blob: 5b35b8e1e5d18e421f346cf4f4f65b9ec3a92d34 [file] [log] [blame]
/*
* 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_static_pad_template (gstelement_class, &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;
}
}
}