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sys/: New plugin for audio capture and playback using Windows Audio Session

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Original commit message from CVS:
* sys/Makefile.am:
* sys/wasapi/Makefile.am:
* sys/wasapi/gstwasapi.c:
* sys/wasapi/gstwasapisink.c:
* sys/wasapi/gstwasapisink.h:
* sys/wasapi/gstwasapisrc.c:
* sys/wasapi/gstwasapisrc.h:
* sys/wasapi/gstwasapiutil.c:
* sys/wasapi/gstwasapiutil.h:
New plugin for audio capture and playback using Windows Audio Session
API (WASAPI) available with Vista and newer (#520901).
Comes with hardcoded caps and obviously needs lots of love. Haven't
had time to work on this code since it was written, was initially just
a quick experiment to play around with this new API.
This commit is contained in:
Ole André Vadla Ravnås 2008-09-30 11:19:10 +00:00
parent 34a993ef50
commit 69fad589ac
10 changed files with 1222 additions and 1 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

18
ChangeLog
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@ -1,3 +1,21 @@
2008-09-30 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* sys/Makefile.am:
* sys/wasapi/Makefile.am:
* sys/wasapi/gstwasapi.c:
* sys/wasapi/gstwasapisink.c:
* sys/wasapi/gstwasapisink.h:
* sys/wasapi/gstwasapisrc.c:
* sys/wasapi/gstwasapisrc.h:
* sys/wasapi/gstwasapiutil.c:
* sys/wasapi/gstwasapiutil.h:
New plugin for audio capture and playback using Windows Audio Session
API (WASAPI) available with Vista and newer (#520901).
Comes with hardcoded caps and obviously needs lots of love. Haven't
had time to work on this code since it was written, was initially just
a quick experiment to play around with this new API.
2008-09-30 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com> 2008-09-30 Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
* sys/dshowdecwrapper/gstdshowaudiodec.cpp * sys/dshowdecwrapper/gstdshowaudiodec.cpp

2
sys/Makefile.am
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@ -61,5 +61,5 @@ endif
SUBDIRS = $(ACM_DIR) $(DVB_DIR) $(FBDEV_DIR) $(OSS4_DIR) $(QT_DIR) $(VCD_DIR) $(WININET_DIR) SUBDIRS = $(ACM_DIR) $(DVB_DIR) $(FBDEV_DIR) $(OSS4_DIR) $(QT_DIR) $(VCD_DIR) $(WININET_DIR)
DIST_SUBDIRS = acmenc dvb fbdev dshowdecwrapper dshowsrcwrapper dshowvideosink \ DIST_SUBDIRS = acmenc dvb fbdev dshowdecwrapper dshowsrcwrapper dshowvideosink \
oss4 qtwrapper vcd wininet winks winscreencap oss4 qtwrapper vcd wasapi wininet winks winscreencap

5
sys/wasapi/Makefile.am Normal file
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@ -0,0 +1,5 @@
EXTRA_DIST = \
gstwasapi.c \
gstwasapisrc.c gstwasapisrc.h \
gstwasapisink.c gstwasapisink.h \
gstwasapiutil.c gstwasapiutil.h

45
sys/wasapi/gstwasapi.c Normal file
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@ -0,0 +1,45 @@
/*
* 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.
*/
#include "gstwasapisrc.h"
#include "gstwasapisink.h"
#ifdef HAVE_CONFIG_H
# include <config.h>
#endif
static gboolean
plugin_init (GstPlugin * plugin)
{
gboolean ret;
ret = gst_element_register (plugin, "wasapisrc",
GST_RANK_NONE, GST_TYPE_WASAPI_SRC);
if (!ret)
return ret;
return gst_element_register (plugin, "wasapisink",
GST_RANK_NONE, GST_TYPE_WASAPI_SINK);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"wasapi",
"Windows audio session API plugin",
plugin_init, VERSION, "LGPL", "GStreamer", "http://gstreamer.net/")

267
sys/wasapi/gstwasapisink.c Normal file
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@ -0,0 +1,267 @@
/*
* 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;
}

67
sys/wasapi/gstwasapisink.h Normal file
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@ -0,0 +1,67 @@
/*
* 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.
