[PATCH v6] dsound: Commit next audio chunk between play cursor and write cursor to playing.

14 Sep 2021

This region of the audio buffer is forbidden to be written to by the
DirectSound specification. The documentation states: "The write cursor
is the point after which it is safe to write data into the buffer. The
block between the play cursor and the write cursor is already committed
to be played, and cannot be changed safely." However, some applications
still do this, which has lead to audio glitches only when using the Wine
DirectSound implementation. Experiments showed that the native DirctSound
implementation will still play the old audio the first time around when the
buffer region gets overwritten. Use an approach of copying the next forbidden
region into a "committed buffer" to add the same behavior to the Wine
implementation.

Out of performance considerations, only copy data to the committed buffer
when we detect that an overwrite is possible (i.e. the current mixing
region of the buffer gets locked).

Signed-off-by: Eduard Permyakov <epermyakov(a)codeweavers.com>
---
 dlls/dsound/buffer.c         | 66 ++++++++++++++++++++++++++++++++++--
 dlls/dsound/dsound_convert.c | 33 +++++++++---------
 dlls/dsound/dsound_private.h | 11 ++++--
 dlls/dsound/mixer.c          | 48 ++++++++++++++++++++------
 4 files changed, 125 insertions(+), 33 deletions(-)

diff --git a/dlls/dsound/buffer.c b/dlls/dsound/buffer.c
index dc3b54906ce..c595b19dd63 100644
--- a/dlls/dsound/buffer.c
+++ b/dlls/dsound/buffer.c
@@ -101,6 +101,27 @@ static int __cdecl notify_compar(const void *l, const void *r)
     return 1;
 }
 
+static void commit_next_chunk(IDirectSoundBufferImpl *dsb)
+{
+    void *dstbuff = dsb->committedbuff, *srcbuff = dsb->buffer->memory;
+    DWORD srcoff = dsb->sec_mixpos, srcsize = dsb->buflen, cpysize = dsb->writelead;
+
+    if(dsb->state != STATE_PLAYING)
+        return;
+
+    if(cpysize > srcsize - srcoff) {
+        DWORD overflow = cpysize - (srcsize - srcoff);
+        memcpy(dstbuff, (BYTE*)srcbuff + srcoff, srcsize - srcoff);
+        memcpy((BYTE*)dstbuff + (srcsize - srcoff), srcbuff, overflow);
+    }else{
+        memcpy(dstbuff, (BYTE*)srcbuff + srcoff, cpysize);
+    }
+
+    dsb->use_committed = TRUE;
+    dsb->committed_mixpos = 0;
+    TRACE("committing %u bytes from offset %u\n", dsb->writelead, dsb->sec_mixpos);
+}
+
 static HRESULT WINAPI IDirectSoundNotifyImpl_SetNotificationPositions(IDirectSoundNotify *iface,
         DWORD howmuch, const DSBPOSITIONNOTIFY *notify)
 {
@@ -244,6 +265,7 @@ static HRESULT WINAPI IDirectSoundBufferImpl_SetFrequency(IDirectSoundBuffer8 *i
 {
         IDirectSoundBufferImpl *This = impl_from_IDirectSoundBuffer8(iface);
 	DWORD oldFreq;
+	void *newcommitted;
 
 	TRACE("(%p,%d)\n",This,freq);
 
@@ -274,6 +296,13 @@ static HRESULT WINAPI IDirectSoundBufferImpl_SetFrequency(IDirectSoundBuffer8 *i
 		This->freqAdjustDen = This->device->pwfx->nSamplesPerSec;
 		This->nAvgBytesPerSec = freq * This->pwfx->nBlockAlign;
 		DSOUND_RecalcFormat(This);
+
+		newcommitted = HeapReAlloc(GetProcessHeap(), 0, This->committedbuff, This->writelead);
+		if(!newcommitted) {
+			ReleaseSRWLockExclusive(&This->lock);
+			return DSERR_OUTOFMEMORY;
+		}
+		This->committedbuff = newcommitted;
 	}
 
 	ReleaseSRWLockExclusive(&This->lock);
@@ -319,6 +348,8 @@ static HRESULT WINAPI IDirectSoundBufferImpl_Stop(IDirectSoundBuffer8 *iface)
 	if (This->state == STATE_PLAYING || This->state == STATE_STARTING)
 	{
 		This->state = STATE_STOPPED;
+		This->use_committed = FALSE;
+		This->committed_mixpos = 0;
 		DSOUND_CheckEvent(This, 0, 0);
 	}
 
