mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-23 10:11:08 +00:00
wasapi: Initial port to 1.0
This should really use GstAudioSink and GstAudioSrc.
This commit is contained in:
Sebastian Dröge
parent
d5d37fafa7
commit
e7a69bb8de
9 changed files with 83 additions and 93 deletions
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@ -328,7 +328,7 @@ GST_PLUGINS_NONPORTED=" aiff \
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gsettings ladspa \
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musepack musicbrainz nas neon ofa openal rsvg sdl sndfile timidity \
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directdraw direct3d9 acm wininet \
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xvid lv2 teletextdec sndio wasapi"
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xvid lv2 teletextdec sndio"
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AC_SUBST(GST_PLUGINS_NONPORTED)
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dnl these are all the gst plug-ins, compilable without additional libs
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@ -7,7 +7,7 @@ libgstwasapi_la_SOURCES = gstwasapi.c \
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libgstwasapi_la_CFLAGS = $(GST_BASE_CFLAGS) $(GST_CFLAGS) -DCOBJMACROS=1
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libgstwasapi_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) \
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-lgstaudio-$(GST_MAJORMINOR) -lgstinterfaces-$(GST_MAJORMINOR) \
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-lgstaudio-$(GST_API_VERSION) \
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$(WASAPI_LIBS)
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libgstwasapi_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
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libgstwasapi_la_LIBTOOLFLAGS = --tag=disable-static
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@ -42,4 +42,4 @@ GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
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GST_VERSION_MINOR,
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wasapi,
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"Windows audio session API plugin",
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plugin_init, VERSION, "LGPL", "GStreamer", "http://gstreamer.net/")
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plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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@ -26,7 +26,7 @@
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch-0.10 -v audiotestsrc samplesperbuffer=160 ! wasapisink
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* gst-launch-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapisink
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* ]| Generate 20 ms buffers and render to the default audio device.
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* </refsect2>
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*/
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@ -42,13 +42,10 @@ GST_DEBUG_CATEGORY_STATIC (gst_wasapi_sink_debug);
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static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) 8000, "
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"channels = (int) 1, "
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"signed = (boolean) TRUE, "
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER)));
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) S16LE, "
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"layout = (string) interleaved, "
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"rate = (int) 8000, " "channels = (int) 1"));
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static void gst_wasapi_sink_dispose (GObject * object);
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static void gst_wasapi_sink_finalize (GObject * object);
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@ -60,31 +57,25 @@ static gboolean gst_wasapi_sink_stop (GstBaseSink * sink);
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static GstFlowReturn gst_wasapi_sink_render (GstBaseSink * sink,
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GstBuffer * buffer);
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GST_BOILERPLATE (GstWasapiSink, gst_wasapi_sink, GstBaseSink,
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GST_TYPE_BASE_SINK);
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static void
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gst_wasapi_sink_base_init (gpointer gclass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_set_static_metadata (element_class, "WasapiSrc",
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"Sink/Audio",
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"Stream audio to an audio capture device through WASAPI",
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"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
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}
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G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_BASE_SINK);
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static void
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gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
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gobject_class->dispose = gst_wasapi_sink_dispose;
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gobject_class->finalize = gst_wasapi_sink_finalize;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&sink_template));
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gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
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"Sink/Audio",
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"Stream audio to an audio capture device through WASAPI",
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"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
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gstbasesink_class->get_times = gst_wasapi_sink_get_times;
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gstbasesink_class->start = gst_wasapi_sink_start;
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gstbasesink_class->stop = gst_wasapi_sink_stop;
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@ -95,7 +86,7 @@ gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
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}
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static void
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gst_wasapi_sink_init (GstWasapiSink * self, GstWasapiSinkClass * gclass)
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gst_wasapi_sink_init (GstWasapiSink * self)
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{
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self->rate = 8000;
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self->buffer_time = 20 * GST_MSECOND;
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@ -117,17 +108,15 @@ gst_wasapi_sink_dispose (GObject * object)
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self->event_handle = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->dispose (object);
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}
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static void
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gst_wasapi_sink_finalize (GObject * object)
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{
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GstWasapiSink *self = GST_WASAPI_SINK (object);
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CoUninitialize ();
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G_OBJECT_CLASS (parent_class)->finalize (object);
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G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->finalize (object);
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}
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static void
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@ -171,7 +160,7 @@ gst_wasapi_sink_start (GstBaseSink * sink)
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}
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hr = IAudioClient_GetService (client, &IID_IAudioRenderClient,
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&render_client);
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(void **) &render_client);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::GetService "
