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wasapi: Initial port to 1.0

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This should really use GstAudioSink and GstAudioSrc.
This commit is contained in:
Sebastian Dröge 2013-03-26 15:22:16 +01:00
parent d5d37fafa7
commit e7a69bb8de
9 changed files with 83 additions and 93 deletions
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2
configure.ac
View file

@ -328,7 +328,7 @@ GST_PLUGINS_NONPORTED=" aiff \
gsettings ladspa \ gsettings ladspa \
musepack musicbrainz nas neon ofa openal rsvg sdl sndfile timidity \ musepack musicbrainz nas neon ofa openal rsvg sdl sndfile timidity \
directdraw direct3d9 acm wininet \ directdraw direct3d9 acm wininet \
xvid lv2 teletextdec sndio wasapi" xvid lv2 teletextdec sndio"
AC_SUBST(GST_PLUGINS_NONPORTED) AC_SUBST(GST_PLUGINS_NONPORTED)
dnl these are all the gst plug-ins, compilable without additional libs dnl these are all the gst plug-ins, compilable without additional libs

2
sys/wasapi/Makefile.am
View file

@ -7,7 +7,7 @@ libgstwasapi_la_SOURCES = gstwasapi.c \
libgstwasapi_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) -DCOBJMACROS=1 libgstwasapi_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) -DCOBJMACROS=1
libgstwasapi_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) \ libgstwasapi_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) \
-lgstaudio-$(GST_MAJORMINOR) -lgstinterfaces-$(GST_MAJORMINOR) \ -lgstaudio-$(GST_API_VERSION) \
$(WASAPI_LIBS) $(WASAPI_LIBS)
libgstwasapi_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) libgstwasapi_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstwasapi_la_LIBTOOLFLAGS = --tag=disable-static libgstwasapi_la_LIBTOOLFLAGS = --tag=disable-static

2
sys/wasapi/gstwasapi.c
View file

@ -42,4 +42,4 @@ GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR, GST_VERSION_MINOR,
wasapi, wasapi,
"Windows audio session API plugin", "Windows audio session API plugin",
plugin_init, VERSION, "LGPL", "GStreamer", "http://gstreamer.net/") plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)