@@ -503,8 +534,10 @@ static HRESULT WINAPI IDirectSoundBufferImpl_Lock(IDirectSoundBuffer8 *iface, DW
 
 	if (writecursor+writebytes <= This->buflen) {
 		*(LPBYTE*)lplpaudioptr1 = This->buffer->memory+writecursor;
-		if (This->sec_mixpos >= writecursor && This->sec_mixpos < writecursor + writebytes && This->state == STATE_PLAYING)
+		if (This->sec_mixpos >= writecursor && This->sec_mixpos < writecursor + writebytes && This->state == STATE_PLAYING) {
 			WARN("Overwriting mixing position, case 1\n");
+			commit_next_chunk(This);
+		}
 		*audiobytes1 = writebytes;
 		if (lplpaudioptr2)
 			*(LPBYTE*)lplpaudioptr2 = NULL;
@@ -519,16 +552,20 @@ static HRESULT WINAPI IDirectSoundBufferImpl_Lock(IDirectSoundBuffer8 *iface, DW
 		*(LPBYTE*)lplpaudioptr1 = This->buffer->memory+writecursor;
 		*audiobytes1 = This->buflen-writecursor;
 		This->buffer->lockedbytes += *audiobytes1;
-		if (This->sec_mixpos >= writecursor && This->sec_mixpos < writecursor + writebytes && This->state == STATE_PLAYING)
+		if (This->sec_mixpos >= writecursor && This->sec_mixpos < writecursor + writebytes && This->state == STATE_PLAYING) {
 			WARN("Overwriting mixing position, case 2\n");
+			commit_next_chunk(This);
+		}
 		if (lplpaudioptr2)
 			*(LPBYTE*)lplpaudioptr2 = This->buffer->memory;
 		if (audiobytes2) {
 			*audiobytes2 = writebytes-(This->buflen-writecursor);
 			This->buffer->lockedbytes += *audiobytes2;
 		}
-		if (audiobytes2 && This->sec_mixpos < remainder && This->state == STATE_PLAYING)
+		if (audiobytes2 && This->sec_mixpos < remainder && This->state == STATE_PLAYING) {
 			WARN("Overwriting mixing position, case 3\n");
+			commit_next_chunk(This);
+		}
 		TRACE("Locked %p(%i bytes) and %p(%i bytes) writecursor=%d\n", *(LPBYTE*)lplpaudioptr1, *audiobytes1, lplpaudioptr2 ? *(LPBYTE*)lplpaudioptr2 : NULL, audiobytes2 ? *audiobytes2: 0, writecursor);
 	}
 
@@ -552,6 +589,9 @@ static HRESULT WINAPI IDirectSoundBufferImpl_SetCurrentPosition(IDirectSoundBuff
 	newpos -= newpos%This->pwfx->nBlockAlign;
 	This->sec_mixpos = newpos;
 
+	This->use_committed = FALSE;
+	This->committed_mixpos = 0;
+
 	/* at this point, do not attempt to reset buffers, mess with primary mix position,
            or anything like that to reduce latency. The data already prebuffered cannot be changed */
 
@@ -1081,6 +1121,12 @@ HRESULT secondarybuffer_create(DirectSoundDevice *device, const DSBUFFERDESC *ds
 	/* calculate fragment size and write lead */
 	DSOUND_RecalcFormat(dsb);
 
+	dsb->committedbuff = HeapAlloc(GetProcessHeap(), 0, dsb->writelead);
+	if(!dsb->committedbuff) {
+		IDirectSoundBuffer8_Release(&dsb->IDirectSoundBuffer8_iface);
+		return DSERR_OUTOFMEMORY;
+	}
+
 	if (dsb->dsbd.dwFlags & DSBCAPS_CTRL3D) {
 		dsb->ds3db_ds3db.dwSize = sizeof(DS3DBUFFER);
 		dsb->ds3db_ds3db.vPosition.x = 0.0;
@@ -1135,6 +1181,7 @@ void secondarybuffer_destroy(IDirectSoundBufferImpl *This)
 