*/
#ifndef __GST_WASAPI_SINK_H__
#define __GST_WASAPI_SINK_H__
#include "gstwasapiutil.h"
#include <gst/base/gstbasesink.h>
G_BEGIN_DECLS
#define GST_TYPE_WASAPI_SINK \
(gst_wasapi_sink_get_type ())
#define GST_WASAPI_SINK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_WASAPI_SINK, GstWasapiSink))
#define GST_WASAPI_SINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_WASAPI_SINK, GstWasapiSinkClass))
#define GST_IS_WASAPI_SINK(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_WASAPI_SINK))
#define GST_IS_WASAPI_SINK_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_WASAPI_SINK))
typedef struct _GstWasapiSink GstWasapiSink;
typedef struct _GstWasapiSinkClass GstWasapiSinkClass;
struct _GstWasapiSink
{
GstBaseSink base_sink;
guint rate;
GstClockTime buffer_time;
GstClockTime period_time;
GstClockTime latency;
IAudioClient * client;
IAudioRenderClient * render_client;
HANDLE event_handle;
};
struct _GstWasapiSinkClass
{
GstBaseSinkClass parent_class;
};
GType gst_wasapi_sink_get_type (void);
G_END_DECLS
#endif /* __GST_WASAPI_SINK_H__ */

443
sys/wasapi/gstwasapisrc.c Normal file
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@ -0,0 +1,443 @@
/*
* 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-wasapisrc
*
* Provides audio capture from the Windows Audio Session API available with
* Vista and newer.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-0.10 -v wasapisrc ! fakesink
* ]| Capture from the default audio device and render to fakesink.
* </refsect2>
*/
#include "gstwasapisrc.h"
#include <gst/audio/gstaudioclock.h>
GST_DEBUG_CATEGORY_STATIC (gst_wasapi_src_debug);
#define GST_CAT_DEFAULT gst_wasapi_src_debug
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
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_src_dispose (GObject * object);
static void gst_wasapi_src_finalize (GObject * object);
static GstClock *gst_wasapi_src_provide_clock (GstElement * element);
static gboolean gst_wasapi_src_start (GstBaseSrc * src);
static gboolean gst_wasapi_src_stop (GstBaseSrc * src);
static gboolean gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query);
static GstFlowReturn gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf);
static GstClockTime gst_wasapi_src_get_time (GstClock * clock,
gpointer user_data);
GST_BOILERPLATE (GstWasapiSrc, gst_wasapi_src, GstPushSrc, GST_TYPE_PUSH_SRC);
static void
gst_wasapi_src_base_init (gpointer gclass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (gclass);
static GstElementDetails element_details = {
"WasapiSrc",
"Source/Audio",
"Stream audio from 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 (&src_template));
gst_element_class_set_details (element_class, &element_details);
}
static void
gst_wasapi_src_class_init (GstWasapiSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
GstPushSrcClass *gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
gobject_class->dispose = gst_wasapi_src_dispose;
gobject_class->finalize = gst_wasapi_src_finalize;
gstelement_class->provide_clock = gst_wasapi_src_provide_clock;
gstbasesrc_class->start = gst_wasapi_src_start;
gstbasesrc_class->stop = gst_wasapi_src_stop;
gstbasesrc_class->query = gst_wasapi_src_query;
gstpushsrc_class->create = gst_wasapi_src_create;
GST_DEBUG_CATEGORY_INIT (gst_wasapi_src_debug, "wasapisrc",
0, "Windows audio session API source");
}
static void
gst_wasapi_src_init (GstWasapiSrc * self, GstWasapiSrcClass * gclass)
{
GstBaseSrc *basesrc = GST_BASE_SRC (self);
gst_base_src_set_format (basesrc, GST_FORMAT_TIME);
gst_base_src_set_live (basesrc, TRUE);
self->rate = 8000;
self->buffer_time = 20 * GST_MSECOND;
self->period_time = 20 * GST_MSECOND;
self->latency = GST_CLOCK_TIME_NONE;
self->samples_per_buffer = self->rate / (GST_SECOND / self->period_time);
self->start_time = GST_CLOCK_TIME_NONE;
self->next_time = GST_CLOCK_TIME_NONE;
self->clock = gst_audio_clock_new ("GstWasapiSrcClock",
gst_wasapi_src_get_time, self);
CoInitialize (NULL);
}
static void
gst_wasapi_src_dispose (GObject * object)
{
GstWasapiSrc *self = GST_WASAPI_SRC (object);
if (self->clock != NULL) {
gst_object_unref (self->clock);
self->clock = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_wasapi_src_finalize (GObject * object)
{
GstWasapiSrc *self = GST_WASAPI_SRC (object);
CoUninitialize ();
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstClock *
