutive frame
-        // calls with advancing offsets produce a seamless noise stream.
-        const startClock = Math.floor(offset * freq) | 0
+        // Advance to deterministic position for this tBase so consecutive frame
+        // calls with monotonically advancing phaseOffset produce a seamless noise stream.
+        const startClock = Math.floor(tBase * freq) | 0
         let lfsr      = lfsrAdvance(1, startClock % period, mode)
         let prevClock = startClock
         mixInto(buf, length, offset, op, amp, pan, function(i) {
-            const currentClock = Math.floor((offset + i / HW_SAMPLING_RATE) * freq) | 0
+            const currentClock = Math.floor((tBase + i / HW_SAMPLING_RATE) * freq) | 0
             const delta = currentClock - prevClock
             if (delta > 0) {
                 const steps = delta % period
@@ -292,6 +298,25 @@ function makeNoise(buf, length, offset, freq, type, op, amp, pan) {
     }
 }
 
+function makeAliasedTriangleNES(buf, length, offset, freq, duty, op, amp, pan, phaseOffset) {
+    // NES APU triangle — quantised to the authentic 32-step, 4-bit (0..15) staircase.
+    // The 32-step sequence is: 15,14,...,1,0, 0,1,...,14,15 (descending then ascending).
+    // This mirrors the real NES triangle DAC which has 32 equal-height steps per period.
+    // duty parameter is accepted for API symmetry but ignored (NES triangle is always symmetric).
+    // phaseOffset: optional absolute-time base (seconds) — see makeSquare for details.
+    if (op  == null) op  = 'add'
+    if (amp == null) amp = 0.5
+    if (pan == null) pan = 0.0
+    const tBase = (phaseOffset || 0) + offset
+    mixInto(buf, length, offset, op, amp, pan, function(i) {
+        const phase = ((tBase + i / HW_SAMPLING_RATE) * freq) % 1
+        const step32 = Math.floor(phase * 32) | 0  // 0..31
+        // step 0..15: descend from 15 to 0; step 16..31: ascend from 0 to 15
+        const level = (step32 < 16) ? (15 - step32) : (step32 - 16)
+        return level / 7.5 - 1.0  // map 0..15 → -1..+1
+    })
+}
+
 // ── Send to audio hardware ──────────────────────────────────────────────────
 
 function sendBuffer(buf, playhead, offsetSec, lengthSec, stagingPtr) {
@@ -322,8 +347,49 @@ function sendBuffer(buf, playhead, offsetSec, lengthSec, stagingPtr) {
             sys.poke(stagingPtr + 2 * i + 1, readU8(buf, 1, cursor + i))
         }
         // Wait for room in the playback queue (mirrors playwav.js idiom)
-        while (audio.getPosition(playhead) > 2) sys.sleep(2)
-        audio.putPcmDataByPtr(stagingPtr, take * 2, 0)
+        // while (audio.getPosition(playhead) > 2) sys.sleep(2)
+        audio.putPcmDataByPtr(playhead, stagingPtr, take * 2, 0)
+        audio.setSampleUploadLength(playhead, take * 2)
+        audio.startSampleUpload(playhead)
+        remaining -= take
+        cursor    += take
+    }
+
+    if (ownsStaging) sys.free(stagingPtr)
+}
+
+// Lazily-allocated JS-side interleave scratch; shared across sendBufferFast calls.
+let _sendFastScratch = null
+
+function sendBufferFast(buf, playhead, offsetSec, lengthSec, stagingPtr) {
+    // Like sendBuffer but interleaves L/R via a JS Uint8Array + one sys.pokeBytes per chunk,
+    // instead of ~2n sys.poke calls.  Requires a non-native (JS-backed) buffer.
+    // Falls back to sendBuffer for native buffers.
+    if (isNative(buf)) { sendBuffer(buf, playhead, offsetSec, lengthSec, stagingPtr); return }
+
+    const start = (offsetSec != null) ? secToSamples(offsetSec) : 0
+    const total = (lengthSec != null) ? secToSamples(lengthSec) : (buf.samples - start)
+    const MAX_CHUNK = 32768
+    const ownsStaging = (stagingPtr == null)
+    if (ownsStaging) stagingPtr = sys.malloc(Math.min(total, MAX_CHUNK) * 2)
+
+    const scratchNeeded = Math.min(total, MAX_CHUNK) * 2
+    if (_sendFastScratch == null || _sendFastScratch.length < scratchNeeded) {
+        _sendFastScratch = new Uint8Array(scratchNeeded)
+    }
+
+    let remaining = total
+    let cursor    = start
+    while (remaining > 0) {
+        const take = Math.min(remaining, MAX_CHUNK)
+        const L = buf[0], R = buf[1], sc = _sendFastScratch
+        for (let i = 0; i < take; i++) {
+            sc[2 * i]     = L[cursor + i]
+            sc[2 * i + 1] = R[cursor + i]
+        }
+        sys.pokeBytes(stagingPtr, sc.subarray(0, take * 2), take * 2)
+        // while (audio.getPosition(playhead) > 2) sys.sleep(2)
+        audio.putPcmDataByPtr(playhead, stagingPtr, take * 2, 0)
         audio.setSampleUploadLength(playhead, take * 2)
         audio.startSampleUpload(playhead)
         remaining -= take
@@ -336,6 +402,6 @@ function sendBuffer(buf, playhead, offsetSec, lengthSec, stagingPtr) {
 exports = {
     HW_SAMPLING_RATE,
     makeBuffer, makeBufferNative, freeBufferNative, clearBuffer,
-    makeSquare, makeTriangle, makeAliasedTriangle, makeNoise,
-    sendBuffer
+    makeSquare, makeTriangle, makeAliasedTriangle, makeAliasedTriangleNES, makeNoise,
+    sendBuffer, sendBufferFast
 }

