winegstreamer: Get rid of the AudioConvert filter.
Signed-off-by: Zebediah Figura <z.figura12@gmail.com> Signed-off-by: Alexandre Julliard <julliard@winehq.org>
This commit is contained in:
Zebediah Figura
Alexandre Julliard
parent
1052d5cfb1
commit
943b6564f5
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@ -19,7 +19,6 @@
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* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
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*/
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DEFINE_GUID(CLSID_Gstreamer_AudioConvert, 0x334b2ec9, 0xf2b5, 0x40b9, 0x84, 0x32, 0x4a, 0x00, 0xe0, 0x03, 0x86, 0xa8);
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DEFINE_GUID(CLSID_Gstreamer_Mp3, 0x728dcf55, 0x128f, 0x4dd1, 0xad, 0x22, 0xbe, 0xcf, 0xa6, 0x6c, 0xe7, 0xaa);
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DEFINE_GUID(CLSID_Gstreamer_Splitter, 0xf9d8d64e, 0xa144, 0x47dc, 0x8e, 0xe0, 0xf5, 0x34, 0x98, 0x37, 0x2c, 0x29);
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DEFINE_GUID(WINESUBTYPE_Gstreamer, 0xffffffff, 0x128f, 0x4dd1, 0xad, 0x22, 0xbe, 0xcf, 0xa6, 0x6c, 0xe7, 0xaa);
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@ -37,7 +37,6 @@
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IUnknown * CALLBACK avi_splitter_create(IUnknown *outer, HRESULT *phr) DECLSPEC_HIDDEN;
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IUnknown * CALLBACK mpeg_splitter_create(IUnknown *outer, HRESULT *phr) DECLSPEC_HIDDEN;
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IUnknown * CALLBACK Gstreamer_AudioConvert_create(IUnknown *pUnkOuter, HRESULT *phr);
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IUnknown * CALLBACK Gstreamer_Mp3_create(IUnknown *pUnkOuter, HRESULT *phr);
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IUnknown * CALLBACK Gstreamer_Splitter_create(IUnknown *pUnkOuter, HRESULT *phr);
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IUnknown * CALLBACK wave_parser_create(IUnknown *outer, HRESULT *phr) DECLSPEC_HIDDEN;
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@ -626,123 +626,3 @@ IUnknown * CALLBACK Gstreamer_Mp3_create(IUnknown *punkouter, HRESULT *phr)
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return obj;
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}
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static HRESULT WINAPI Gstreamer_AudioConvert_QueryConnect(TransformFilter *iface, const AM_MEDIA_TYPE *amt)
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{
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GstTfImpl *This = (GstTfImpl*)iface;
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TRACE("%p %p\n", This, amt);
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if (!IsEqualGUID(&amt->majortype, &MEDIATYPE_Audio) ||
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!IsEqualGUID(&amt->subtype, &MEDIASUBTYPE_PCM) ||
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!IsEqualGUID(&amt->formattype, &FORMAT_WaveFormatEx))
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return S_FALSE;
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return S_OK;
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}
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static HRESULT audio_converter_connect_sink(TransformFilter *tf, const AM_MEDIA_TYPE *amt)
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{
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GstTfImpl *This = (GstTfImpl*)tf;
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GstCaps *capsin, *capsout;
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AM_MEDIA_TYPE *outpmt = &This->tf.pmt;
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WAVEFORMATEX *inwfe;
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WAVEFORMATEX *outwfe;
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WAVEFORMATEXTENSIBLE *outwfx;
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GstAudioFormat format;
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HRESULT hr;
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BOOL inisfloat = FALSE;
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int indepth;
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mark_wine_thread();
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if (Gstreamer_AudioConvert_QueryConnect(&This->tf, amt) == S_FALSE || !amt->pbFormat)
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return E_FAIL;
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FreeMediaType(outpmt);
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*outpmt = *amt;
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outpmt->pUnk = NULL;
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outpmt->cbFormat = sizeof(WAVEFORMATEXTENSIBLE);
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outpmt->pbFormat = CoTaskMemAlloc(outpmt->cbFormat);
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inwfe = (WAVEFORMATEX*)amt->pbFormat;
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indepth = inwfe->wBitsPerSample;
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if (inwfe->wFormatTag == WAVE_FORMAT_EXTENSIBLE) {
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WAVEFORMATEXTENSIBLE *inwfx = (WAVEFORMATEXTENSIBLE*)inwfe;
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inisfloat = IsEqualGUID(&inwfx->SubFormat, &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT);
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if (inwfx->Samples.wValidBitsPerSample)
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indepth = inwfx->Samples.wValidBitsPerSample;
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} else if (inwfe->wFormatTag == WAVE_FORMAT_IEEE_FLOAT)
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inisfloat = TRUE;
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if (inisfloat)
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format = inwfe->wBitsPerSample == 64 ? GST_AUDIO_FORMAT_F64LE : GST_AUDIO_FORMAT_F32LE;
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else
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format = gst_audio_format_build_integer(inwfe->wBitsPerSample != 8, G_LITTLE_ENDIAN,
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inwfe->wBitsPerSample, indepth);
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capsin = gst_caps_new_simple("audio/x-raw",
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"format", G_TYPE_STRING, gst_audio_format_to_string(format),
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"channels", G_TYPE_INT, inwfe->nChannels,
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"rate", G_TYPE_INT, inwfe->nSamplesPerSec,
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NULL);
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outwfe = (WAVEFORMATEX*)outpmt->pbFormat;
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outwfx = (WAVEFORMATEXTENSIBLE*)outwfe;
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outwfe->wFormatTag = WAVE_FORMAT_EXTENSIBLE;
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outwfe->nChannels = 2;
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outwfe->nSamplesPerSec = inwfe->nSamplesPerSec;
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outwfe->wBitsPerSample = 16;
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outwfe->nBlockAlign = outwfe->nChannels * outwfe->wBitsPerSample / 8;
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outwfe->nAvgBytesPerSec = outwfe->nBlockAlign * outwfe->nSamplesPerSec;
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outwfe->cbSize = sizeof(*outwfx) - sizeof(*outwfe);
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outwfx->Samples.wValidBitsPerSample = outwfe->wBitsPerSample;
