mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-22 16:26:39 +00:00
327 lines
9.2 KiB
C
327 lines
9.2 KiB
C
/*
|
|
* Copyright (C) 2008 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
|
|
* Copyright (C) 2013 Collabora Ltd.
|
|
* Author: Sebastian Dröge <sebastian.droege@collabora.co.uk>
|
|
*
|
|
* 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., 51 Franklin St, Fifth Floor,
|
|
* Boston, MA 02110-1301, 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-1.0 -v audiotestsrc samplesperbuffer=160 ! wasapisink
|
|
* ]| Generate 20 ms buffers and render to the default audio device.
|
|
* </refsect2>
|
|
*/
|
|
#ifdef HAVE_CONFIG_H
|
|
# include <config.h>
|
|
#endif
|
|
|
|
#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, "
|
|
"format = (string) S16LE, "
|
|
"layout = (string) interleaved, "
|
|
"rate = (int) 44100, " "channels = (int) 2"));
|
|
|
|
static void gst_wasapi_sink_dispose (GObject * object);
|
|
static void gst_wasapi_sink_finalize (GObject * object);
|
|
|
|
static GstCaps *gst_wasapi_sink_get_caps (GstBaseSink * bsink,
|
|
GstCaps * filter);
|
|
static gboolean gst_wasapi_sink_prepare (GstAudioSink * asink,
|
|
GstAudioRingBufferSpec * spec);
|
|
static gboolean gst_wasapi_sink_unprepare (GstAudioSink * asink);
|
|
static gboolean gst_wasapi_sink_open (GstAudioSink * asink);
|
|
static gboolean gst_wasapi_sink_close (GstAudioSink * asink);
|
|
static gint gst_wasapi_sink_write (GstAudioSink * asink,
|
|
gpointer data, guint length);
|
|
static guint gst_wasapi_sink_delay (GstAudioSink * asink);
|
|
static void gst_wasapi_sink_reset (GstAudioSink * asink);
|
|
|
|
G_DEFINE_TYPE (GstWasapiSink, gst_wasapi_sink, GST_TYPE_AUDIO_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);
|
|
GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_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 <ole.andre.ravnas@tandberg.com>");
|
|
|
|
gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_wasapi_sink_get_caps);
|
|
|
|
gstaudiosink_class->prepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_prepare);
|
|
gstaudiosink_class->unprepare = GST_DEBUG_FUNCPTR (gst_wasapi_sink_unprepare);
|
|
gstaudiosink_class->open = GST_DEBUG_FUNCPTR (gst_wasapi_sink_open);
|
|
gstaudiosink_class->close = GST_DEBUG_FUNCPTR (gst_wasapi_sink_close);
|
|
gstaudiosink_class->write = GST_DEBUG_FUNCPTR (gst_wasapi_sink_write);
|
|
gstaudiosink_class->delay = GST_DEBUG_FUNCPTR (gst_wasapi_sink_delay);
|
|
gstaudiosink_class->reset = GST_DEBUG_FUNCPTR (gst_wasapi_sink_reset);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_wasapi_sink_debug, "wasapisink",
|
|
0, "Windows audio session API sink");
|
|
}
|
|
|
|
static void
|
|
gst_wasapi_sink_init (GstWasapiSink * self)
|
|
{
|
|
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 (gst_wasapi_sink_parent_class)->dispose (object);
|
|
}
|
|
|
|
static void
|
|
gst_wasapi_sink_finalize (GObject * object)
|
|
{
|
|
CoUninitialize ();
|
|
|
|
G_OBJECT_CLASS (gst_wasapi_sink_parent_class)->finalize (object);
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_wasapi_sink_get_caps (GstBaseSink * bsink, GstCaps * filter)
|
|
{
|
|
/* FIXME: Implement */
|
|
return NULL;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wasapi_sink_open (GstAudioSink * asink)
|
|
{
|
|
GstWasapiSink *self = GST_WASAPI_SINK (asink);
|
|
gboolean res = FALSE;
|
|
IAudioClient *client = NULL;
|
|
|
|
if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self), FALSE,
|
|
&client)) {
|
|
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
|
|
("Failed to get default device"));
|
|
goto beach;
|
|
}
|
|
|
|
self->client = client;
|
|
res = TRUE;
|
|
|
|
beach:
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wasapi_sink_close (GstAudioSink * asink)
|
|
{
|
|
GstWasapiSink *self = GST_WASAPI_SINK (asink);
|
|
|
|
if (self->client != NULL) {
|
|
IUnknown_Release (self->client);