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"(IID_IAudioRenderClient) failed");
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@ -229,11 +218,23 @@ gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
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GstWasapiSink *self = GST_WASAPI_SINK (sink);
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GstFlowReturn ret = GST_FLOW_OK;
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HRESULT hr;
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gint16 *src = (gint16 *) GST_BUFFER_DATA (buffer);
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GstMapInfo minfo;
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const gint16 *src;
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gint16 *dst = NULL;
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guint nsamples = GST_BUFFER_SIZE (buffer) / sizeof (gint16);
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guint nsamples;
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guint i;
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memset (&minfo, 0, sizeof (minfo));
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if (!gst_buffer_map (buffer, &minfo, GST_MAP_READ)) {
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GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL),
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("Failed to map input buffer"));
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ret = GST_FLOW_ERROR;
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goto beach;
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}
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nsamples = minfo.size / sizeof (gint16);
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WaitForSingleObject (self->event_handle, INFINITE);
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hr = IAudioRenderClient_GetBuffer (self->render_client, nsamples,
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@ -246,6 +247,7 @@ gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
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goto beach;
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}
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src = (const gint16 *) minfo.data;
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for (i = 0; i < nsamples; i++) {
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dst[0] = *src;
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dst[1] = *src;
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@ -263,5 +265,8 @@ gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
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}
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beach:
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if (minfo.data)
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gst_buffer_unmap (buffer, &minfo);
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return ret;
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}
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@ -22,8 +22,6 @@
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#include "gstwasapiutil.h"
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#include <gst/base/gstbasesink.h>
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G_BEGIN_DECLS
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#define GST_TYPE_WASAPI_SINK \
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@ -26,7 +26,7 @@
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch-0.10 -v wasapisrc ! fakesink
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* gst-launch-1.0 -v wasapisrc ! fakesink
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* ]| Capture from the default audio device and render to fakesink.
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* </refsect2>
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*/
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@ -43,13 +43,10 @@ GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw-int, "
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"width = (int) 16, "
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"depth = (int) 16, "
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"rate = (int) 8000, "
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"channels = (int) 1, "
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"signed = (boolean) TRUE, "
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"endianness = (int) " G_STRINGIFY (G_BYTE_ORDER)));
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GST_STATIC_CAPS ("audio/x-raw, "
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"format = (string) S16LE, "
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"layout = (string) interleaved, "
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"rate = (int) 8000, " "channels = (int) 1"));
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static void gst_wasapi_src_dispose (GObject * object);
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static void gst_wasapi_src_finalize (GObject * object);
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@ -65,20 +62,7 @@ static GstFlowReturn gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf);
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static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
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gpointer user_data);
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GST_BOILERPLATE (GstWasapiSrc, gst_wasapi_src, GstPushSrc, GST_TYPE_PUSH_SRC);
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static void
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gst_wasapi_src_base_init (gpointer gclass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_static_metadata (element_class, "WasapiSrc",
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"Source/Audio",
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"Stream audio from an audio capture device through WASAPI",
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"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
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}
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G_DEFINE_TYPE (GstWasapiSrc, gst_wasapi_src, GST_TYPE_PUSH_SRC);
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static void
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gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
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@ -93,6 +77,13 @@ gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
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gstelement_class->provide_clock = gst_wasapi_src_provide_clock;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_static_metadata (gstelement_class, "WasapiSrc",
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"Source/Audio",
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"Stream audio from an audio capture device through WASAPI",
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"Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>");
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gstbasesrc_class->start = gst_wasapi_src_start;
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gstbasesrc_class->stop = gst_wasapi_src_stop;
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gstbasesrc_class->query = gst_wasapi_src_query;
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@ -104,7 +95,7 @@ gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
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}
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static void
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gst_wasapi_src_init (GstWasapiSrc * self, GstWasapiSrcClass * gclass)
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gst_wasapi_src_init (GstWasapiSrc * self)
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{
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GstBaseSrc *basesrc = GST_BASE_SRC (self);
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@ -120,14 +111,9 @@ gst_wasapi_src_init (GstWasapiSrc * self, GstWasapiSrcClass * gclass)
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self->start_time = GST_CLOCK_TIME_NONE;
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self->next_time = GST_CLOCK_TIME_NONE;
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#if GST_CHECK_VERSION(0, 10, 31) || (GST_CHECK_VERSION(0, 10, 30) && GST_VERSION_NANO > 0)
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self->clock = gst_audio_clock_new_full ("GstWasapiSrcClock",
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self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
