diff --git a/configure.ac b/configure.ac
index 31de789d48..b4cba5feb5 100644
--- a/configure.ac
+++ b/configure.ac
@@ -328,7 +328,7 @@ GST_PLUGINS_NONPORTED=" aiff \
gsettings ladspa \
musepack musicbrainz nas neon ofa openal rsvg sdl sndfile timidity \
directdraw direct3d9 acm wininet \
- xvid lv2 teletextdec sndio wasapi"
+ xvid lv2 teletextdec sndio"
AC_SUBST(GST_PLUGINS_NONPORTED)
dnl these are all the gst plug-ins, compilable without additional libs
diff --git a/sys/wasapi/Makefile.am b/sys/wasapi/Makefile.am
index 2f5f459803..93fa3a1ebd 100644
--- a/sys/wasapi/Makefile.am
+++ b/sys/wasapi/Makefile.am
@@ -7,7 +7,7 @@ libgstwasapi_la_SOURCES = gstwasapi.c \
libgstwasapi_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) -DCOBJMACROS=1
libgstwasapi_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) \
- -lgstaudio-$(GST_MAJORMINOR) -lgstinterfaces-$(GST_MAJORMINOR) \
+ -lgstaudio-$(GST_API_VERSION) \
$(WASAPI_LIBS)
libgstwasapi_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
libgstwasapi_la_LIBTOOLFLAGS = --tag=disable-static
diff --git a/sys/wasapi/gstwasapi.c b/sys/wasapi/gstwasapi.c
index 9c901d08cc..920e2d0d96 100644
--- a/sys/wasapi/gstwasapi.c
+++ b/sys/wasapi/gstwasapi.c
@@ -42,4 +42,4 @@ GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
wasapi,
"Windows audio session API plugin",
- plugin_init, VERSION, "LGPL", "GStreamer", "http://gstreamer.net/")
+ plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
diff --git a/sys/wasapi/gstwasapisink.c b/sys/wasapi/gstwasapisink.c
index 193a608f1a..f16e611cfd 100644
--- a/sys/wasapi/gstwasapisink.c
+++ b/sys/wasapi/gstwasapisink.c
@@ -26,7 +26,7 @@
*
* Example pipelines
* |[
- * 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.
*
*/
@@ -42,13 +42,10 @@ GST_DEBUG_CATEGORY_STATIC (gst_wasapi_sink_debug);
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "rate = (int) 8000, "
- "channels = (int) 1, "
- "signed = (boolean) TRUE, "
- "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER)));
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) S16LE, "
+ "layout = (string) interleaved, "
+ "rate = (int) 8000, " "channels = (int) 1"));
static void gst_wasapi_sink_dispose (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,
GstBuffer * buffer);
-GST_BOILERPLATE (GstWasapiSink, gst_wasapi_sink, GstBaseSink,
- 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 ");
-}
+G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_BASE_SINK);
static void
gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+ GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
gobject_class->dispose = gst_wasapi_sink_dispose;
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 ");
+
gstbasesink_class->get_times = gst_wasapi_sink_get_times;
gstbasesink_class->start = gst_wasapi_sink_start;
gstbasesink_class->stop = gst_wasapi_sink_stop;
@@ -95,7 +86,7 @@ gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
}
static void
-gst_wasapi_sink_init (GstWasapiSink * self, GstWasapiSinkClass * gclass)
+gst_wasapi_sink_init (GstWasapiSink * self)
{
self->rate = 8000;
self->buffer_time = 20 * GST_MSECOND;
@@ -117,17 +108,15 @@ gst_wasapi_sink_dispose (GObject * object)
self->event_handle = NULL;
}
- G_OBJECT_CLASS (parent_class)->dispose (object);
+ G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->dispose (object);
}
static void
gst_wasapi_sink_finalize (GObject * object)
{
- GstWasapiSink *self = GST_WASAPI_SINK (object);
-
CoUninitialize ();
- G_OBJECT_CLASS (parent_class)->finalize (object);
+ G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->finalize (object);
}
static void
@@ -171,7 +160,7 @@ gst_wasapi_sink_start (GstBaseSink * sink)
}
hr = IAudioClient_GetService (client, &IID_IAudioRenderClient,
- &render_client);
+ (void **) &render_client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService "
"(IID_IAudioRenderClient) failed");