     HeapFree(GetProcessHeap(), 0, This->notifies);
     HeapFree(GetProcessHeap(), 0, This->pwfx);
+    HeapFree(GetProcessHeap(), 0, This->committedbuff);
 
     if (This->filters) {
         int i;
@@ -1157,6 +1204,7 @@ HRESULT IDirectSoundBufferImpl_Duplicate(
 {
     IDirectSoundBufferImpl *dsb;
     HRESULT hres = DS_OK;
+    VOID *committedbuff;
     TRACE("(%p,%p,%p)\n", device, ppdsb, pdsb);
 
     dsb = HeapAlloc(GetProcessHeap(),0,sizeof(*dsb));
@@ -1166,6 +1214,13 @@ HRESULT IDirectSoundBufferImpl_Duplicate(
         return DSERR_OUTOFMEMORY;
     }
 
+    committedbuff = HeapAlloc(GetProcessHeap(),0,pdsb->writelead);
+    if (committedbuff == NULL) {
+        HeapFree(GetProcessHeap(),0,dsb);
+        *ppdsb = NULL;
+        return DSERR_OUTOFMEMORY;
+    }
+
     AcquireSRWLockShared(&pdsb->lock);
 
     CopyMemory(dsb, pdsb, sizeof(*dsb));
@@ -1175,6 +1230,7 @@ HRESULT IDirectSoundBufferImpl_Duplicate(
     ReleaseSRWLockShared(&pdsb->lock);
 
     if (dsb->pwfx == NULL) {
+        HeapFree(GetProcessHeap(),0,committedbuff);
         HeapFree(GetProcessHeap(),0,dsb);
         *ppdsb = NULL;
         return DSERR_OUTOFMEMORY;
@@ -1192,6 +1248,9 @@ HRESULT IDirectSoundBufferImpl_Duplicate(
     dsb->notifies = NULL;
     dsb->nrofnotifies = 0;
     dsb->device = device;
+    dsb->committedbuff = committedbuff;
+    dsb->use_committed = FALSE;
+    dsb->committed_mixpos = 0;
     DSOUND_RecalcFormat(dsb);
 
     InitializeSRWLock(&dsb->lock);
@@ -1202,6 +1261,7 @@ HRESULT IDirectSoundBufferImpl_Duplicate(
         list_remove(&dsb->entry);
         dsb->buffer->ref--;
         HeapFree(GetProcessHeap(),0,dsb->pwfx);
+        HeapFree(GetProcessHeap(),0,dsb->committedbuff);
         HeapFree(GetProcessHeap(),0,dsb);
         dsb = NULL;
     }else
diff --git a/dlls/dsound/dsound_convert.c b/dlls/dsound/dsound_convert.c
index 4735178a9a8..67344565860 100644
--- a/dlls/dsound/dsound_convert.c
+++ b/dlls/dsound/dsound_convert.c
@@ -55,26 +55,25 @@ WINE_DEFAULT_DEBUG_CHANNEL(dsound);
 #define le32(x) (x)
 #endif
 
-static float get8(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
+static float get8(const IDirectSoundBufferImpl *dsb, BYTE *base, DWORD channel)
 {
-    const BYTE* buf = dsb->buffer->memory;
-    buf += pos + channel;
+    const BYTE *buf = base + channel;
     return (buf[0] - 0x80) / (float)0x80;
 }
 
-static float get16(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
+static float get16(const IDirectSoundBufferImpl *dsb, BYTE *base, DWORD channel)
 {
-    const BYTE* buf = dsb->buffer->memory;
-    const SHORT *sbuf = (const SHORT*)(buf + pos + 2 * channel);
+    const BYTE *buf = base + 2 * channel;
+    const SHORT *sbuf = (const SHORT*)(buf);
     SHORT sample = (SHORT)le16(*sbuf);
     return sample / (float)0x8000;
 }
 
-static float get24(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
+static float get24(const IDirectSoundBufferImpl *dsb, BYTE *base, DWORD channel)
 {
     LONG sample;
-    const BYTE* buf = dsb->buffer->memory;
-    buf += pos + 3 * channel;
+    const BYTE *buf = base + 3 * channel;
+
     /* The next expression deliberately has an overflow for buf[2] >= 0x80,
        this is how negative values are made.
      */
@@ -82,32 +81,32 @@ static float get24(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
     return sample / (float)0x80000000U;
 }
 
