gstreamer/sys/wasapi/gstwasapisink.c
Tim-Philipp Müller 32ba17cd0f Use gst_element_class_set_static_metadata()
where possible. Avoids some string copies. Also re-indent
some stuff. Also some indent fixes here and there.
2012-10-17 17:46:34 +01:00

264 lines
7.2 KiB
C

/*
* 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);
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
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;
}