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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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264 lines
7.2 KiB
C
264 lines
7.2 KiB
C
/*
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* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-wasapisink
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*
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* Provides audio playback using the Windows Audio Session API available with
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* Vista and newer.
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*
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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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* ]| Generate 20 ms buffers and render to the default audio device.
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* </refsect2>
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*/
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#include "gstwasapisink.h"
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GST_DEBUG_CATEGORY_STATIC (gst_wasapi_sink_debug);
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#define GST_CAT_DEFAULT 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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static void gst_wasapi_sink_dispose (GObject * object);
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static void gst_wasapi_sink_finalize (GObject * object);
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static void gst_wasapi_sink_get_times (GstBaseSink * sink, GstBuffer * buffer,
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GstClockTime * start, GstClockTime * end);
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static gboolean gst_wasapi_sink_start (GstBaseSink * sink);
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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_details_simple (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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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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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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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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gstbasesink_class->render = gst_wasapi_sink_render;
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GST_DEBUG_CATEGORY_INIT (gst_wasapi_sink_debug, "wasapisink",
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0, "Windows audio session API sink");
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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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{
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self->rate = 8000;
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self->buffer_time = 20 * GST_MSECOND;
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self->period_time = 20 * GST_MSECOND;
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self->latency = GST_CLOCK_TIME_NONE;
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self->event_handle = CreateEvent (NULL, FALSE, FALSE, NULL);
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CoInitialize (NULL);
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}
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static void
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gst_wasapi_sink_dispose (GObject * object)
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{
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GstWasapiSink *self = GST_WASAPI_SINK (object);
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if (self->event_handle != NULL) {
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CloseHandle (self->event_handle);
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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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}
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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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}
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static void
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gst_wasapi_sink_get_times (GstBaseSink * sink,
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GstBuffer * buffer, GstClockTime * start, GstClockTime * end)
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{
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GstWasapiSink *self = GST_WASAPI_SINK (sink);
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if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
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*start = GST_BUFFER_TIMESTAMP (buffer);
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if (GST_BUFFER_DURATION_IS_VALID (buffer)) {
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*end = *start + GST_BUFFER_DURATION (buffer);
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} else {
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*end = *start + self->buffer_time;
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}
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*start += self->latency;
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*end += self->latency;
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}
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}
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static gboolean
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gst_wasapi_sink_start (GstBaseSink * sink)
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{
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GstWasapiSink *self = GST_WASAPI_SINK (sink);
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gboolean res = FALSE;
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IAudioClient *client = NULL;
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HRESULT hr;
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IAudioRenderClient *render_client = NULL;
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if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
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FALSE, self->rate, self->buffer_time, self->period_time,
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AUDCLNT_STREAMFLAGS_EVENTCALLBACK, &client, &self->latency))
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goto beach;
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hr = IAudioClient_SetEventHandle (client, self->event_handle);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
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goto beach;
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}
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hr = IAudioClient_GetService (client, &IID_IAudioRenderClient,
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&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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goto beach;
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}
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hr = IAudioClient_Start (client);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
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goto beach;
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}
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self->client = client;
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self->render_client = render_client;
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res = TRUE;
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beach:
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if (!res) {
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if (render_client != NULL)
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IUnknown_Release (render_client);
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if (client != NULL)
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IUnknown_Release (client);
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}
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return res;
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}
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static gboolean
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gst_wasapi_sink_stop (GstBaseSink * sink)
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{
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GstWasapiSink *self = GST_WASAPI_SINK (sink);
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if (self->client != NULL) {
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IAudioClient_Stop (self->client);
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}
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if (self->render_client != NULL) {
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IUnknown_Release (self->render_client);
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self->render_client = NULL;
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}
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if (self->client != NULL) {
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IUnknown_Release (self->client);
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self->client = NULL;
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}
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return TRUE;
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}
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static GstFlowReturn
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gst_wasapi_sink_render (GstBaseSink * sink, GstBuffer * buffer)
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{
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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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gint16 *dst = NULL;
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guint nsamples = GST_BUFFER_SIZE (buffer) / sizeof (gint16);
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guint i;
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WaitForSingleObject (self->event_handle, INFINITE);
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hr = IAudioRenderClient_GetBuffer (self->render_client, nsamples,
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(BYTE **) & dst);
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if (hr != S_OK) {
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GST_ELEMENT_ERROR (self, RESOURCE, WRITE, (NULL),
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("IAudioRenderClient::GetBuffer () failed: %s",
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gst_wasapi_util_hresult_to_string (hr)));
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ret = GST_FLOW_ERROR;
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goto beach;
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}
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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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src++;
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dst += 2;
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}
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hr = IAudioRenderClient_ReleaseBuffer (self->render_client, nsamples, 0);
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if (hr != S_OK) {
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GST_ERROR_OBJECT (self, "IAudioRenderClient::ReleaseBuffer () failed: %s",
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gst_wasapi_util_hresult_to_string (hr));
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ret = GST_FLOW_ERROR;
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goto beach;
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}
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beach:
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return ret;
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}
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