-static float get32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
+static float get32(const IDirectSoundBufferImpl *dsb, BYTE *base, DWORD channel)
 {
-    const BYTE* buf = dsb->buffer->memory;
-    const LONG *sbuf = (const LONG*)(buf + pos + 4 * channel);
+    const BYTE *buf = base + 4 * channel;
+    const LONG *sbuf = (const LONG*)(buf);
     LONG sample = le32(*sbuf);
     return sample / (float)0x80000000U;
 }
 
-static float getieee32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
+static float getieee32(const IDirectSoundBufferImpl *dsb, BYTE *base, DWORD channel)
 {
-    const BYTE* buf = dsb->buffer->memory;
-    const float *sbuf = (const float*)(buf + pos + 4 * channel);
+    const BYTE *buf = base + 4 * channel;
+    const float *sbuf = (const float*)(buf);
     /* The value will be clipped later, when put into some non-float buffer */
     return *sbuf;
 }
 
 const bitsgetfunc getbpp[5] = {get8, get16, get24, get32, getieee32};
 
-float get_mono(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
+float get_mono(const IDirectSoundBufferImpl *dsb, BYTE *base, DWORD channel)
 {
     DWORD channels = dsb->pwfx->nChannels;
     DWORD c;
     float val = 0;
     /* XXX: does Windows include LFE into the mix? */
     for (c = 0; c < channels; c++)
-        val += dsb->get_aux(dsb, pos, c);
+        val += dsb->get_aux(dsb, base, c);
     val /= channels;
     return val;
 }
diff --git a/dlls/dsound/dsound_private.h b/dlls/dsound/dsound_private.h
index bdb9ebee544..9ec96504a50 100644
--- a/dlls/dsound/dsound_private.h
+++ b/dlls/dsound/dsound_private.h
@@ -43,7 +43,7 @@ typedef struct IDirectSoundBufferImpl        IDirectSoundBufferImpl;
 typedef struct DirectSoundDevice             DirectSoundDevice;
 
 /* dsound_convert.h */
-typedef float (*bitsgetfunc)(const IDirectSoundBufferImpl *, DWORD, DWORD);
+typedef float (*bitsgetfunc)(const IDirectSoundBufferImpl *, BYTE *, DWORD);
 typedef void (*bitsputfunc)(const IDirectSoundBufferImpl *, DWORD, DWORD, float);
 extern const bitsgetfunc getbpp[5] DECLSPEC_HIDDEN;
 void putieee32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value) DECLSPEC_HIDDEN;
@@ -153,7 +153,12 @@ struct IDirectSoundBufferImpl
     LONG64                      freqAccNum;
     /* used for mixing */
     DWORD                       sec_mixpos;
-
+    /* Holds a copy of the next 'writelead' bytes, to be used for mixing. This makes it
+     * so that these bytes get played once even if this region of the buffer gets overwritten,
+     * which is more in-line with native DirectSound behavior. */
+    BOOL                        use_committed;
+    LPVOID                      committedbuff;
+    DWORD                       committed_mixpos;
     /* IDirectSoundNotify fields */
     LPDSBPOSITIONNOTIFY         notifies;
     int                         nrofnotifies;
@@ -171,7 +176,7 @@ struct IDirectSoundBufferImpl
     struct list entry;
 };
 
-float get_mono(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel) DECLSPEC_HIDDEN;
+float get_mono(const IDirectSoundBufferImpl *dsb, BYTE *base, DWORD channel) DECLSPEC_HIDDEN;
 void put_mono2stereo(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value) DECLSPEC_HIDDEN;
 void put_mono2quad(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value) DECLSPEC_HIDDEN;
 void put_stereo2quad(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value) DECLSPEC_HIDDEN;
diff --git a/dlls/dsound/mixer.c b/dlls/dsound/mixer.c
index 124d1e4fba1..e9d8a6e3cdd 100644
--- a/dlls/dsound/mixer.c
+++ b/dlls/dsound/mixer.c
@@ -276,22 +276,35 @@ void DSOUND_CheckEvent(const IDirectSoundBufferImpl *dsb, DWORD playpos, int len
 }
 
