[PATCH 1/5] winegstreamer: Report MEDIASUBTYPE_IEEE_FLOAT for float types.

[Zebediah Figura]

Signed-off-by: Zebediah Figura <z.figura12(a)gmail.com>
---
 dlls/winegstreamer/gstdemux.c | 1 +
 1 file changed, 1 insertion(+)

diff --git a/dlls/winegstreamer/gstdemux.c b/dlls/winegstreamer/gstdemux.c
index 8213bfdb439..de508b4167b 100644
--- a/dlls/winegstreamer/gstdemux.c
+++ b/dlls/winegstreamer/gstdemux.c
@@ -169,6 +169,7 @@ static gboolean amt_from_gst_caps_audio_raw(const GstCaps *caps, AM_MEDIA_TYPE *
         wfe->dwChannelMask = 0;
     }
     if (GST_AUDIO_INFO_IS_FLOAT(&ainfo)) {
+        amt->subtype = MEDIASUBTYPE_IEEE_FLOAT;
         wfe->SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
         wfx->wBitsPerSample = wfe->Samples.wValidBitsPerSample = 32;
     } else {
-- 
2.25.0

[PATCH 2/5] winegstreamer: Correctly represent non-32-bit float types.

[Zebediah Figura]

Signed-off-by: Zebediah Figura <z.figura12(a)gmail.com>
---
 dlls/winegstreamer/gstdemux.c | 1 -
 1 file changed, 1 deletion(-)

diff --git a/dlls/winegstreamer/gstdemux.c b/dlls/winegstreamer/gstdemux.c
index de508b4167b..66557b63a59 100644
--- a/dlls/winegstreamer/gstdemux.c
+++ b/dlls/winegstreamer/gstdemux.c
@@ -171,7 +171,6 @@ static gboolean amt_from_gst_caps_audio_raw(const GstCaps *caps, AM_MEDIA_TYPE *
     if (GST_AUDIO_INFO_IS_FLOAT(&ainfo)) {
         amt->subtype = MEDIASUBTYPE_IEEE_FLOAT;
         wfe->SubFormat = KSDATAFORMAT_SUBTYPE_IEEE_FLOAT;
-        wfx->wBitsPerSample = wfe->Samples.wValidBitsPerSample = 32;
     } else {
         wfe->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
         if (wfx->nChannels <= 2 && bpp <= 16 && depth == bpp)  {
-- 
2.25.0

[PATCH 3/5] winegstreamer: Append an audioconvert element to raw audio streams.

[Zebediah Figura]

Signed-off-by: Zebediah Figura <z.figura12(a)gmail.com>
---
 dlls/winegstreamer/gstdemux.c | 155 +++++++++++++++++-----------------
 1 file changed, 78 insertions(+), 77 deletions(-)

diff --git a/dlls/winegstreamer/gstdemux.c b/dlls/winegstreamer/gstdemux.c
index 66557b63a59..8e36644030a 100644
--- a/dlls/winegstreamer/gstdemux.c
+++ b/dlls/winegstreamer/gstdemux.c
@@ -85,10 +85,7 @@ struct gstdemux_source
     struct strmbase_source pin;
     IQualityControl IQualityControl_iface;
 
-    GstElement *flipfilter;
-    GstPad *flip_sink, *flip_src;
-    GstPad *their_src;
-    GstPad *my_sink;
+    GstPad *their_src, *post_sink, *post_src, *my_sink;
     AM_MEDIA_TYPE mt;
     HANDLE caps_event;
     GstSegment *segment;
@@ -1004,11 +1001,10 @@ static void removed_decoded_pad(GstElement *bin, GstPad *pad, gpointer user)
 
         if (pin->their_src == pad)
         {
-            if(pin->flipfilter)
-                gst_pad_unlink(pin->their_src, pin->flip_sink);
+            if (pin->post_sink)
+                gst_pad_unlink(pin->their_src, pin->post_sink);
             else
                 gst_pad_unlink(pin->their_src, pin->my_sink);
-
             gst_object_unref(pin->their_src);
             pin->their_src = NULL;
             return;
@@ -1052,85 +1048,91 @@ static void init_new_decoded_pad(GstElement *bin, GstPad *pad, struct gstdemux *
 
     if (!strcmp(typename, "video/x-raw"))
     {
-        GstElement *vconv;
-
-        TRACE("setting up videoflip filter for pin %p, my_sink: %p, their_src: %p\n",
-                pin, pin->my_sink, pad);
+        GstElement *vconv, *flip;
 