gst_wasapi_src_provide_clock (GstElement * element)
{
GstWasapiSrc *self = GST_WASAPI_SRC (element);
GstClock *clock;
GST_OBJECT_LOCK (self);
if (self->client_clock == NULL)
goto wrong_state;
clock = GST_CLOCK (gst_object_ref (self->clock));
GST_OBJECT_UNLOCK (self);
return clock;
/* ERRORS */
wrong_state:
{
GST_OBJECT_UNLOCK (self);
GST_DEBUG_OBJECT (self, "IAudioClock not acquired");
return NULL;
}
}
static gboolean
gst_wasapi_src_start (GstBaseSrc * src)
{
GstWasapiSrc *self = GST_WASAPI_SRC (src);
gboolean res = FALSE;
IAudioClient *client = NULL;
IAudioClock *client_clock = NULL;
guint64 client_clock_freq = 0;
IAudioCaptureClient *capture_client = NULL;
HRESULT hr;
if (!gst_wasapi_util_get_default_device_client (GST_ELEMENT (self),
TRUE, self->rate, self->buffer_time, self->period_time, 0, &client,
&self->latency))
goto beach;
hr = IAudioClient_GetService (client, &IID_IAudioClock, &client_clock);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService (IID_IAudioClock) "
"failed");
goto beach;
}
hr = IAudioClock_GetFrequency (client_clock, &client_clock_freq);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClock::GetFrequency () failed");
goto beach;
}
hr = IAudioClient_GetService (client, &IID_IAudioCaptureClient,
&capture_client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::GetService "
"(IID_IAudioCaptureClient) failed");
goto beach;
}
hr = IAudioClient_Start (client);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioClient::Start failed");
goto beach;
}
self->client = client;
self->client_clock = client_clock;
self->client_clock_freq = client_clock_freq;
self->capture_client = capture_client;
res = TRUE;
beach:
if (!res) {
if (capture_client != NULL)
IUnknown_Release (capture_client);
if (client_clock != NULL)
IUnknown_Release (client_clock);
if (client != NULL)
IUnknown_Release (client);
}
return res;
}
static gboolean
gst_wasapi_src_stop (GstBaseSrc * src)
{
GstWasapiSrc *self = GST_WASAPI_SRC (src);
if (self->client != NULL) {
IAudioClient_Stop (self->client);
}
if (self->capture_client != NULL) {
IUnknown_Release (self->capture_client);
self->capture_client = NULL;
}
if (self->client_clock != NULL) {
IUnknown_Release (self->client_clock);
self->client_clock = NULL;
}
if (self->client != NULL) {
IUnknown_Release (self->client);
self->client = NULL;
}
return TRUE;
}
static gboolean
gst_wasapi_src_query (GstBaseSrc * src, GstQuery * query)
{
GstWasapiSrc *self = GST_WASAPI_SRC (src);
gboolean ret = FALSE;
GST_DEBUG_OBJECT (self, "query for %s",
gst_query_type_get_name (GST_QUERY_TYPE (query)));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:{
GstClockTime min_latency, max_latency;
min_latency = self->latency + self->period_time;
max_latency = min_latency;
GST_DEBUG_OBJECT (self, "reporting latency of min %" GST_TIME_FORMAT
" max %" GST_TIME_FORMAT,
GST_TIME_ARGS (min_latency), GST_TIME_ARGS (max_latency));
gst_query_set_latency (query, TRUE, min_latency, max_latency);
ret = TRUE;
break;
}
default:
ret = GST_BASE_SRC_CLASS (parent_class)->query (src, query);
break;
}
return ret;
}
static GstFlowReturn
gst_wasapi_src_create (GstPushSrc * src, GstBuffer ** buf)
{
GstWasapiSrc *self = GST_WASAPI_SRC (src);
GstFlowReturn ret = GST_FLOW_OK;
GstClock *clock;
GstClockTime timestamp, duration = self->period_time;
HRESULT hr;
gint16 *samples = NULL;
guint32 nsamples_read = 0, nsamples;
DWORD flags = 0;
guint64 devpos;
GST_OBJECT_LOCK (self);
clock = GST_ELEMENT_CLOCK (self);
if (clock != NULL)
gst_object_ref (clock);
GST_OBJECT_UNLOCK (self);
if (clock != NULL && GST_CLOCK_TIME_IS_VALID (self->next_time)) {
GstClockID id;
id = gst_clock_new_single_shot_id (clock, self->next_time);
gst_clock_id_wait (id, NULL);
gst_clock_id_unref (id);
}
do {
hr = IAudioCaptureClient_GetBuffer (self->capture_client,
(BYTE **) & samples, &nsamples_read, &flags, &devpos, NULL);
}
while (hr == AUDCLNT_S_BUFFER_EMPTY);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioCaptureClient::GetBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
ret = GST_FLOW_ERROR;
goto beach;
}
if (flags != 0) {
GST_WARNING_OBJECT (self, "devpos %" G_GUINT64_FORMAT ": flags=0x%08x",
devpos, flags);
}
/* FIXME: Why do we get 1024 sometimes and not a multiple of
* samples_per_buffer? Shouldn't WASAPI provide a DISCONT
* flag if we read too slow?