terranmon.txt

@@ -1978,6 +1978,11 @@ Sound Adapter
 Endianness: little
 
 
+TSVM Sound Adapter is consisted of 4 playheads, each playhead is capable of playing one PCM or Tracker track.
+
+Synchronisation between playheads are not guaranteed. Do not play music in multiple tracks.
+
+
 Memory Space
 
 0..114687 RW: Sample bin
@@ -2014,17 +2019,17 @@ notes are tuned as 4096 Tone-Equal Temperament. Tuning is set per-sample using t
 
 Sound Adapter MMIO
 
-0..1 RW:  Play head #1 position
-2..3 RW:  Play head #1 length param
-4 RW:     Play head #1 master volume
-5 RW:     Play head #1 master pan
-6..9 RW:  Play head #1 flags
+0..1 RW:  Play head #0 position (how many samples has been queued)
+2..3 RW:  Play head #0 length param
+4 RW:     Play head #0 master volume
+5 RW:     Play head #0 master pan
+6..9 RW:  Play head #0 flags
 
-10..11 RW:Play head #2 position
-12..13 RW:Play head #2 length param
-14 RW:    Play head #2 master volume
-15 RW:    Play head #2 master pan
-16..19 RW:Play head #2 flags
+10..11 RW:Play head #1 position (how many samples has been queued)
+12..13 RW:Play head #1 length param
+14 RW:    Play head #1 master volume
+15 RW:    Play head #1 master pan
+16..19 RW:Play head #1 flags
 
 ... auto-fill to Play head #4
 
@@ -2034,13 +2039,16 @@ Sound Adapter MMIO
 
     When called with byte 17, initialisation will precede before the decoding
 
-41 RO: Media Decoder Status
+41 RO: MP2 Decoder Status
     Non-zero value indicates the decoder is busy
 
 42 WO: TAD Decoder Control
     Write 1 to decode TAD data
 43 RW: TAD Quality
     Must be set to appropriate value before decoding
+44 RW: TAD Decoder Status
+
+45 RW: Select PCM Bin for playhead (writing causes side effects)
 
 64..2367 RW: MP2 Decoded Samples (unsigned 8-bit stereo)
 2368..4095 RW: MP2 Frame to be decoded

tsvm_core/src/net/torvald/tsvm/AudioJSR223Delegate.kt

@@ -5,6 +5,23 @@ import net.torvald.tsvm.peripheral.AudioAdapter
 import net.torvald.tsvm.peripheral.MP2Env
 