-        /* gstreamer outputs video top-down, but dshow expects bottom-up, so
-         * make new transform filter to invert video */
-        vconv = gst_element_factory_make("videoconvert", NULL);
-        if(!vconv){
-            ERR("Missing videoconvert filter?\n");
-            ret = -1;
-            goto exit;
+        /* decodebin considers many YUV formats to be "raw", but some quartz
+         * filters can't handle those. Also, videoflip can't handle all "raw"
+         * formats either. Add a videoconvert to swap color spaces. */
+        if (!(vconv = gst_element_factory_make("videoconvert", NULL)))
+        {
+            ERR("Failed to create videoconvert, are %u-bit GStreamer \"base\" plugins installed?\n",
+                    8 * (int)sizeof(void *));
+            return;
         }
 
-        pin->flipfilter = gst_element_factory_make("videoflip", NULL);
-        if(!pin->flipfilter){
-            ERR("Missing videoflip filter?\n");
-            ret = -1;
-            goto exit;
+        /* GStreamer outputs video top-down, but DirectShow expects bottom-up. */
+        if (!(flip = gst_element_factory_make("videoflip", NULL)))
+        {
+            ERR("Failed to create videoflip, are %u-bit GStreamer \"good\" plugins installed?\n",
+                    8 * (int)sizeof(void *));
+            return;
         }
 
-        gst_util_set_object_arg(G_OBJECT(pin->flipfilter), "method", "vertical-flip");
+        gst_util_set_object_arg(G_OBJECT(flip), "method", "vertical-flip");
 
         gst_bin_add(GST_BIN(This->container), vconv); /* bin takes ownership */
         gst_element_sync_state_with_parent(vconv);
-        gst_bin_add(GST_BIN(This->container), pin->flipfilter); /* bin takes ownership */
-        gst_element_sync_state_with_parent(pin->flipfilter);
+        gst_bin_add(GST_BIN(This->container), flip); /* bin takes ownership */
+        gst_element_sync_state_with_parent(flip);
 
-        gst_element_link (vconv, pin->flipfilter);
+        gst_element_link(vconv, flip);
 
-        pin->flip_sink = gst_element_get_static_pad(vconv, "sink");
-        if(!pin->flip_sink){
-            WARN("Couldn't find sink on flip filter\n");
-            pin->flipfilter = NULL;
-            ret = -1;
-            goto exit;
-        }
+        pin->post_sink = gst_element_get_static_pad(vconv, "sink");
+        pin->post_src = gst_element_get_static_pad(flip, "src");
+    }
+    else if (!strcmp(typename, "audio/x-raw"))
+    {
+        GstElement *convert;
 
-        ret = gst_pad_link(pad, pin->flip_sink);
-        if(ret < 0){
-            WARN("gst_pad_link failed: %d\n", ret);
-            gst_object_unref(pin->flip_sink);
-            pin->flip_sink = NULL;
-            pin->flipfilter = NULL;
-            goto exit;
+        /* Currently our dsound can't handle 64-bit formats or all
+         * surround-sound configurations. Native dsound can't always handle
+         * 64-bit formats either. Add an audioconvert to allow changing bit
+         * depth and channel count. */
+        if (!(convert = gst_element_factory_make("audioconvert", NULL)))
+        {
+            ERR("Failed to create audioconvert, are %u-bit GStreamer \"base\" plugins installed?\n",
+                    8 * (int)sizeof(void *));
+            return;
         }
 
-        pin->flip_src = gst_element_get_static_pad(pin->flipfilter, "src");
-        if(!pin->flip_src){
-            WARN("Couldn't find src on flip filter\n");
-            gst_object_unref(pin->flip_sink);
-            pin->flip_sink = NULL;
-            pin->flipfilter = NULL;
-            ret = -1;
-            goto exit;
+        gst_bin_add(GST_BIN(This->container), convert);
+        gst_element_sync_state_with_parent(convert);
+
+        pin->post_sink = gst_element_get_static_pad(convert, "sink");
+        pin->post_src = gst_element_get_static_pad(convert, "src");
+    }
+
+    if (pin->post_sink)
+    {
+        if ((ret = gst_pad_link(pad, pin->post_sink)) < 0)
+        {
+            ERR("Failed to link decodebin source pad to post-processing elements, error %s.\n",
+                    gst_pad_link_get_name(ret));
+            gst_object_unref(pin->post_sink);
+            pin->post_sink = NULL;
+            return;
         }
 