*/
nsamples = nsamples_read;
g_assert (nsamples >= self->samples_per_buffer);
if (nsamples > self->samples_per_buffer) {
GST_WARNING_OBJECT (self,
"devpos %" G_GUINT64_FORMAT ": got %d samples, expected %d, clipping!",
devpos, nsamples, self->samples_per_buffer);
nsamples = self->samples_per_buffer;
}
if (clock == NULL || clock == self->clock) {
timestamp =
gst_util_uint64_scale (devpos, GST_SECOND, self->client_clock_freq);
} else {
GstClockTime base_time;
timestamp = gst_clock_get_time (clock);
base_time = GST_ELEMENT_CAST (self)->base_time;
if (timestamp > base_time)
timestamp -= base_time;
else
timestamp = 0;
if (timestamp > duration)
timestamp -= duration;
else
timestamp = 0;
}
ret = gst_pad_alloc_buffer_and_set_caps (GST_BASE_SRC_PAD (self),
devpos,
nsamples * sizeof (gint16), GST_PAD_CAPS (GST_BASE_SRC_PAD (self)), buf);
if (ret == GST_FLOW_OK) {
guint i;
gint16 *dst;
GST_BUFFER_OFFSET_END (*buf) = devpos + self->samples_per_buffer;
GST_BUFFER_TIMESTAMP (*buf) = timestamp;
GST_BUFFER_DURATION (*buf) = duration;
dst = (gint16 *) GST_BUFFER_DATA (*buf);
for (i = 0; i < nsamples; i++) {
*dst = *samples;
samples += 2;
dst++;
}
}
hr = IAudioCaptureClient_ReleaseBuffer (self->capture_client, nsamples_read);
if (hr != S_OK) {
GST_ERROR_OBJECT (self, "IAudioCaptureClient::ReleaseBuffer () failed: %s",
gst_wasapi_util_hresult_to_string (hr));
ret = GST_FLOW_ERROR;
goto beach;
}
beach:
if (clock != NULL)
gst_object_unref (clock);
return ret;
}
static GstClockTime
gst_wasapi_src_get_time (GstClock * clock, gpointer user_data)
{
GstWasapiSrc *self = GST_WASAPI_SRC (user_data);
HRESULT hr;
guint64 devpos;
GstClockTime result;
if (G_UNLIKELY (self->client_clock == NULL))
return GST_CLOCK_TIME_NONE;
hr = IAudioClock_GetPosition (self->client_clock, &devpos, NULL);
if (G_UNLIKELY (hr != S_OK))
return GST_CLOCK_TIME_NONE;
result = gst_util_uint64_scale_int (devpos, GST_SECOND,
self->client_clock_freq);
/*
GST_DEBUG_OBJECT (self, "devpos = %" G_GUINT64_FORMAT
" frequency = %" G_GUINT64_FORMAT
" result = %" G_GUINT64_FORMAT " ms",
devpos, self->client_clock_freq, GST_TIME_AS_MSECONDS (result));
*/
return result;
}

74
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@ -0,0 +1,74 @@
/*
* 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.
