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-rw-r--r--sys/wasapi/gstwasapisink.c267
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diff --git a/sys/wasapi/gstwasapisink.c b/sys/wasapi/gstwasapisink.c
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+/*
+ * Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-wasapisink
+ *
+ * Provides audio playback using the Windows Audio Session API available with
+ * Vista and newer.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch-0.10 -v audiotestsrc samplesperbuffer=160 ! wasapisink
+ * ]| Generate 20 ms buffers and render to the default audio device.
+ * </refsect2>
+ */
+
+#include "gstwasapisink.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_wasapi_sink_debug);
+#define GST_CAT_DEFAULT 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)));
+
+static void gst_wasapi_sink_dispose (GObject * object);
+static void gst_wasapi_sink_finalize (GObject * object);
+
+static void gst_wasapi_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
+ GstClockTime * start, GstClockTime * end);
+static gboolean gst_wasapi_sink_start (GstBaseSink * sink);
+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);
+ static GstElementDetails element_details = {
+ "WasapiSrc",
+ "Sink/Audio",
+ "Stream audio to an audio capture device through WASAPI",
+ "Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>"
+ };
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&sink_template));
+ gst_element_class_set_details (element_class, &element_details);
+}
+
+static void
+gst_wasapi_sink_class_init (GstWasapiSinkClass * klass)
+{
+ GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
+ GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
+
+ gobject_class->dispose = gst_wasapi_sink_dispose;
+ gobject_class->finalize = gst_wasapi_sink_finalize;
+
+ gstbasesink_class->get_times = gst_wasapi_sink_get_times;
+ gstbasesink_class->start = gst_wasapi_sink_start;
+ gstbasesink_class->stop = gst_wasapi_sink_stop;
+ gstbasesink_class->render = gst_wasapi_sink_render;
+
+ GST_DEBUG_CATEGORY_INIT (gst_wasapi_sink_debug, "wasapisink",
+ 0, "Windows audio session API sink");
+}
+
+static void
+gst_wasapi_sink_init (GstWasapiSink * self, GstWasapiSinkClass * gclass)
+{
+ self->rate = 8000;
+ self->buffer_time = 20 * GST_MSECOND;
+ self->period_time = 20 * GST_MSECOND;
+ self->latency = GST_CLOCK_TIME_NONE;
+
+ self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
+
+ CoInitialize (NULL);
+}
+
+static void
+gst_wasapi_sink_dispose (GObject * object)
+{
+ GstWasapiSink *self = GST_WASAPI_SINK (object);
+
+ if (self->event_handle != NULL) {
+ CloseHandle (self->event_handle);
+ self->event_handle = NULL;
+ }
+
+ G_OBJECT_CLASS (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);
+}
+
+static void
+gst_wasapi_sink_get_times (GstBaseSink * sink,
+ GstBuffer * buffer, GstClockTime * start, GstClockTime * end)
+{
+ GstWasapiSink *self = GST_WASAPI_SINK (sink);
+
+ if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
+ *start = GST_BUFFER_TIMESTAMP (buffer);
+
+ if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
+ *end = *start + GST_BUFFER_DURATION (buffer);
+ } else {
+ *end = *start + self->buffer_time;
+ }
+
+ *start += self->latency;
+ *end += self->latency;
+ }
+}
+
+static gboolean
+gst_wasapi_sink_start (GstBaseSink * sink)
+{
+ GstWasapiSink *self = GST_WASAPI_SINK (sink);
+ gboolean res = FALSE;
+ IAudioClient *client = NULL;
+ HRESULT hr;
+ IAudioRenderClient *render_client = NULL;
+
+ if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
+ FALSE, self->rate, self->buffer_time, self->period_time,
+ AUDCLNT_STREAMFLAGS_EVENTCALLBACK, &client, &self->latency))
+ goto beach;
+
+ hr = IAudioClient_SetEventHandle (client, self->event_handle);
+ if (hr != S_OK) {
+ GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
+ goto beach;
+ }
+
+ hr = IAudioClient_GetService (client, &IID_IAudioRenderClient,
+ &render_client);
+ if (hr != S_OK) {
+ GST_ERROR_OBJECT (self, "IAudioClient::GetService "
+ "(IID_IAudioRenderClient) failed");
+ goto beach;
+ }
+
+ hr = IAudioClient_Start (client);
+ if (hr != S_OK) {
+ GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
+ goto beach;
+ }
+
+ self->client = client;
+ self->render_client = render_client;
+
+ res = TRUE;
+
+beach:
+ if (!res) {
+ if (render_client != NULL)
+ IUnknown_Release (render_client);
+
+ if (client != NULL)
+ IUnknown_Release (client);
+ }
+
+ return res;
+}
+
+static gboolean
+gst_wasapi_sink_stop (GstBaseSink * sink)
+{
+ GstWasapiSink *self = GST_WASAPI_SINK (sink);
+
+ if (self->client != NULL) {
+ IAudioClient_Stop (self->client);
+ }
+
+ if (self->render_client != NULL) {
+ IUnknown_Release (self->render_client);
+ self->render_client = NULL;
+ }
+
+ if (self->client != NULL) {
+ IUnknown_Release (self->client);
+ self->client = NULL;
+ }
+
+ return TRUE;
+}
+
+static GstFlowReturn
+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);
+ gint16 *dst = NULL;
+ guint nsamples = GST_BUFFER_SIZE (buffer) / sizeof (gint16);
+ guint i;
+
+ WaitForSingleObject (self->event_handle, INFINITE);
+
+ hr = IAudioRenderClient_GetBuffer (self->render_client, nsamples,
+ (BYTE **) & dst);
+ if (hr != S_OK) {
+ GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL),
+ ("IAudioRenderClient::GetBuffer () failed: %s",
+ gst_wasapi_util_hresult_to_string (hr)));
+ ret = GST_FLOW_ERROR;
+ goto beach;
+ }
+
+ for (i = 0; i < nsamples; i++) {
+ dst[0] = *src;
+ dst[1] = *src;
+
+ src++;
+ dst += 2;
+ }
+
+ hr = IAudioRenderClient_ReleaseBuffer (self->render_client, nsamples, 0);
+ if (hr != S_OK) {
+ GST_ERROR_OBJECT (self, "IAudioRenderClient::ReleaseBuffer () failed: %s",
+ gst_wasapi_util_hresult_to_string (hr));
+ ret = GST_FLOW_ERROR;
+ goto beach;
+ }
+
+beach:
+ return ret;
+}
generated by cgit v1.2.1 (git 2.18.0) at 2025-01-30 17:49:06 +0000