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Added patch to clear the dsound mixing buffer after allocation.
This commit is contained in:
Sebastian Lackner
@@ -1,4 +1,4 @@
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From ed0264f5193ecf9d7724be5de6c9bda17919105f Mon Sep 17 00:00:00 2001
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From b23019f93121d8a1a57912c4ca335fa1c534c8d7 Mon Sep 17 00:00:00 2001
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From: Sebastian Lackner <sebastian@fds-team.de>
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Date: Sat, 28 Mar 2015 08:18:10 +0100
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Subject: dsound: Apply filters before sound is multiplied to speakers.
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@@ -7,11 +7,11 @@ Based on a patch by Mark Harmstone.
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---
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dlls/dsound/dsound.c | 1 +
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dlls/dsound/dsound_private.h | 4 +-
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dlls/dsound/mixer.c | 104 ++++++++++++++++++++++++++++++-------------
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3 files changed, 77 insertions(+), 32 deletions(-)
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dlls/dsound/mixer.c | 108 ++++++++++++++++++++++++++++++-------------
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3 files changed, 79 insertions(+), 34 deletions(-)
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diff --git a/dlls/dsound/dsound.c b/dlls/dsound/dsound.c
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index 64bfbd3..d73610e 100644
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index 4d515adcaf3..65a00787098 100644
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--- a/dlls/dsound/dsound.c
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+++ b/dlls/dsound/dsound.c
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@@ -233,6 +233,7 @@ static ULONG DirectSoundDevice_Release(DirectSoundDevice * device)
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@@ -23,10 +23,10 @@ index 64bfbd3..d73610e 100644
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HeapFree(GetProcessHeap(), 0, device->cp_buffer);
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HeapFree(GetProcessHeap(), 0, device->buffer);
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diff --git a/dlls/dsound/dsound_private.h b/dlls/dsound/dsound_private.h
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index 34bab65..f580915 100644
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index 8e1e2dafc7c..f22849fb0a2 100644
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--- a/dlls/dsound/dsound_private.h
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+++ b/dlls/dsound/dsound_private.h
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@@ -88,8 +88,8 @@ struct DirectSoundDevice
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@@ -89,8 +89,8 @@ struct DirectSoundDevice
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int speaker_num[DS_MAX_CHANNELS];
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int num_speakers;
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int lfe_channel;
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@@ -38,10 +38,10 @@ index 34bab65..f580915 100644
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DSVOLUMEPAN volpan;
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diff --git a/dlls/dsound/mixer.c b/dlls/dsound/mixer.c
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index a2fd768..db4235d 100644
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index 1a99eebf30c..5b4af70093f 100644
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--- a/dlls/dsound/mixer.c
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+++ b/dlls/dsound/mixer.c
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@@ -266,23 +266,22 @@ static inline float get_current_sample(const IDirectSoundBufferImpl *dsb,
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@@ -278,23 +278,22 @@ static inline float get_current_sample(const IDirectSoundBufferImpl *dsb,
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return dsb->get(dsb, mixpos % dsb->buflen, channel);
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}
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@@ -70,7 +70,7 @@ index a2fd768..db4235d 100644
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UINT channels = dsb->mix_channels;
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LONG64 freqAcc_start = *freqAccNum;
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@@ -310,7 +309,7 @@ static UINT cp_fields_resample_lq(IDirectSoundBufferImpl *dsb, UINT count, LONG6
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@@ -322,7 +321,7 @@ static UINT cp_fields_resample_lq(IDirectSoundBufferImpl *dsb, UINT count, LONG6
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float s1 = get_current_sample(dsb, idx, channel);
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float s2 = get_current_sample(dsb, idx + istride, channel);
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float result = s1 * cur_freqAcc2 + s2 * cur_freqAcc;
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@@ -79,7 +79,7 @@ index a2fd768..db4235d 100644
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}
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}
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@@ -318,11 +317,11 @@ static UINT cp_fields_resample_lq(IDirectSoundBufferImpl *dsb, UINT count, LONG6
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@@ -330,11 +329,11 @@ static UINT cp_fields_resample_lq(IDirectSoundBufferImpl *dsb, UINT count, LONG6
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return max_ipos;
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}
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@@ -93,7 +93,7 @@ index a2fd768..db4235d 100644
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LONG64 freqAcc_start = *freqAccNum;
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LONG64 freqAcc_end = freqAcc_start + count * dsb->freqAdjustNum;
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@@ -383,7 +382,7 @@ static UINT cp_fields_resample_hq(IDirectSoundBufferImpl *dsb, UINT count, LONG6
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@@ -395,7 +394,7 @@ static UINT cp_fields_resample_hq(IDirectSoundBufferImpl *dsb, UINT count, LONG6
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float* cache = &intermediate[channel * required_input + ipos];
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for (j = 0; j < fir_used; j++)
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sum += fir_copy[j] * cache[j];
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@@ -102,7 +102,7 @@ index a2fd768..db4235d 100644
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}
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}
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@@ -392,16 +391,17 @@ static UINT cp_fields_resample_hq(IDirectSoundBufferImpl *dsb, UINT count, LONG6