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gst_wasapi_src_get_time, gst_object_ref (self),
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(GDestroyNotify) gst_object_unref);
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#else
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self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
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gst_wasapi_src_get_time, self);
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#endif
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CoInitialize (NULL);
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}
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@ -142,17 +128,15 @@ gst_wasapi_src_dispose (GObject * object)
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self->clock = NULL;
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}
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G_OBJECT_CLASS (parent_class)->dispose (object);
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G_OBJECT_CLASS (gst_wasapi_src_parent_class)->dispose (object);
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}
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static void
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gst_wasapi_src_finalize (GObject * object)
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{
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GstWasapiSrc *self = GST_WASAPI_SRC (object);
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CoUninitialize ();
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G_OBJECT_CLASS (parent_class)->finalize (object);
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G_OBJECT_CLASS (gst_wasapi_src_parent_class)->finalize (object);
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}
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static GstClock *
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@ -196,7 +180,7 @@ gst_wasapi_src_start (GstBaseSrc * src)
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&self->latency))
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goto beach;
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hr = IAudioClient_GetService (client, &IID_IAudioClock, &client_clock);
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hr = IAudioClient_GetService (client, &IID_IAudioClock, (void**) &client_clock);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
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"failed");
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@ -210,7 +194,7 @@ gst_wasapi_src_start (GstBaseSrc * src)
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}
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hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
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&capture_client);
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(void**) &capture_client);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::GetService "
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"(IID_IAudioCaptureClient) failed");
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@ -298,7 +282,8 @@ gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query)
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}
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default:
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ret = GST_BASE_SRC_CLASS (parent_class)->query (src, query);
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ret =
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GST_BASE_SRC_CLASS (gst_wasapi_src_parent_class)->query (src, query);
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break;
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}
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@ -317,6 +302,9 @@ gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
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guint32 nsamples_read = 0, nsamples;
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DWORD flags = 0;
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guint64 devpos;
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guint i;
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GstMapInfo minfo;
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gint16 *dst;
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GST_OBJECT_LOCK (self);
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clock = GST_ELEMENT_CLOCK (self);
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@ -347,7 +335,7 @@ gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
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if (flags != 0) {
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GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x",
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devpos, flags);
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devpos, (guint) flags);
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}
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/* FIXME: Why do we get 1024 sometimes and not a multiple of
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@ -384,26 +372,21 @@ gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
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timestamp = 0;
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}
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ret = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (self),
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devpos,
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nsamples * sizeof (gint16), GST_PAD_CAPS (GST_BASE_SRC_PAD (self)), buf);
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*buf = gst_buffer_new_and_alloc (nsamples * sizeof (gint16));
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if (ret == GST_FLOW_OK) {
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guint i;
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gint16 *dst;
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GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
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GST_BUFFER_TIMESTAMP (*buf) = timestamp;
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GST_BUFFER_DURATION (*buf) = duration;
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GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
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GST_BUFFER_TIMESTAMP (*buf) = timestamp;
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GST_BUFFER_DURATION (*buf) = duration;
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gst_buffer_map (*buf, &minfo, GST_MAP_WRITE);
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dst = (gint16 *) minfo.data;
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for (i = 0; i < nsamples; i++) {
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*dst = *samples;
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dst = (gint16 *) GST_BUFFER_DATA (*buf);
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for (i = 0; i < nsamples; i++) {
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*dst = *samples;
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samples += 2;
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dst++;
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}
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samples += 2;
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dst++;
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}
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gst_buffer_unmap (*buf, &minfo);
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hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
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if (hr != S_OK) {
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@ -22,8 +22,6 @@
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#include "gstwasapiutil.h"
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#include <gst/base/gstpushsrc.h>
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G_BEGIN_DECLS
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#define GST_TYPE_WASAPI_SRC \
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@ -28,18 +28,23 @@
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|||
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;
|
||||
|
|
|
|||
|
|
@ -21,6 +21,7 @@
|
|||
#define __GST_WASAPI_UTIL_H__
|
||||
|
||||
#include <gst/gst.h>
|
||||
#include <gst/audio/audio.h>
|
||||
|
||||
#include <audioclient.h>
|
||||
|
||||
|
|
|
|||
Loading…
Reference in a new issue