 /**
+ * Each playhead is separate OpenAL device with its own PCM sample buffers.
+ * Media decoders (MP2, TAD) are independent to the playheads and there is only one.
+ *
+ * NOTES:
+ * 1. tracker mode is currently unimplemented.
+ * 2. Synchronisation between playheads are not guaranteed. Do not play music in multiple tracks.
+ *
+ * ## How to upload PCM audio into a playhead
+ *
+ * 1. prepare PCM data
+ * 2. queue up PCM data by `audio.putPcmDataByPtr(pcmDataPtr, pcmDataLength, playhead)`
+ * 3. specify PCM upload length by `audio.setSampleUploadLength(playhead, pcmDataLength)`
+ * 4. start uploading `audio.startSampleUpload(playhead)`
+ * 5. sample will be ready after a few microseconds.
+ *
+ * Uploaded samples will be queued by the playhead for gapless playback
+ *
  * Created by minjaesong on 2022-12-31.
  */
 class AudioJSR223Delegate(private val vm: VM) {
@@ -51,16 +68,16 @@ class AudioJSR223Delegate(private val vm: VM) {
     fun setTickRate(playhead: Int, rate: Int) { getPlayhead(playhead)?.tickRate = rate and 255 }
     fun getTickRate(playhead: Int) = getPlayhead(playhead)?.tickRate
 
-    fun putPcmDataByPtr(ptr: Int, length: Int, destOffset: Int) {
+    fun putPcmDataByPtr(playhead: Int, ptr: Int, length: Int, destOffset: Int) {
         getFirstSnd()?.let {
             val vkMult = if (ptr >= 0) 1 else -1
             for (k in 0L until length) {
                 val vk = k * vkMult
-                it.pcmBin[k + destOffset] = vm.peek(ptr + vk)!!
+                it.pcmBin[playhead][k + destOffset] = vm.peek(ptr + vk)!!
             }
         }
     }
-    fun getPcmData(index: Int) = getFirstSnd()?.pcmBin?.get(index.toLong())
+    fun getPcmData(playhead: Int, index: Int) = getFirstSnd()?.pcmBin?.get(playhead)?.get(index.toLong())
 
     fun setPcmQueueCapacityIndex(playhead: Int, index: Int) { getPlayhead(playhead)?.pcmQueueSizeIndex = index }
     fun getPcmQueueCapacityIndex(playhead: Int) { getPlayhead(playhead)?.pcmQueueSizeIndex }

tsvm_core/src/net/torvald/tsvm/VM.kt

@@ -763,7 +763,7 @@ class VM(
             else if (dev is AudioAdapter) {
                 if (relPtrInDev(fromRel, len, 64, 2367)) dev.mediaDecodedBin.ptr + fromRel - 64
                 else if (relPtrInDev(fromRel, len, 2368, 4096)) dev.mediaFrameBin.ptr + fromRel - 2368
-                else if (relPtrInDev(fromRel, len, 65536, 131072)) dev.pcmBin.ptr + fromRel - 65536
+                else if (relPtrInDev(fromRel, len, 65536, 131072)) dev.pcmBin[dev.selectedPcmBin].ptr + fromRel - 65536
                 else null
             }
             else if (dev is GraphicsAdapter) {

tsvm_core/src/net/torvald/tsvm/peripheral/AudioAdapter.kt

@@ -41,7 +41,7 @@ private class RenderRunnable(val playhead: AudioAdapter.Playhead) : Runnable {
 
 //                printdbg("P${playhead.index+1} go back to spinning")
 
-                        Thread.sleep(12)
+                        Thread.sleep(6)
                     }
                     else if (playhead.isPlaying && writeQueue.isEmpty) {
                         printdbg("!! QUEUE EXHAUSTED !! QUEUE EXHAUSTED !! QUEUE EXHAUSTED !! QUEUE EXHAUSTED !! QUEUE EXHAUSTED !! QUEUE EXHAUSTED ")
@@ -49,7 +49,7 @@ private class RenderRunnable(val playhead: AudioAdapter.Playhead) : Runnable {
                         // TODO: wait for 1-2 seconds then finally stop the device
 //                    playhead.audioDevice.stop()
 
-                        Thread.sleep(12)
+                        Thread.sleep(6)
                     }
                 }
 