-        ret = gst_pad_link(pin->flip_src, pin->my_sink);
-        if(ret < 0){
-            WARN("gst_pad_link failed: %d\n", ret);
-            gst_object_unref(pin->flip_src);
-            pin->flip_src = NULL;
-            gst_object_unref(pin->flip_sink);
-            pin->flip_sink = NULL;
-            pin->flipfilter = NULL;
-            goto exit;
+        if ((ret = gst_pad_link(pin->post_src, pin->my_sink)) < 0)
+        {
+            ERR("Failed to link post-processing elements to our sink pad, error %s.\n",
+                    gst_pad_link_get_name(ret));
+            gst_object_unref(pin->post_src);
+            pin->post_src = NULL;
+            gst_object_unref(pin->post_sink);
+            pin->post_sink = NULL;
+            return;
         }
-    } else
-        ret = gst_pad_link(pad, pin->my_sink);
+    }
+    else if ((ret = gst_pad_link(pad, pin->my_sink)) < 0)
+    {
+        ERR("Failed to link decodebin source pad to our sink pad, error %s.\n",
+                gst_pad_link_get_name(ret));
+        return;
+    }
 
     gst_pad_set_active(pin->my_sink, 1);
-
-exit:
-    TRACE("Linking: %i\n", ret);
-
-    if (ret >= 0) {
-        pin->their_src = pad;
-        gst_object_ref(pin->their_src);
-    }
+    gst_object_ref(pin->their_src = pad);
 }
 
 static void existing_new_pad(GstElement *bin, GstPad *pad, gpointer user)
@@ -1156,8 +1158,8 @@ static void existing_new_pad(GstElement *bin, GstPad *pad, gpointer user)
         if (!pin->their_src) {
             gst_segment_init(pin->segment, GST_FORMAT_TIME);
 
-            if (pin->flipfilter)
-                ret = gst_pad_link(pad, pin->flip_sink);
+            if (pin->post_sink)
+                ret = gst_pad_link(pad, pin->post_sink);
             else
                 ret = gst_pad_link(pad, pin->my_sink);
 
@@ -2136,14 +2138,13 @@ static void free_source_pin(struct gstdemux_source *pin)
 
     if (pin->their_src)
     {
-        if (pin->flipfilter)
+        if (pin->post_sink)
         {
-            gst_pad_unlink(pin->their_src, pin->flip_sink);
-            gst_pad_unlink(pin->flip_src, pin->my_sink);
-            gst_object_unref(pin->flip_src);
-            gst_object_unref(pin->flip_sink);
-            pin->flipfilter = NULL;
-            pin->flip_src = pin->flip_sink = NULL;
+            gst_pad_unlink(pin->their_src, pin->post_sink);
+            gst_pad_unlink(pin->post_src, pin->my_sink);
+            gst_object_unref(pin->post_src);
+            gst_object_unref(pin->post_sink);
+            pin->post_src = pin->post_sink = NULL;
         }
         else
             gst_pad_unlink(pin->their_src, pin->my_sink);
-- 
2.25.0

[PATCH 4/5] winegstreamer: Allow converting to 16-bit 2-channel PCM within the splitter filter.

[Zebediah Figura]

Signed-off-by: Zebediah Figura <z.figura12(a)gmail.com>
---
 dlls/winegstreamer/gstdemux.c | 18 ++++++++++++++++++
 1 file changed, 18 insertions(+)

diff --git a/dlls/winegstreamer/gstdemux.c b/dlls/winegstreamer/gstdemux.c
index 8e36644030a..f9eed906b6e 100644
--- a/dlls/winegstreamer/gstdemux.c
+++ b/dlls/winegstreamer/gstdemux.c
@@ -1745,6 +1745,24 @@ static HRESULT gstdecoder_source_get_media_type(struct gstdemux_source *pin,
                 * vih->bmiHeader.biHeight * vih->bmiHeader.biBitCount / 8;
         return S_OK;
     }
+    else if (IsEqualGUID(&pin->mt.majortype, &MEDIATYPE_Audio) && index == 1)
+    {
+        const WAVEFORMATEX *our_format = (WAVEFORMATEX *)pin->mt.pbFormat;
+        WAVEFORMATEX *format;
+
+        *mt = pin->mt;
+        mt->subtype = MEDIASUBTYPE_PCM;
+        mt->pbFormat = CoTaskMemAlloc(sizeof(WAVEFORMATEX));
+        format = (WAVEFORMATEX *)mt->pbFormat;
+        format->wFormatTag = WAVE_FORMAT_PCM;
+        format->nChannels = 2;
+        format->nSamplesPerSec = our_format->nSamplesPerSec;
+        format->wBitsPerSample = 16;
+        format->nBlockAlign = 4;
+        format->nAvgBytesPerSec = format->nSamplesPerSec * 4;
+        format->cbSize = 0;
+        return S_OK;
+    }
 