@@ -229,11 +218,23 @@ gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
GstWasapiSink *self = GST_WASAPI_SINK (sink);
GstFlowReturn ret = GST_FLOW_OK;
HRESULT hr;
- gint16 *src = (gint16 *) GST_BUFFER_DATA (buffer);
+ GstMapInfo minfo;
+ const gint16 *src;
gint16 *dst = NULL;
- guint nsamples = GST_BUFFER_SIZE (buffer) / sizeof (gint16);
+ guint nsamples;
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);
hr = IAudioRenderClient_GetBuffer (self->render_client, nsamples,
@@ -246,6 +247,7 @@ gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
goto beach;
}
+ src = (const gint16 *) minfo.data;
for (i = 0; i < nsamples; i++) {
dst[0] = *src;
dst[1] = *src;
@@ -263,5 +265,8 @@ gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
}
beach:
+ if (minfo.data)
+ gst_buffer_unmap (buffer, &minfo);
+
return ret;
}
diff --git a/sys/wasapi/gstwasapisink.h b/sys/wasapi/gstwasapisink.h
index bbdc85dff4..2c354917b3 100644
--- a/sys/wasapi/gstwasapisink.h
+++ b/sys/wasapi/gstwasapisink.h
@@ -22,8 +22,6 @@
#include "gstwasapiutil.h"
-#include
-
G_BEGIN_DECLS
#define GST_TYPE_WASAPI_SINK \
diff --git a/sys/wasapi/gstwasapisrc.c b/sys/wasapi/gstwasapisrc.c
index e7e24e7101..3cee9cefe2 100644
--- a/sys/wasapi/gstwasapisrc.c
+++ b/sys/wasapi/gstwasapisrc.c
@@ -26,7 +26,7 @@
*
* Example pipelines
* |[
- * gst-launch-0.10 -v wasapisrc ! fakesink
+ * gst-launch-1.0 -v wasapisrc ! fakesink
* ]| Capture from the default audio device and render to fakesink.
*
*/
@@ -43,13 +43,10 @@ GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "rate = (int) 8000, "
- "channels = (int) 1, "
- "signed = (boolean) TRUE, "
- "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER)));
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) S16LE, "
+ "layout = (string) interleaved, "
+ "rate = (int) 8000, " "channels = (int) 1"));
static void gst_wasapi_src_dispose (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,
gpointer user_data);
-GST_BOILERPLATE (GstWasapiSrc, gst_wasapi_src, GstPushSrc, 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 ");
-}
+G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_PUSH_SRC);
static void
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;
+ 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 ");
+
gstbasesrc_class->start = gst_wasapi_src_start;
gstbasesrc_class->stop = gst_wasapi_src_stop;
gstbasesrc_class->query = gst_wasapi_src_query;
@@ -104,7 +95,7 @@ gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
}
static void
-gst_wasapi_src_init (GstWasapiSrc * self, GstWasapiSrcClass * gclass)
+gst_wasapi_src_init (GstWasapiSrc * 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->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_full ("GstWasapiSrcClock",
+ self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
gst_wasapi_src_get_time, gst_object_ref (self),
(GDestroyNotify) gst_object_unref);
-#else
- self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
- gst_wasapi_src_get_time, self);
-#endif
CoInitialize (NULL);
}
@@ -142,17 +128,15 @@ gst_wasapi_src_dispose (GObject * object)
self->clock = NULL;
}
- G_OBJECT_CLASS (parent_class)->dispose (object);
+ G_OBJECT_CLASS (gst_wasapi_src_parent_class)->dispose (object);
}
static void
gst_wasapi_src_finalize (GObject * object)
{
- GstWasapiSrc *self = GST_WASAPI_SRC (object);
-
CoUninitialize ();
- G_OBJECT_CLASS (parent_class)->finalize (object);
+ G_OBJECT_CLASS (gst_wasapi_src_parent_class)->finalize (object);
}
static GstClock *
@@ -196,7 +180,7 @@ gst_wasapi_src_start (GstBaseSrc * src)
&self->latency))
goto beach;
- hr = IAudioClient_GetService (client, &IID_IAudioClock, &client_clock);
+ hr = IAudioClient_GetService (client, &IID_IAudioClock, (void**) &client_clock);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
"failed");
@@ -210,7 +194,7 @@ gst_wasapi_src_start (GstBaseSrc * src)
}
hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