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outwfx->dwChannelMask = SPEAKER_FRONT_LEFT|SPEAKER_FRONT_RIGHT;
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outwfx->SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
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capsout = gst_caps_new_simple("audio/x-raw",
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"format", G_TYPE_STRING, "S16LE",
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"channels", G_TYPE_INT, outwfe->nChannels,
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"rate", G_TYPE_INT, outwfe->nSamplesPerSec,
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NULL);
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hr = Gstreamer_transform_ConnectInput(This, amt, capsin, capsout);
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gst_caps_unref(capsin);
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gst_caps_unref(capsout);
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This->cbBuffer = inwfe->nAvgBytesPerSec;
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return hr;
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}
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static const TransformFilterFuncTable Gstreamer_AudioConvert_vtbl = {
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.pfnDecideBufferSize = Gstreamer_transform_DecideBufferSize,
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.pfnStartStreaming = Gstreamer_transform_ProcessBegin,
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.pfnReceive = Gstreamer_transform_ProcessData,
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.pfnStopStreaming = Gstreamer_transform_ProcessEnd,
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.pfnCheckInputType = Gstreamer_AudioConvert_QueryConnect,
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.transform_connect_sink = audio_converter_connect_sink,
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.pfnBreakConnect = Gstreamer_transform_Cleanup,
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.pfnEndOfStream = Gstreamer_transform_EndOfStream,
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.pfnBeginFlush = Gstreamer_transform_BeginFlush,
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.pfnEndFlush = Gstreamer_transform_EndFlush,
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.pfnNewSegment = Gstreamer_transform_NewSegment,
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.pfnNotify = Gstreamer_transform_QOS,
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};
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IUnknown * CALLBACK Gstreamer_AudioConvert_create(IUnknown *punkouter, HRESULT *phr)
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{
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IUnknown *obj = NULL;
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TRACE("%p %p\n", punkouter, phr);
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if (!init_gstreamer())
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{
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*phr = E_FAIL;
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return NULL;
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}
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*phr = Gstreamer_transform_create(punkouter, &CLSID_Gstreamer_AudioConvert, "audioconvert", &Gstreamer_AudioConvert_vtbl, (LPVOID*)&obj);
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TRACE("returning %p\n", obj);
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return obj;
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}
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@ -40,8 +40,6 @@ static const WCHAR wGstreamer_Splitter[] =
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{'G','S','t','r','e','a','m','e','r',' ','s','p','l','i','t','t','e','r',' ','f','i','l','t','e','r',0};
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static const WCHAR wGstreamer_Mp3[] =
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{'G','S','t','r','e','a','m','e','r',' ','M','p','3',' ','f','i','l','t','e','r',0};
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static const WCHAR wGstreamer_AudioConvert[] =
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{'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};
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static const WCHAR wave_parserW[] =
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{'W','a','v','e',' ','P','a','r','s','e','r',0};
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static const WCHAR avi_splitterW[] =
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@ -122,32 +120,6 @@ AMOVIESETUP_FILTER const amfMp3 =
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amfMp3Pin
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};
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AMOVIESETUP_PIN amfAudioConvertPin[] =
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{ { wNull,
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FALSE, FALSE, FALSE, FALSE,
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&GUID_NULL,
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NULL,
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1,
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amfMTaudio
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},
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{
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wNull,
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FALSE, TRUE, FALSE, FALSE,
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&GUID_NULL,
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NULL,
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1,
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amfMTaudio
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},
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};
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AMOVIESETUP_FILTER const amfAudioConvert =
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{ &CLSID_Gstreamer_AudioConvert,
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wGstreamer_AudioConvert,
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MERIT_UNLIKELY,
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2,
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amfAudioConvertPin
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};
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static const AMOVIESETUP_MEDIATYPE wave_parser_sink_type_data[] =
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{
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{&MEDIATYPE_Stream, &MEDIASUBTYPE_WAVE},
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@ -295,13 +267,6 @@ FactoryTemplate const g_Templates[] = {
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NULL,
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&amfMp3,
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},
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{
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wGstreamer_AudioConvert,
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&CLSID_Gstreamer_AudioConvert,
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Gstreamer_AudioConvert_create,
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NULL,
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&amfAudioConvert,
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},
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{
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wave_parserW,
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&CLSID_WAVEParser,
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