67
sys/wasapi/gstwasapisink.c
View file

@ -26,7 +26,7 @@
* <refsect2> * <refsect2>
* <title>Example pipelines</title> * <title>Example pipelines</title>
* |[ * |[
* gst-launch-0.10 -v audiotestsrc samplesperbuffer=160 ! wasapisink * gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapisink
* ]| Generate 20 ms buffers and render to the default audio device. * ]| Generate 20 ms buffers and render to the default audio device.
* </refsect2> * </refsect2>
*/ */
@ -42,13 +42,10 @@ GST_DEBUG_CATEGORY_STATIC (gst_wasapi_sink_debug);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_SINK,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, " GST_STATIC_CAPS ("audio/x-raw, "
"width = (int) 16, " "format = (string) S16LE, "
"depth = (int) 16, " "layout = (string) interleaved, "
"rate = (int) 8000, " "rate = (int) 8000, " "channels = (int) 1"));
"channels = (int) 1, "
"signed = (boolean) TRUE, "
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER)));
static void gst_wasapi_sink_dispose (GObject * object); static void gst_wasapi_sink_dispose (GObject * object);
static void gst_wasapi_sink_finalize (GObject * object); static void gst_wasapi_sink_finalize (GObject * object);
@ -60,31 +57,25 @@ static gboolean gst_wasapi_sink_stop (GstBaseSink * sink);
static GstFlowReturn gst_wasapi_sink_render (GstBaseSink * sink, static GstFlowReturn gst_wasapi_sink_render (GstBaseSink * sink,
GstBuffer * buffer); GstBuffer * buffer);
GST_BOILERPLATE (GstWasapiSink, gst_wasapi_sink, GstBaseSink, G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_BASE_SINK);
GST_TYPE_BASE_SINK);
static void
gst_wasapi_sink_base_init (gpointer gclass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_static_metadata (element_class, "WasapiSrc",
"Sink/Audio",
"Stream audio to an audio capture device through WASAPI",
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
}
static void static void
gst_wasapi_sink_class_init (GstWasapiSinkClass * klass) gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
{ {
GObjectClass *gobject_class = G_OBJECT_CLASS (klass); GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass); GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
gobject_class->dispose = gst_wasapi_sink_dispose; gobject_class->dispose = gst_wasapi_sink_dispose;
gobject_class->finalize = gst_wasapi_sink_finalize; gobject_class->finalize = gst_wasapi_sink_finalize;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
"Sink/Audio",
"Stream audio to an audio capture device through WASAPI",
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
gstbasesink_class->get_times = gst_wasapi_sink_get_times; gstbasesink_class->get_times = gst_wasapi_sink_get_times;
gstbasesink_class->start = gst_wasapi_sink_start; gstbasesink_class->start = gst_wasapi_sink_start;
gstbasesink_class->stop = gst_wasapi_sink_stop; gstbasesink_class->stop = gst_wasapi_sink_stop;
@ -95,7 +86,7 @@ gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
} }
static void static void
gst_wasapi_sink_init (GstWasapiSink * self, GstWasapiSinkClass * gclass) gst_wasapi_sink_init (GstWasapiSink * self)
{ {
self->rate = 8000; self->rate = 8000;
self->buffer_time = 20 * GST_MSECOND; self->buffer_time = 20 * GST_MSECOND;
@ -117,17 +108,15 @@ gst_wasapi_sink_dispose (GObject * object)
self->event_handle = NULL; self->event_handle = NULL;
} }
G_OBJECT_CLASS (parent_class)->dispose (object); G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->dispose (object);
} }
static void static void
gst_wasapi_sink_finalize (GObject * object) gst_wasapi_sink_finalize (GObject * object)
{ {
GstWasapiSink *self = GST_WASAPI_SINK (object);
CoUninitialize (); CoUninitialize ();
G_OBJECT_CLASS (parent_class)->finalize (object); G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->finalize (object);
} }
static void static void
@ -171,7 +160,7 @@ gst_wasapi_sink_start (GstBaseSink * sink)
} }
hr = IAudioClient_GetService (client, &IID_IAudioRenderClient, hr = IAudioClient_GetService (client, &IID_IAudioRenderClient,
&render_client); (void **) &render_client);
if (hr != S_OK) { if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService " GST_ERROR_OBJECT (self, "IAudioClient::GetService "
"(IID_IAudioRenderClient) failed"); "(IID_IAudioRenderClient) failed");
@ -229,11 +218,23 @@ gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
GstWasapiSink *self = GST_WASAPI_SINK (sink); GstWasapiSink *self = GST_WASAPI_SINK (sink);
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
HRESULT hr; HRESULT hr;
gint16 *src = (gint16 *) GST_BUFFER_DATA (buffer); GstMapInfo minfo;
const gint16 *src;
gint16 *dst = NULL; gint16 *dst = NULL;
guint nsamples = GST_BUFFER_SIZE (buffer) / sizeof (gint16); guint nsamples;
guint i; guint i;
memset (&minfo, 0, sizeof (minfo));
if (!gst_buffer_map (buffer, &minfo, GST_MAP_READ)) {
GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL),
("Failed to map input buffer"));
ret = GST_FLOW_ERROR;
goto beach;
}
nsamples = minfo.size / sizeof (gint16);
WaitForSingleObject (self->event_handle, INFINITE); WaitForSingleObject (self->event_handle, INFINITE);
hr = IAudioRenderClient_GetBuffer (self->render_client, nsamples, hr = IAudioRenderClient_GetBuffer (self->render_client, nsamples,
@ -246,6 +247,7 @@ gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
goto beach; goto beach;
} }
src = (const gint16 *) minfo.data;
for (i = 0; i < nsamples; i++) { for (i = 0; i < nsamples; i++) {
dst[0] = *src; dst[0] = *src;
dst[1] = *src; dst[1] = *src;
@ -263,5 +265,8 @@ gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
} }
beach: beach:
if (minfo.data)
gst_buffer_unmap (buffer, &minfo);
return ret; return ret;
} }