- &capture_client);
+ (void**) &capture_client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService "
"(IID_IAudioCaptureClient) failed");
@@ -298,7 +282,8 @@ gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query)
}
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;
}
@@ -317,6 +302,9 @@ gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
guint32 nsamples_read = 0, nsamples;
DWORD flags = 0;
guint64 devpos;
+ guint i;
+ GstMapInfo minfo;
+ gint16 *dst;
GST_OBJECT_LOCK (self);
clock = GST_ELEMENT_CLOCK (self);
@@ -347,7 +335,7 @@ gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
if (flags != 0) {
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
@@ -384,26 +372,21 @@ gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
timestamp = 0;
}
- ret = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (self),
- devpos,
- nsamples * sizeof (gint16), GST_PAD_CAPS (GST_BASE_SRC_PAD (self)), buf);
+ *buf = gst_buffer_new_and_alloc (nsamples * sizeof (gint16));
- if (ret == GST_FLOW_OK) {
- guint i;
- gint16 *dst;
+ GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
+ GST_BUFFER_TIMESTAMP (*buf) = timestamp;
+ GST_BUFFER_DURATION (*buf) = duration;
- GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
- GST_BUFFER_TIMESTAMP (*buf) = timestamp;
- GST_BUFFER_DURATION (*buf) = duration;
+ gst_buffer_map (*buf, &minfo, GST_MAP_WRITE);
+ dst = (gint16 *) minfo.data;
+ for (i = 0; i < nsamples; i++) {
+ *dst = *samples;
- dst = (gint16 *) GST_BUFFER_DATA (*buf);
- for (i = 0; i < nsamples; i++) {
- *dst = *samples;
-
- samples += 2;
- dst++;
- }
+ samples += 2;
+ dst++;
}
+ gst_buffer_unmap (*buf, &minfo);
hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
if (hr != S_OK) {
diff --git a/sys/wasapi/gstwasapisrc.h b/sys/wasapi/gstwasapisrc.h
index 5b868c7660..0158d13273 100644
--- a/sys/wasapi/gstwasapisrc.h
+++ b/sys/wasapi/gstwasapisrc.h
@@ -22,8 +22,6 @@
#include "gstwasapiutil.h"
-#include
-
G_BEGIN_DECLS
#define GST_TYPE_WASAPI_SRC \
diff --git a/sys/wasapi/gstwasapiutil.c b/sys/wasapi/gstwasapiutil.c
index ce6b440699..fa3e39bc4a 100644
--- a/sys/wasapi/gstwasapiutil.c
+++ b/sys/wasapi/gstwasapiutil.c
@@ -28,18 +28,23 @@
const CLSID CLSID_MMDeviceEnumerator = { 0xbcde0395, 0xe52f, 0x467c,
{0x8e, 0x3d, 0xc4, 0x57, 0x92, 0x91, 0x69, 0x2e}
};
+
const IID IID_IMMDeviceEnumerator = { 0xa95664d2, 0x9614, 0x4f35,
{0xa7, 0x46, 0xde, 0x8d, 0xb6, 0x36, 0x17, 0xe6}
};
+
const IID IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32,
{0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2}
};
+
const IID IID_IAudioClock = { 0xcd63314f, 0x3fba, 0x4a1b,
{0x81, 0x2c, 0xef, 0x96, 0x35, 0x87, 0x28, 0xe7}
};
+
const IID IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0,
{0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17}
};
+
const IID IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483,
{0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2}
};
@@ -147,7 +152,7 @@ gst_wasapi_util_get_default_device_client (GstElement * element,
WAVEFORMATEXTENSIBLE format;
hr = CoCreateInstance (&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
- &IID_IMMDeviceEnumerator, &enumerator);
+ &IID_IMMDeviceEnumerator, (void **) &enumerator);
if (hr != S_OK) {
GST_ERROR_OBJECT (element, "CoCreateInstance (MMDeviceEnumerator) failed");
goto beach;
@@ -162,7 +167,7 @@ gst_wasapi_util_get_default_device_client (GstElement * element,
}
hr = IMMDevice_Activate (device, &IID_IAudioClient, CLSCTX_ALL, NULL,
- &client);
+ (void **) &client);
if (hr != S_OK) {
GST_ERROR_OBJECT (element, "IMMDevice::Activate (IID_IAudioClient) failed");
goto beach;
diff --git a/sys/wasapi/gstwasapiutil.h b/sys/wasapi/gstwasapiutil.h
index 7ddd81a5dd..b32b05cc73 100644
--- a/sys/wasapi/gstwasapiutil.h
+++ b/sys/wasapi/gstwasapiutil.h
@@ -21,6 +21,7 @@
#define __GST_WASAPI_UTIL_H__
#include
+#include
#include