*/
#ifndef __GST_WASAPI_SRC_H__
#define __GST_WASAPI_SRC_H__
#include "gstwasapiutil.h"
#include <gst/base/gstpushsrc.h>
G_BEGIN_DECLS
#define GST_TYPE_WASAPI_SRC \
(gst_wasapi_src_get_type ())
#define GST_WASAPI_SRC(obj) \
(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_WASAPI_SRC, GstWasapiSrc))
#define GST_WASAPI_SRC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST ((klass), GST_TYPE_WASAPI_SRC, GstWasapiSrcClass))
#define GST_IS_WASAPI_SRC(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE ((obj), GST_TYPE_WASAPI_SRC))
#define GST_IS_WASAPI_SRC_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE ((klass), GST_TYPE_WASAPI_SRC))
typedef struct _GstWasapiSrc GstWasapiSrc;
typedef struct _GstWasapiSrcClass GstWasapiSrcClass;
struct _GstWasapiSrc
{
GstPushSrc audio_src;
GstClock * clock;
guint rate;
GstClockTime buffer_time;
GstClockTime period_time;
GstClockTime latency;
guint samples_per_buffer;
IAudioClient * client;
IAudioClock * client_clock;
guint64 client_clock_freq;
IAudioCaptureClient * capture_client;
GstClockTime start_time;
GstClockTime next_time;
};
struct _GstWasapiSrcClass
{
GstPushSrcClass parent_class;
};
GType gst_wasapi_src_get_type (void);
G_END_DECLS
#endif /* __GST_WASAPI_SRC_H__ */

261
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@ -0,0 +1,261 @@
/*
* 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.
*/
#include "gstwasapiutil.h"
#include <mmdeviceapi.h>
/* These seem to be missing in the Windows SDK... */
const CLSID CLSID_MMDeviceEnumerator = { 0xbcde0395, 0xe52f, 0x467c,
{0x8e, 0x3d, 0xc4, 0x57, 0x92, 0x91, 0x69, 0x2e}
};
const IID IID_IMMDeviceEnumerator = { 0xa95664d2, 0x9614, 0x4f35,
{0xa7, 0x46, 0xde, 0x8d, 0xb6, 0x36, 0x17, 0xe6}
};
const IID IID_IAudioClient = { 0x1cb9ad4c, 0xdbfa, 0x4c32,
{0xb1, 0x78, 0xc2, 0xf5, 0x68, 0xa7, 0x03, 0xb2}
};
const IID IID_IAudioClock = { 0xcd63314f, 0x3fba, 0x4a1b,
{0x81, 0x2c, 0xef, 0x96, 0x35, 0x87, 0x28, 0xe7}
};
const IID IID_IAudioCaptureClient = { 0xc8adbd64, 0xe71e, 0x48a0,
{0xa4, 0xde, 0x18, 0x5c, 0x39, 0x5c, 0xd3, 0x17}
};
const IID IID_IAudioRenderClient = { 0xf294acfc, 0x3146, 0x4483,
{0xa7, 0xbf, 0xad, 0xdc, 0xa7, 0xc2, 0x60, 0xe2}
};
const gchar *
gst_wasapi_util_hresult_to_string (HRESULT hr)
{
const gchar *s = "AUDCLNT_E_UNKNOWN";
switch (hr) {
case AUDCLNT_E_NOT_INITIALIZED:
s = "AUDCLNT_E_NOT_INITIALIZED";
break;
case AUDCLNT_E_ALREADY_INITIALIZED:
s = "AUDCLNT_E_ALREADY_INITIALIZED";
break;
case AUDCLNT_E_WRONG_ENDPOINT_TYPE:
s = "AUDCLNT_E_WRONG_ENDPOINT_TYPE";
break;
case AUDCLNT_E_DEVICE_INVALIDATED:
s = "AUDCLNT_E_DEVICE_INVALIDATED";
break;
case AUDCLNT_E_NOT_STOPPED:
s = "AUDCLNT_E_NOT_STOPPED";
break;
case AUDCLNT_E_BUFFER_TOO_LARGE:
s = "AUDCLNT_E_BUFFER_TOO_LARGE";
break;
case AUDCLNT_E_OUT_OF_ORDER:
s = "AUDCLNT_E_OUT_OF_ORDER";
break;
case AUDCLNT_E_UNSUPPORTED_FORMAT:
s = "AUDCLNT_E_UNSUPPORTED_FORMAT";
break;
case AUDCLNT_E_INVALID_SIZE:
s = "AUDCLNT_E_INVALID_SIZE";
break;
case AUDCLNT_E_DEVICE_IN_USE:
s = "AUDCLNT_E_DEVICE_IN_USE";
break;
case AUDCLNT_E_BUFFER_OPERATION_PENDING:
s = "AUDCLNT_E_BUFFER_OPERATION_PENDING";
break;
case AUDCLNT_E_THREAD_NOT_REGISTERED:
s = "AUDCLNT_E_THREAD_NOT_REGISTERED";