     return VFW_S_NO_MORE_ITEMS;
 }
-- 
2.25.0

[PATCH 5/5] winegstreamer: Get rid of the AudioConvert filter.

[Zebediah Figura]

Signed-off-by: Zebediah Figura <z.figura12(a)gmail.com>
---
 dlls/winegstreamer/gst_guids.h   |   1 -
 dlls/winegstreamer/gst_private.h |   1 -
 dlls/winegstreamer/gsttffilter.c | 120 -------------------------------
 dlls/winegstreamer/main.c        |  35 ---------
 4 files changed, 157 deletions(-)

diff --git a/dlls/winegstreamer/gst_guids.h b/dlls/winegstreamer/gst_guids.h
index fe7b65e4454..e50ecb48519 100644
--- a/dlls/winegstreamer/gst_guids.h
+++ b/dlls/winegstreamer/gst_guids.h
@@ -19,7 +19,6 @@
  * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
  */
 
-DEFINE_GUID(CLSID_Gstreamer_AudioConvert, 0x334b2ec9, 0xf2b5, 0x40b9, 0x84, 0x32, 0x4a, 0x00, 0xe0, 0x03, 0x86, 0xa8);
 DEFINE_GUID(CLSID_Gstreamer_Mp3, 0x728dcf55, 0x128f, 0x4dd1, 0xad, 0x22, 0xbe, 0xcf, 0xa6, 0x6c, 0xe7, 0xaa);
 DEFINE_GUID(CLSID_Gstreamer_Splitter, 0xf9d8d64e, 0xa144, 0x47dc, 0x8e, 0xe0, 0xf5, 0x34, 0x98, 0x37, 0x2c, 0x29);
 DEFINE_GUID(WINESUBTYPE_Gstreamer, 0xffffffff, 0x128f, 0x4dd1, 0xad, 0x22, 0xbe, 0xcf, 0xa6, 0x6c, 0xe7, 0xaa);
diff --git a/dlls/winegstreamer/gst_private.h b/dlls/winegstreamer/gst_private.h
index 37b19ea6265..7108a32e401 100644
--- a/dlls/winegstreamer/gst_private.h
+++ b/dlls/winegstreamer/gst_private.h
@@ -37,7 +37,6 @@
 
 IUnknown * CALLBACK avi_splitter_create(IUnknown *outer, HRESULT *phr) DECLSPEC_HIDDEN;
 IUnknown * CALLBACK mpeg_splitter_create(IUnknown *outer, HRESULT *phr) DECLSPEC_HIDDEN;
-IUnknown * CALLBACK Gstreamer_AudioConvert_create(IUnknown *pUnkOuter, HRESULT *phr);
 IUnknown * CALLBACK Gstreamer_Mp3_create(IUnknown *pUnkOuter, HRESULT *phr);
 IUnknown * CALLBACK Gstreamer_Splitter_create(IUnknown *pUnkOuter, HRESULT *phr);
 IUnknown * CALLBACK wave_parser_create(IUnknown *outer, HRESULT *phr) DECLSPEC_HIDDEN;
diff --git a/dlls/winegstreamer/gsttffilter.c b/dlls/winegstreamer/gsttffilter.c
index 5d1a98b0f5c..21feefe09d2 100644
--- a/dlls/winegstreamer/gsttffilter.c
+++ b/dlls/winegstreamer/gsttffilter.c
@@ -626,123 +626,3 @@ IUnknown * CALLBACK Gstreamer_Mp3_create(IUnknown *punkouter, HRESULT *phr)
 