|
|
self->client = NULL;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wasapi_sink_prepare (GstAudioSink * asink, GstAudioRingBufferSpec * spec)
|
|
{
|
|
GstWasapiSink *self = GST_WASAPI_SINK (asink);
|
|
gboolean res = FALSE;
|
|
HRESULT hr;
|
|
REFERENCE_TIME latency_rt, def_period, min_period;
|
|
WAVEFORMATEXTENSIBLE format;
|
|
IAudioRenderClient *render_client = NULL;
|
|
|
|
hr = IAudioClient_GetDevicePeriod (self->client, &def_period, &min_period);
|
|
if (hr != S_OK) {
|
|
GST_ERROR_OBJECT (self, "IAudioClient::GetDevicePeriod () failed");
|
|
goto beach;
|
|
}
|
|
|
|
gst_wasapi_util_audio_info_to_waveformatex (&spec->info, &format);
|
|
self->info = spec->info;
|
|
|
|
hr = IAudioClient_Initialize (self->client, AUDCLNT_SHAREMODE_SHARED,
|
|
AUDCLNT_STREAMFLAGS_EVENTCALLBACK,
|
|
spec->buffer_time / 100, 0, (WAVEFORMATEX *) & format, NULL);
|
|
if (hr != S_OK) {
|
|
GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL),
|
|
("IAudioClient::Initialize () failed: %s",
|
|
gst_wasapi_util_hresult_to_string (hr)));
|
|
goto beach;
|
|
}
|
|
|
|
hr = IAudioClient_GetStreamLatency (self->client, &latency_rt);
|
|
if (hr != S_OK) {
|
|
GST_ERROR_OBJECT (self, "IAudioClient::GetStreamLatency () failed");
|
|
goto beach;
|
|
}
|
|
|
|
GST_INFO_OBJECT (self, "default period: %d (%d ms), "
|
|
"minimum period: %d (%d ms), "
|
|
"latency: %d (%d ms)",
|
|
(guint32) def_period, (guint32) def_period / 10000,
|
|
(guint32) min_period, (guint32) min_period / 10000,
|
|
(guint32) latency_rt, (guint32) latency_rt / 10000);
|
|
|
|
/* FIXME: What to do with the latency? */
|
|
|
|
hr = IAudioClient_SetEventHandle (self->client, self->event_handle);
|
|
if (hr != S_OK) {
|
|
GST_ERROR_OBJECT (self, "IAudioClient::SetEventHandle () failed");
|
|
goto beach;
|
|
}
|
|
|
|
if (!gst_wasapi_util_get_render_client (GST_ELEMENT (self), self->client,
|
|
&render_client)) {
|
|
goto beach;
|
|
}
|
|
|
|
hr = IAudioClient_Start (self->client);
|
|
if (hr != S_OK) {
|
|
GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
|
|
goto beach;
|
|
}
|
|
|
|
self->render_client = render_client;
|
|
render_client = NULL;
|
|
|
|
res = TRUE;
|
|
|
|
beach:
|
|
if (render_client != NULL)
|
|
IUnknown_Release (render_client);
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_wasapi_sink_unprepare (GstAudioSink * asink)
|
|
{
|
|
GstWasapiSink *self = GST_WASAPI_SINK (asink);
|
|
|
|
if (self->client != NULL) {
|
|
IAudioClient_Stop (self->client);
|
|
}
|
|
|
|
if (self->render_client != NULL) {
|
|
IUnknown_Release (self->render_client);
|
|
self->render_client = NULL;
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gint
|
|
gst_wasapi_sink_write (GstAudioSink * asink, gpointer data, guint length)
|
|
{
|
|
GstWasapiSink *self = GST_WASAPI_SINK (asink);
|
|
HRESULT hr;
|
|
gint16 *dst = NULL;
|
|
guint nsamples;
|
|
|
|
nsamples = length / self->info.bpf;
|
|
|
|
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)));
|
|
length = 0;
|
|
goto beach;
|
|
}
|
|
|
|
memcpy (dst, data, length);
|
|
|
|
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));
|
|
length = 0;
|
|
goto beach;
|
|
}
|
|
|
|
beach:
|
|
|
|
return length;
|
|
}
|
|
|
|
static guint
|
|
gst_wasapi_sink_delay (GstAudioSink * asink)
|
|
{
|
|
/* FIXME: Implement */
|
|
return 0;
|
|
}
|
|
|
|
static void
|
|
gst_wasapi_sink_reset (GstAudioSink * asink)
|
|
{
|
|
GstWasapiSink *self = GST_WASAPI_SINK (asink);
|
|
HRESULT hr;
|
|
|
|
if (self->client) {
|
|
hr = IAudioClient_Stop (self->client);
|
|
if (hr != S_OK) {
|
|
GST_ERROR_OBJECT (self, "IAudioClient::Stop () failed: %s",
|
|
gst_wasapi_util_hresult_to_string (hr));
|
|
return;
|
|
}
|
|
|
|
hr = IAudioClient_Reset (self->client);
|
|
if (hr != S_OK) {
|
|
GST_ERROR_OBJECT (self, "IAudioClient::Reset () failed: %s",
|
|
gst_wasapi_util_hresult_to_string (hr));
|
|
return;
|
|
}
|
|
}
|
|
}
|