2
sys/wasapi/gstwasapisink.h
View file

@ -22,8 +22,6 @@
#include "gstwasapiutil.h" #include "gstwasapiutil.h"
#include <gst/base/gstbasesink.h>
G_BEGIN_DECLS G_BEGIN_DECLS
#define GST_TYPE_WASAPI_SINK \ #define GST_TYPE_WASAPI_SINK \

89
sys/wasapi/gstwasapisrc.c
View file

@ -26,7 +26,7 @@
* <refsect2> * <refsect2>
* <title>Example pipelines</title> * <title>Example pipelines</title>
* |[ * |[
* gst-launch-0.10 -v wasapisrc ! fakesink * gst-launch-1.0 -v wasapisrc ! fakesink
* ]| Capture from the default audio device and render to fakesink. * ]| Capture from the default audio device and render to fakesink.
* </refsect2> * </refsect2>
*/ */
@ -43,13 +43,10 @@ GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_SRC,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, " GST_STATIC_CAPS ("audio/x-raw, "
"width = (int) 16, " "format = (string) S16LE, "
"depth = (int) 16, " "layout = (string) interleaved, "
"rate = (int) 8000, " "rate = (int) 8000, " "channels = (int) 1"));
"channels = (int) 1, "
"signed = (boolean) TRUE, "
"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER)));
static void gst_wasapi_src_dispose (GObject * object); static void gst_wasapi_src_dispose (GObject * object);
static void gst_wasapi_src_finalize (GObject * object); static void gst_wasapi_src_finalize (GObject * object);
@ -65,20 +62,7 @@ static GstFlowReturn gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf);
static GstClockTime gst_wasapi_src_get_time (GstClock * clock, static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
gpointer user_data); gpointer user_data);
GST_BOILERPLATE (GstWasapiSrc, gst_wasapi_src, GstPushSrc, GST_TYPE_PUSH_SRC); G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_PUSH_SRC);
static void
gst_wasapi_src_base_init (gpointer gclass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_static_metadata (element_class, "WasapiSrc",
"Source/Audio",
"Stream audio from an audio capture device through WASAPI",
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
}
static void static void
gst_wasapi_src_class_init (GstWasapiSrcClass * klass) gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
@ -93,6 +77,13 @@ gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
gstelement_class->provide_clock = gst_wasapi_src_provide_clock; gstelement_class->provide_clock = gst_wasapi_src_provide_clock;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
"Source/Audio",
"Stream audio from an audio capture device through WASAPI",
"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
gstbasesrc_class->start = gst_wasapi_src_start; gstbasesrc_class->start = gst_wasapi_src_start;
gstbasesrc_class->stop = gst_wasapi_src_stop; gstbasesrc_class->stop = gst_wasapi_src_stop;
gstbasesrc_class->query = gst_wasapi_src_query; gstbasesrc_class->query = gst_wasapi_src_query;
@ -104,7 +95,7 @@ gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
} }
static void static void
gst_wasapi_src_init (GstWasapiSrc * self, GstWasapiSrcClass * gclass) gst_wasapi_src_init (GstWasapiSrc * self)
{ {
GstBaseSrc *basesrc = GST_BASE_SRC (self); GstBaseSrc *basesrc = GST_BASE_SRC (self);
@ -120,14 +111,9 @@ gst_wasapi_src_init (GstWasapiSrc * self, GstWasapiSrcClass * gclass)
self->start_time = GST_CLOCK_TIME_NONE; self->start_time = GST_CLOCK_TIME_NONE;
self->next_time = GST_CLOCK_TIME_NONE; self->next_time = GST_CLOCK_TIME_NONE;
#if GST_CHECK_VERSION(0, 10, 31) || (GST_CHECK_VERSION(0, 10, 30) && GST_VERSION_NANO > 0) self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
self->clock = gst_audio_clock_new_full ("GstWasapiSrcClock",
gst_wasapi_src_get_time, gst_object_ref (self), gst_wasapi_src_get_time, gst_object_ref (self),
(GDestroyNotify) gst_object_unref); (GDestroyNotify) gst_object_unref);