     return obj;
 }
-
-static HRESULT WINAPI Gstreamer_AudioConvert_QueryConnect(TransformFilter *iface, const AM_MEDIA_TYPE *amt)
-{
-    GstTfImpl *This = (GstTfImpl*)iface;
-    TRACE("%p %p\n", This, amt);
-
-    if (!IsEqualGUID(&amt->majortype, &MEDIATYPE_Audio) ||
-        !IsEqualGUID(&amt->subtype, &MEDIASUBTYPE_PCM) ||
-        !IsEqualGUID(&amt->formattype, &FORMAT_WaveFormatEx))
-        return S_FALSE;
-    return S_OK;
-}
-
-static HRESULT audio_converter_connect_sink(TransformFilter *tf, const AM_MEDIA_TYPE *amt)
-{
-    GstTfImpl *This = (GstTfImpl*)tf;
-    GstCaps *capsin, *capsout;
-    AM_MEDIA_TYPE *outpmt = &This->tf.pmt;
-    WAVEFORMATEX *inwfe;
-    WAVEFORMATEX *outwfe;
-    WAVEFORMATEXTENSIBLE *outwfx;
-    GstAudioFormat format;
-    HRESULT hr;
-    BOOL inisfloat = FALSE;
-    int indepth;
-
-    mark_wine_thread();
-
-    if (Gstreamer_AudioConvert_QueryConnect(&This->tf, amt) == S_FALSE || !amt->pbFormat)
-        return E_FAIL;
-
-    FreeMediaType(outpmt);
-    *outpmt = *amt;
-    outpmt->pUnk = NULL;
-    outpmt->cbFormat = sizeof(WAVEFORMATEXTENSIBLE);
-    outpmt->pbFormat = CoTaskMemAlloc(outpmt->cbFormat);
-
-    inwfe = (WAVEFORMATEX*)amt->pbFormat;
-    indepth = inwfe->wBitsPerSample;
-    if (inwfe->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
-        WAVEFORMATEXTENSIBLE *inwfx = (WAVEFORMATEXTENSIBLE*)inwfe;
-        inisfloat = IsEqualGUID(&inwfx->SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT);
-        if (inwfx->Samples.wValidBitsPerSample)
-            indepth = inwfx->Samples.wValidBitsPerSample;
-    } else if (inwfe->wFormatTag == WAVE_FORMAT_IEEE_FLOAT)
-        inisfloat = TRUE;
-
-    if (inisfloat)
-        format = inwfe->wBitsPerSample == 64 ? GST_AUDIO_FORMAT_F64LE : GST_AUDIO_FORMAT_F32LE;
-    else
-        format = gst_audio_format_build_integer(inwfe->wBitsPerSample != 8, G_LITTLE_ENDIAN,
-                inwfe->wBitsPerSample, indepth);
-
-    capsin = gst_caps_new_simple("audio/x-raw",
-                                 "format", G_TYPE_STRING, gst_audio_format_to_string(format),
-                                 "channels", G_TYPE_INT, inwfe->nChannels,
-                                 "rate", G_TYPE_INT, inwfe->nSamplesPerSec,
-                                  NULL);
-
-    outwfe = (WAVEFORMATEX*)outpmt->pbFormat;
-    outwfx = (WAVEFORMATEXTENSIBLE*)outwfe;
-    outwfe->wFormatTag = WAVE_FORMAT_EXTENSIBLE;
-    outwfe->nChannels = 2;
-    outwfe->nSamplesPerSec = inwfe->nSamplesPerSec;
-    outwfe->wBitsPerSample = 16;
-    outwfe->nBlockAlign = outwfe->nChannels * outwfe->wBitsPerSample / 8;
-    outwfe->nAvgBytesPerSec = outwfe->nBlockAlign * outwfe->nSamplesPerSec;
-    outwfe->cbSize = sizeof(*outwfx) - sizeof(*outwfe);
-    outwfx->Samples.wValidBitsPerSample = outwfe->wBitsPerSample;
-    outwfx->dwChannelMask = SPEAKER_FRONT_LEFT|SPEAKER_FRONT_RIGHT;
-    outwfx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
-
-    capsout = gst_caps_new_simple("audio/x-raw",
-                                  "format", G_TYPE_STRING, "S16LE",
-                                  "channels", G_TYPE_INT, outwfe->nChannels,
-                                  "rate", G_TYPE_INT, outwfe->nSamplesPerSec,
-                                   NULL);
-
-
-    hr = Gstreamer_transform_ConnectInput(This, amt, capsin, capsout);
-    gst_caps_unref(capsin);
-    gst_caps_unref(capsout);
-
-    This->cbBuffer = inwfe->nAvgBytesPerSec;
-    return hr;
-}
-
-static const TransformFilterFuncTable Gstreamer_AudioConvert_vtbl = {
-    .pfnDecideBufferSize = Gstreamer_transform_DecideBufferSize,
-    .pfnStartStreaming = Gstreamer_transform_ProcessBegin,
-    .pfnReceive = Gstreamer_transform_ProcessData,
-    .pfnStopStreaming = Gstreamer_transform_ProcessEnd,
-    .pfnCheckInputType = Gstreamer_AudioConvert_QueryConnect,
-    .transform_connect_sink = audio_converter_connect_sink,
-    .pfnBreakConnect = Gstreamer_transform_Cleanup,
-    .pfnEndOfStream = Gstreamer_transform_EndOfStream,
-    .pfnBeginFlush = Gstreamer_transform_BeginFlush,
-    .pfnEndFlush = Gstreamer_transform_EndFlush,
-    .pfnNewSegment = Gstreamer_transform_NewSegment,
-    .pfnNotify = Gstreamer_transform_QOS,
-};
-
-IUnknown * CALLBACK Gstreamer_AudioConvert_create(IUnknown *punkouter, HRESULT *phr)
-{
-    IUnknown *obj = NULL;
-
-    TRACE("%p %p\n", punkouter, phr);
-
-    if (!init_gstreamer())
-    {
-        *phr = E_FAIL;
-        return NULL;
-    }
-
-    *phr = Gstreamer_transform_create(punkouter, &CLSID_Gstreamer_AudioConvert, "audioconvert", &Gstreamer_AudioConvert_vtbl, (LPVOID*)&obj);
-
-    TRACE("returning %p\n", obj);
-
-    return obj;
-}
diff --git a/dlls/winegstreamer/main.c b/dlls/winegstreamer/main.c
index cdc3ba0ddce..3e0ef28afb6 100644
--- a/dlls/winegstreamer/main.c
+++ b/dlls/winegstreamer/main.c
@@ -40,8 +40,6 @@ static const WCHAR wGstreamer_Splitter[] =
 {'G','S','t','r','e','a','m','e','r',' ','s','p','l','i','t','t','e','r',' ','f','i','l','t','e','r',0};
 static const WCHAR wGstreamer_Mp3[] =
 {'G','S','t','r','e','a','m','e','r',' ','M','p','3',' ','f','i','l','t','e','r',0};
-static const WCHAR wGstreamer_AudioConvert[] =
-{'G','S','t','r','e','a','m','e','r',' ','A','u','d','i','o','C','o','n','v','e','r','t',' ','f','i','l','t','e','r',0};
 static const WCHAR wave_parserW[] =
 {'W','a','v','e',' ','P','a','r','s','e','r',0};
 static const WCHAR avi_splitterW[] =
@@ -122,32 +120,6 @@ AMOVIESETUP_FILTER const amfMp3 =
     amfMp3Pin
 };
 