@@ -122,13 +122,20 @@ class AudioAdapter(val vm: VM) : PeriBase(VM.PERITYPE_SOUND) {
     internal val playdata = Array(256) { Array(64) { TaudPlayData(0,0,0,0,0,0,0,0) } }
     internal val playheads: Array<Playhead>
     internal val cueSheet = Array(2048) { PlayCue() }
-    internal val pcmBin = UnsafeHelper.allocate(65536L, this)
+    internal val pcmBin = arrayOf(
+        UnsafeHelper.allocate(65536L, this),
+        UnsafeHelper.allocate(65536L, this),
+        UnsafeHelper.allocate(65536L, this),
+        UnsafeHelper.allocate(65536L, this),
+    )
 
     internal val mediaFrameBin = UnsafeHelper.allocate(1728, this)
     internal val mediaDecodedBin = UnsafeHelper.allocate(2304, this)
 
     @Volatile private var mp2Busy = false
 
+    @Volatile var selectedPcmBin = 0
+
     // TAD (Terrarum Advanced Audio) decoder buffers
     internal val tadInputBin = UnsafeHelper.allocate(65536L, this)   // Input: compressed TAD chunk (max 64KB)
     internal val tadDecodedBin = UnsafeHelper.allocate(65536L, this) // Output: PCMu8 stereo (32768 samples * 2 channels)
@@ -241,7 +248,7 @@ class AudioAdapter(val vm: VM) : PeriBase(VM.PERITYPE_SOUND) {
 
         renderRunnables = Array(4) { RenderRunnable(playheads[it]) }
         renderThreads = Array(4) { Thread(renderThreadGroup, renderRunnables[it], "AudioRenderHead${it+1}!$hash") }
-        writeQueueingRunnables = Array(4) { WriteQueueingRunnable(playheads[it], pcmBin) }
+        writeQueueingRunnables = Array(4) { WriteQueueingRunnable(playheads[it], pcmBin[it]) }
         writeQueueingThreads = Array(4) { Thread(writeQueueingGroup, writeQueueingRunnables[it], "AudioQueueingHead${it+1}!$hash") }
 
 //        printdbg("AudioAdapter latency: ${audioDevice.latency}")
@@ -315,13 +322,14 @@ class AudioAdapter(val vm: VM) : PeriBase(VM.PERITYPE_SOUND) {
             42 -> -1  // TAD control (write-only)
             43 -> tadQuality.toByte()
             44 -> tadBusy.toInt().toByte()
+            45 -> selectedPcmBin.toByte()
             in 64..2367 -> mediaDecodedBin[addr - 64]
             in 2368..4095 -> mediaFrameBin[addr - 2368]
             in 4096..4097 -> 0
             in 32768..65535 -> (adi - 32768).let {
                 cueSheet[it / 16].read(it % 15)
             }
-            in 65536..131071 -> pcmBin[addr - 65536]
+            in 65536..131071 -> pcmBin[selectedPcmBin][addr - 65536]
             else -> {
                 println("[AudioAdapter] Bus mirroring on mmio_reading while trying to read address $addr")
                 mmio_read(addr % 131072)
@@ -349,12 +357,13 @@ class AudioAdapter(val vm: VM) : PeriBase(VM.PERITYPE_SOUND) {
                 // TAD quality (0-5)
                 tadQuality = bi.coerceIn(0, 5)
             }
+            45 -> selectedPcmBin = bi % 4
             in 64..2367 -> { mediaDecodedBin[addr - 64] = byte }
             in 2368..4095 -> { mediaFrameBin[addr - 2368] = byte }
             in 32768..65535 -> { (adi - 32768).let {
                 cueSheet[it / 16].write(it % 15, bi)
             } }
-            in 65536..131071 -> { pcmBin[addr - 65536] = byte }
+            in 65536..131071 -> { pcmBin[selectedPcmBin][addr - 65536] = byte }
         }
     }
 
@@ -368,7 +377,7 @@ class AudioAdapter(val vm: VM) : PeriBase(VM.PERITYPE_SOUND) {
             writeQueueingGroup.interrupt()
             playheads.forEach { it.dispose() }
             sampleBin.destroy()
-            pcmBin.destroy()
+            pcmBin.forEach { it.destroy() }
             mediaFrameBin.destroy()
             mediaDecodedBin.destroy()
             tadInputBin.destroy()