#else
self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
gst_wasapi_src_get_time, self);
#endif
CoInitialize (NULL); CoInitialize (NULL);
} }
@ -142,17 +128,15 @@ gst_wasapi_src_dispose (GObject * object)
self->clock = NULL; self->clock = NULL;
} }
G_OBJECT_CLASS (parent_class)->dispose (object); G_OBJECT_CLASS (gst_wasapi_src_parent_class)->dispose (object);
} }
static void static void
gst_wasapi_src_finalize (GObject * object) gst_wasapi_src_finalize (GObject * object)
{ {
GstWasapiSrc *self = GST_WASAPI_SRC (object);
CoUninitialize (); CoUninitialize ();
G_OBJECT_CLASS (parent_class)->finalize (object); G_OBJECT_CLASS (gst_wasapi_src_parent_class)->finalize (object);
} }
static GstClock * static GstClock *
@ -196,7 +180,7 @@ gst_wasapi_src_start (GstBaseSrc * src)
&self->latency)) &self->latency))
goto beach; goto beach;
hr = IAudioClient_GetService (client, &IID_IAudioClock, &client_clock); hr = IAudioClient_GetService (client, &IID_IAudioClock, (void**) &client_clock);
if (hr != S_OK) { if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) " GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
"failed"); "failed");
@ -210,7 +194,7 @@ gst_wasapi_src_start (GstBaseSrc * src)
} }
hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient, hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
&capture_client); (void**) &capture_client);
if (hr != S_OK) { if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService " GST_ERROR_OBJECT (self, "IAudioClient::GetService "
"(IID_IAudioCaptureClient) failed"); "(IID_IAudioCaptureClient) failed");
@ -298,7 +282,8 @@ gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query)
} }
default: default:
ret = GST_BASE_SRC_CLASS (parent_class)->query (src, query); ret =
GST_BASE_SRC_CLASS (gst_wasapi_src_parent_class)->query (src, query);
break; break;
} }
@ -317,6 +302,9 @@ gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
guint32 nsamples_read = 0, nsamples; guint32 nsamples_read = 0, nsamples;
DWORD flags = 0; DWORD flags = 0;
guint64 devpos; guint64 devpos;
guint i;
GstMapInfo minfo;
gint16 *dst;
GST_OBJECT_LOCK (self); GST_OBJECT_LOCK (self);
clock = GST_ELEMENT_CLOCK (self); clock = GST_ELEMENT_CLOCK (self);
@ -347,7 +335,7 @@ gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
if (flags != 0) { if (flags != 0) {
GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x", GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x",
devpos, flags); devpos, (guint) flags);
} }
/* FIXME: Why do we get 1024 sometimes and not a multiple of /* FIXME: Why do we get 1024 sometimes and not a multiple of
@ -384,26 +372,21 @@ gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
timestamp = 0; timestamp = 0;
} }
ret = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (self), *buf = gst_buffer_new_and_alloc (nsamples * sizeof (gint16));
devpos,
nsamples * sizeof (gint16), GST_PAD_CAPS (GST_BASE_SRC_PAD (self)), buf);
if (ret == GST_FLOW_OK) { GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
guint i; GST_BUFFER_TIMESTAMP (*buf) = timestamp;
gint16 *dst; GST_BUFFER_DURATION (*buf) = duration;
GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer; gst_buffer_map (*buf, &minfo, GST_MAP_WRITE);
GST_BUFFER_TIMESTAMP (*buf) = timestamp; dst = (gint16 *) minfo.data;
GST_BUFFER_DURATION (*buf) = duration; for (i = 0; i < nsamples; i++) {
*dst = *samples;
dst = (gint16 *) GST_BUFFER_DATA (*buf); samples += 2;
for (i = 0; i < nsamples; i++) { dst++;
*dst = *samples;
samples += 2;
dst++;
}
} }
gst_buffer_unmap (*buf, &minfo);
hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read); hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
if (hr != S_OK) { if (hr != S_OK) {