-AMOVIESETUP_PIN amfAudioConvertPin[] =
-{   {   wNull,
-        FALSE, FALSE, FALSE, FALSE,
-        &GUID_NULL,
-        NULL,
-        1,
-        amfMTaudio
-    },
-    {
-        wNull,
-        FALSE, TRUE, FALSE, FALSE,
-        &GUID_NULL,
-        NULL,
-        1,
-        amfMTaudio
-    },
-};
-
-AMOVIESETUP_FILTER const amfAudioConvert =
-{   &CLSID_Gstreamer_AudioConvert,
-    wGstreamer_AudioConvert,
-    MERIT_UNLIKELY,
-    2,
-    amfAudioConvertPin
-};
-
 static const AMOVIESETUP_MEDIATYPE wave_parser_sink_type_data[] =
 {
     {&MEDIATYPE_Stream, &MEDIASUBTYPE_WAVE},
@@ -295,13 +267,6 @@ FactoryTemplate const g_Templates[] = {
         NULL,
         &amfMp3,
     },
-    {
-        wGstreamer_AudioConvert,
-        &CLSID_Gstreamer_AudioConvert,
-        Gstreamer_AudioConvert_create,
-        NULL,
-        &amfAudioConvert,
-    },
     {
         wave_parserW,
         &CLSID_WAVEParser,
-- 
2.25.0