Aric Stewart : winegstreamer: Add gstreamer audio convert transform filter.

[Alexandre Julliard]

Module: wine
Branch: master
Commit: 0e55930effbfb5b6e79a77b5da4f4e0e5a52455e
URL:    http://source.winehq.org/git/wine.git/?a=commit;h=0e55930effbfb5b6e79a77b5da...

Author: Aric Stewart <aric(a)codeweavers.com>
Date:   Thu Oct 28 14:42:33 2010 -0500

winegstreamer: Add gstreamer audio convert transform filter.

---

 dlls/winegstreamer/gst_guids.h   |    1 +
 dlls/winegstreamer/gst_private.h |    1 +
 dlls/winegstreamer/gsttffilter.c |  113 ++++++++++++++++++++++++++++++++++++++
 dlls/winegstreamer/main.c        |   35 ++++++++++++
 4 files changed, 150 insertions(+), 0 deletions(-)

diff --git a/dlls/winegstreamer/gst_guids.h b/dlls/winegstreamer/gst_guids.h
index 092c4f6..089cd9d 100644
--- a/dlls/winegstreamer/gst_guids.h
+++ b/dlls/winegstreamer/gst_guids.h
@@ -22,6 +22,7 @@
  * 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 a838179..741fdf4 100644
--- a/dlls/winegstreamer/gst_private.h
+++ b/dlls/winegstreamer/gst_private.h
@@ -40,6 +40,7 @@
 /* enum media */
 void dump_AM_MEDIA_TYPE(const AM_MEDIA_TYPE * pmt);
 
+IUnknown * CALLBACK Gstreamer_AudioConvert_create(IUnknown *pUnkOuter, HRESULT *phr);
 IUnknown * CALLBACK Gstreamer_Mp3_create(IUnknown *pUnkOuter, HRESULT *phr);
 IUnknown * CALLBACK Gstreamer_YUV_create(IUnknown *pUnkOuter, HRESULT *phr);
 IUnknown * CALLBACK Gstreamer_Splitter_create(IUnknown *pUnkOuter, HRESULT *phr);
diff --git a/dlls/winegstreamer/gsttffilter.c b/dlls/winegstreamer/gsttffilter.c
index 6237fac..9339299 100644
--- a/dlls/winegstreamer/gsttffilter.c
+++ b/dlls/winegstreamer/gsttffilter.c
@@ -690,6 +690,119 @@ IUnknown * CALLBACK Gstreamer_YUV_create(IUnknown *punkout, 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);
+    dump_AM_MEDIA_TYPE(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 WINAPI Gstreamer_AudioConvert_ConnectInput(TransformFilter *tf, PIN_DIRECTION dir, IPin *pin)
+{
+    return S_OK;
+}
+
+static HRESULT WINAPI Gstreamer_AudioConvert_SetMediaType(TransformFilter *tf, PIN_DIRECTION dir, 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;
+    HRESULT hr;
+    int inisfloat = 0, indepth;
+
+    if (dir != PINDIR_INPUT)
+        return S_OK;
+
+    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;
+    }
+
+    capsin = gst_caps_new_simple(inisfloat ? "audio/x-raw-float" : "audio/x-raw-int",
+                                 "endianness", G_TYPE_INT, 1234,
+                                 "width", G_TYPE_INT, inwfe->wBitsPerSample,
+                                 "depth", G_TYPE_INT, indepth,
+                                 "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-int",
+                                  "endianness", G_TYPE_INT, 1234,
+                                  "width", G_TYPE_INT, outwfe->wBitsPerSample,
+                                  "depth", G_TYPE_INT, outwfx->Samples.wValidBitsPerSample,
+                                  "channels", G_TYPE_INT, outwfe->nChannels,
+                                  "rate", G_TYPE_INT, outwfe->nSamplesPerSec,
+                                   NULL);
+
+    hr = Gstreamer_transform_ConnectInput(This, amt, capsin, capsout);
+    FIXME("%08x\n", hr);
+    gst_caps_unref(capsin);
+    gst_caps_unref(capsout);
+
+    This->cbBuffer = inwfe->nAvgBytesPerSec;
+    return hr;
+}
+
+static const TransformFilterFuncTable Gstreamer_AudioConvert_vtbl = {
+    Gstreamer_transform_DecideBufferSize,
+    Gstreamer_transform_ProcessBegin,
+    Gstreamer_transform_ProcessData,
+    Gstreamer_transform_ProcessEnd,
+    Gstreamer_AudioConvert_QueryConnect,
+    Gstreamer_AudioConvert_SetMediaType,
+    Gstreamer_AudioConvert_ConnectInput,
+    Gstreamer_transform_Cleanup,
+    Gstreamer_transform_EndOfStream,
+    Gstreamer_transform_BeginFlush,
+    Gstreamer_transform_EndFlush,
+    Gstreamer_transform_NewSegment
+};
+
+IUnknown * CALLBACK Gstreamer_AudioConvert_create(IUnknown *punkout, HRESULT *phr)
+{
+    IUnknown *obj = NULL;
+    if (!Gstreamer_init())
+    {
+        *phr = E_FAIL;
+        return NULL;
+    }
+    *phr = Gstreamer_transform_create(punkout, &CLSID_Gstreamer_AudioConvert, "audioconvert", &Gstreamer_AudioConvert_vtbl, (LPVOID*)&obj);
+    return obj;
+}
+
 HRESULT WINAPI GSTTf_QueryInterface(IBaseFilter * iface, REFIID riid, LPVOID * ppv)
 {
     HRESULT hr;
diff --git a/dlls/winegstreamer/main.c b/dlls/winegstreamer/main.c
index 6007ac7..ac9e97b 100644
--- a/dlls/winegstreamer/main.c
+++ b/dlls/winegstreamer/main.c
@@ -49,6 +49,8 @@ static const WCHAR wGstreamer_YUV[] =
 {'G','S','t','r','e','a','m','e','r',' ','Y','U','V',' ','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 WCHAR wNull[] = {'\0'};
 
@@ -149,6 +151,32 @@ 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
+};
+
 FactoryTemplate const g_Templates[] = {
     {
         wGstreamer_Splitter,
@@ -171,6 +199,13 @@ FactoryTemplate const g_Templates[] = {
         NULL,
         &amfMp3,
     },
+    {
+        wGstreamer_AudioConvert,
+        &CLSID_Gstreamer_AudioConvert,
+        Gstreamer_AudioConvert_create,
+        NULL,
+        &amfAudioConvert,
+    },
 };
 
 const int g_cTemplates = sizeof(g_Templates) / sizeof (g_Templates[0]);