2
sys/wasapi/gstwasapisrc.h
View file

@ -22,8 +22,6 @@
#include "gstwasapiutil.h" #include "gstwasapiutil.h"
#include <gst/base/gstpushsrc.h>
G_BEGIN_DECLS G_BEGIN_DECLS
#define GST_TYPE_WASAPI_SRC \ #define GST_TYPE_WASAPI_SRC \

9
sys/wasapi/gstwasapiutil.c
View file

@ -28,18 +28,23 @@
const CLSID CLSID_MMDeviceEnumerator = { 0xbcde0395, 0xe52f, 0x467c, const CLSID CLSID_MMDeviceEnumerator = { 0xbcde0395, 0xe52f, 0x467c,
{0x8e, 0x3d, 0xc4, 0x57, 0x92, 0x91, 0x69, 0x2e} {0x8e, 0x3d, 0xc4, 0x57, 0x92, 0x91, 0x69, 0x2e}
}; };
const IID IID_IMMDeviceEnumerator = { 0xa95664d2, 0x9614, 0x4f35, const IID IID_IMMDeviceEnumerator = { 0xa95664d2, 0x9614, 0x4f35,
{0xa7, 0x46, 0xde, 0x8d, 0xb6, 0x36, 0x17, 0xe6} {0xa7, 0x46, 0xde, 0x8d, 0xb6, 0x36, 0x17, 0xe6}
}; };
const IID IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32, const IID IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32,
{0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2} {0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2}
}; };
const IID IID_IAudioClock = { 0xcd63314f, 0x3fba, 0x4a1b, const IID IID_IAudioClock = { 0xcd63314f, 0x3fba, 0x4a1b,
{0x81, 0x2c, 0xef, 0x96, 0x35, 0x87, 0x28, 0xe7} {0x81, 0x2c, 0xef, 0x96, 0x35, 0x87, 0x28, 0xe7}
}; };
const IID IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0, const IID IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0,
{0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17} {0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17}
}; };
const IID IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483, const IID IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483,
{0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2} {0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2}
}; };
@ -147,7 +152,7 @@ gst_wasapi_util_get_default_device_client (GstElement * element,
WAVEFORMATEXTENSIBLE format; WAVEFORMATEXTENSIBLE format;
hr = CoCreateInstance (&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL, hr = CoCreateInstance (&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
&IID_IMMDeviceEnumerator, &enumerator); &IID_IMMDeviceEnumerator, (void **) &enumerator);
if (hr != S_OK) { if (hr != S_OK) {
GST_ERROR_OBJECT (element, "CoCreateInstance (MMDeviceEnumerator) failed"); GST_ERROR_OBJECT (element, "CoCreateInstance (MMDeviceEnumerator) failed");
goto beach; goto beach;
@ -162,7 +167,7 @@ gst_wasapi_util_get_default_device_client (GstElement * element,
} }
hr = IMMDevice_Activate (device, &IID_IAudioClient, CLSCTX_ALL, NULL, hr = IMMDevice_Activate (device, &IID_IAudioClient, CLSCTX_ALL, NULL,
&client); (void **) &client);
if (hr != S_OK) { if (hr != S_OK) {
GST_ERROR_OBJECT (element, "IMMDevice::Activate (IID_IAudioClient) failed"); GST_ERROR_OBJECT (element, "IMMDevice::Activate (IID_IAudioClient) failed");
goto beach; goto beach;

1
sys/wasapi/gstwasapiutil.h
View file

@ -21,6 +21,7 @@
#define __GST_WASAPI_UTIL_H__ #define __GST_WASAPI_UTIL_H__
#include <gst/gst.h> #include <gst/gst.h>
#include <gst/audio/audio.h>
#include <audioclient.h> #include <audioclient.h>