 static inline float get_current_sample(const IDirectSoundBufferImpl *dsb,
-        DWORD mixpos, DWORD channel)
+        BYTE *buffer, DWORD buflen, DWORD mixpos, DWORD channel)
 {
-    if (mixpos >= dsb->buflen && !(dsb->playflags & DSBPLAY_LOOPING))
+    if (mixpos >= buflen && !(dsb->playflags & DSBPLAY_LOOPING))
         return 0.0f;
-    return dsb->get(dsb, mixpos % dsb->buflen, channel);
+    return dsb->get(dsb, buffer + (mixpos % buflen), channel);
 }
 
 static UINT cp_fields_noresample(IDirectSoundBufferImpl *dsb, UINT count)
 {
     UINT istride = dsb->pwfx->nBlockAlign;
     UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);
+    UINT committed_samples = 0;
     DWORD channel, i;
-    for (i = 0; i < count; i++)
+
+    if(dsb->use_committed) {
+        committed_samples = (dsb->writelead - dsb->committed_mixpos) / istride;
+        committed_samples = committed_samples <= count ? committed_samples : count;
+    }
+
+    for (i = 0; i < committed_samples; i++)
+        for (channel = 0; channel < dsb->mix_channels; channel++)
+            dsb->put(dsb, i * ostride, channel, get_current_sample(dsb, dsb->committedbuff,
+                dsb->writelead, dsb->committed_mixpos + i * istride, channel));
+
+    for (; i < count; i++)
         for (channel = 0; channel < dsb->mix_channels; channel++)
-            dsb->put(dsb, i * ostride, channel, get_current_sample(dsb,
-                    dsb->sec_mixpos + i * istride, channel));
+            dsb->put(dsb, i * ostride, channel, get_current_sample(dsb, dsb->buffer->memory,
+                dsb->buflen, dsb->sec_mixpos + i * istride, channel));
+
     return count;
 }
 
@@ -300,6 +313,7 @@ static UINT cp_fields_resample(IDirectSoundBufferImpl *dsb, UINT count, LONG64 *
     UINT i, channel;
     UINT istride = dsb->pwfx->nBlockAlign;
     UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);
+    UINT committed_samples = 0;
 
     LONG64 freqAcc_start = *freqAccNum;
     LONG64 freqAcc_end = freqAcc_start + count * dsb->freqAdjustNum;
@@ -326,16 +340,24 @@ static UINT cp_fields_resample(IDirectSoundBufferImpl *dsb, UINT count, LONG64 *
     fir_copy = dsb->device->cp_buffer;
     intermediate = fir_copy + fir_cachesize;
 
+    if(dsb->use_committed) {
+        committed_samples = (dsb->writelead - dsb->committed_mixpos) / istride;
+        committed_samples = committed_samples <= required_input ? committed_samples : required_input;
+    }
 
     /* Important: this buffer MUST be non-interleaved
      * if you want -msse3 to have any effect.
      * This is good for CPU cache effects, too.
      */
     itmp = intermediate;
-    for (channel = 0; channel < channels; channel++)
-        for (i = 0; i < required_input; i++)
-            *(itmp++) = get_current_sample(dsb,
-                    dsb->sec_mixpos + i * istride, channel);
+    for (channel = 0; channel < channels; channel++) {
+        for (i = 0; i < committed_samples; i++)
+            *(itmp++) = get_current_sample(dsb, dsb->committedbuff,
+                dsb->writelead, dsb->committed_mixpos + i * istride, channel);
+        for (; i < required_input; i++)
+            *(itmp++) = get_current_sample(dsb, dsb->buffer->memory,
+                    dsb->buflen, dsb->sec_mixpos + i * istride, channel);
+    }
 
     for(i = 0; i < count; ++i) {
         UINT int_fir_steps = (freqAcc_start + i * dsb->freqAdjustNum) * dsbfirstep / dsb->freqAdjustDen;
@@ -389,6 +411,12 @@ static void cp_fields(IDirectSoundBufferImpl *dsb, UINT count, LONG64 *freqAccNu
     }
 
     dsb->sec_mixpos = ipos;
+
+    if(dsb->use_committed) {
+        dsb->committed_mixpos += adv * dsb->pwfx->nBlockAlign;
+        if(dsb->committed_mixpos >= dsb->writelead)
+            dsb->use_committed = FALSE;
+    }
 }
 
 /**
-- 
2.25.1

    

Eduard Permyakov

Andrew Eikum

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