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@@ -404,16 +403,17 @@ static UINT cp_fields_resample_hq(IDirectSoundBufferImpl *dsb, UINT count, LONG6
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return max_ipos;
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}
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@@ -124,7 +124,7 @@ index a2fd768..db4235d 100644
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ipos = dsb->sec_mixpos + adv * dsb->pwfx->nBlockAlign;
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if (ipos >= dsb->buflen) {
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@@ -431,6 +431,21 @@ static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
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@@ -443,6 +443,21 @@ static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
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return buflen + ptr1 - ptr2;
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}
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}
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@@ -146,7 +146,7 @@ index a2fd768..db4235d 100644
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/**
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* Mix at most the given amount of data into the allocated temporary buffer
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* of the given secondary buffer, starting from the dsb's first currently
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@@ -446,32 +461,61 @@ static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
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@@ -458,35 +473,64 @@ static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
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*/
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static void DSOUND_MixToTemporary(IDirectSoundBufferImpl *dsb, DWORD frames)
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{
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@@ -173,12 +173,21 @@ index a2fd768..db4235d 100644
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+ dsb->device->tmp_buffer_len = size_bytes;
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}
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- if(dsb->put_aux == putieee32_sum)
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- memset(dsb->device->tmp_buffer, 0, dsb->device->tmp_buffer_len);
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+ if(dsb->put_aux == putieee32_sum)
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+ memset(dsb->device->tmp_buffer, 0, dsb->device->tmp_buffer_len);
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- cp_fields(dsb, frames, &dsb->freqAccNum);
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+ if (using_filters) {
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+ put = putieee32_dsp;
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+ ostride = dsb->mix_channels * sizeof(float);
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+ size_bytes = frames * ostride;
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+
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- if (size_bytes > 0) {
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- for (i = 0; i < dsb->num_filters; i++) {
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- if (dsb->filters[i].inplace) {
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- hr = IMediaObjectInPlace_Process(dsb->filters[i].inplace, size_bytes, (BYTE*)dsb->device->tmp_buffer, 0, DMO_INPLACE_NORMAL);
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+ if (dsb->device->dsp_buffer_len < size_bytes || !dsb->device->dsp_buffer) {
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+ if (dsb->device->dsp_buffer)
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+ dsb->device->dsp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->dsp_buffer, size_bytes);
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@@ -188,10 +197,12 @@ index a2fd768..db4235d 100644
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+ }
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+ }
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- if (size_bytes > 0) {
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- for (i = 0; i < dsb->num_filters; i++) {
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- if (dsb->filters[i].inplace) {
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- hr = IMediaObjectInPlace_Process(dsb->filters[i].inplace, size_bytes, (BYTE*)dsb->device->tmp_buffer, 0, DMO_INPLACE_NORMAL);
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- if (FAILED(hr))
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- WARN("IMediaObjectInPlace_Process failed for filter %u\n", i);
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- } else
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- WARN("filter %u has no inplace object - unsupported\n", i);
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- }
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- }
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+ cp_fields(dsb, put, ostride, frames, &dsb->freqAccNum);
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+
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+ if (using_filters) {
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@@ -206,13 +217,7 @@ index a2fd768..db4235d 100644
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+ WARN("filter %u has no inplace object - unsupported\n", i);
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+ }
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+ }
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- if (FAILED(hr))
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- WARN("IMediaObjectInPlace_Process failed for filter %u\n", i);
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- } else
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- WARN("filter %u has no inplace object - unsupported\n", i);
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- }
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- }
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+
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+ istride = ostride;
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+ ostride = dsb->device->pwfx->nChannels * sizeof(float);
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+ for (i = 0; i < frames; i++) {
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@@ -225,5 +230,5 @@ index a2fd768..db4235d 100644
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static void DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT frames)
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--
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2.8.0
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2.11.0
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@@ -1,3 +1,4 @@
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Fixes: Software support for Environmental Audio Extensions (EAX)
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Depends: dsound-Fast_Mixer
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Depends: dsound-Revert_Cleanup
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Depends: dsound-Clear_Mixing_Buffer
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