break;
case AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED:
s = "AUDCLNT_E_EXCLUSIVE_MODE_NOT_ALLOWED";
break;
case AUDCLNT_E_ENDPOINT_CREATE_FAILED:
s = "AUDCLNT_E_ENDPOINT_CREATE_FAILED";
break;
case AUDCLNT_E_SERVICE_NOT_RUNNING:
s = "AUDCLNT_E_SERVICE_NOT_RUNNING";
break;
case AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED:
s = "AUDCLNT_E_EVENTHANDLE_NOT_EXPECTED";
break;
case AUDCLNT_E_EXCLUSIVE_MODE_ONLY:
s = "AUDCLNT_E_EXCLUSIVE_MODE_ONLY";
break;
case AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL:
s = "AUDCLNT_E_BUFDURATION_PERIOD_NOT_EQUAL";
break;
case AUDCLNT_E_EVENTHANDLE_NOT_SET:
s = "AUDCLNT_E_EVENTHANDLE_NOT_SET";
break;
case AUDCLNT_E_INCORRECT_BUFFER_SIZE:
s = "AUDCLNT_E_INCORRECT_BUFFER_SIZE";
break;
case AUDCLNT_E_BUFFER_SIZE_ERROR:
s = "AUDCLNT_E_BUFFER_SIZE_ERROR";
break;
case AUDCLNT_E_CPUUSAGE_EXCEEDED:
s = "AUDCLNT_E_CPUUSAGE_EXCEEDED";
break;
case AUDCLNT_S_BUFFER_EMPTY:
s = "AUDCLNT_S_BUFFER_EMPTY";
break;
case AUDCLNT_S_THREAD_ALREADY_REGISTERED:
s = "AUDCLNT_S_THREAD_ALREADY_REGISTERED";
break;
case AUDCLNT_S_POSITION_STALLED:
s = "AUDCLNT_S_POSITION_STALLED";
break;
}
return s;
}
gboolean
gst_wasapi_util_get_default_device_client (GstElement * element,
gboolean capture,
guint rate,
GstClockTime buffer_time,
GstClockTime period_time,
DWORD flags, IAudioClient ** ret_client, GstClockTime * ret_latency)
{
gboolean res = FALSE;
HRESULT hr;
IMMDeviceEnumerator *enumerator = NULL;
IMMDevice *device = NULL;
IAudioClient *client = NULL;
REFERENCE_TIME latency_rt, def_period, min_period;
WAVEFORMATEXTENSIBLE format;
hr = CoCreateInstance (&CLSID_MMDeviceEnumerator, NULL, CLSCTX_ALL,
&IID_IMMDeviceEnumerator, &enumerator);
if (hr != S_OK) {
GST_ERROR_OBJECT (element, "CoCreateInstance (MMDeviceEnumerator) failed");
goto beach;
}
hr = IMMDeviceEnumerator_GetDefaultAudioEndpoint (enumerator,
(capture) ? eCapture : eRender, eCommunications, &device);
if (hr != S_OK) {
GST_ERROR_OBJECT (element,
"IMMDeviceEnumerator::GetDefaultAudioEndpoint () failed");
goto beach;
}
hr = IMMDevice_Activate (device, &IID_IAudioClient, CLSCTX_ALL, NULL,
&client);
if (hr != S_OK) {
GST_ERROR_OBJECT (element, "IMMDevice::Activate (IID_IAudioClient) failed");
goto beach;
}
hr = IAudioClient_GetDevicePeriod (client, &def_period, &min_period);
if (hr != S_OK) {
GST_ERROR_OBJECT (element, "IAudioClient::GetDevicePeriod () failed");
goto beach;
}
ZeroMemory (&format, sizeof (format));
format.Format.cbSize = sizeof (format) - sizeof (format.Format);
format.Format.wFormatTag = WAVE_FORMAT_EXTENSIBLE;
format.Format.nChannels = 2;
format.Format.nSamplesPerSec = rate;
format.Format.wBitsPerSample = 16;
format.Format.nBlockAlign = format.Format.nChannels
* (format.Format.wBitsPerSample / 8);
format.Format.nAvgBytesPerSec = format.Format.nSamplesPerSec
* format.Format.nBlockAlign;
format.Samples.wValidBitsPerSample = 16;
format.dwChannelMask = SPEAKER_FRONT_LEFT | SPEAKER_FRONT_RIGHT;
format.SubFormat = KSDATAFORMAT_SUBTYPE_PCM;
hr = IAudioClient_Initialize (client, AUDCLNT_SHAREMODE_EXCLUSIVE, /* or AUDCLNT_SHAREMODE_SHARED */
flags, buffer_time / 100, /* buffer duration in 100s of ns */
period_time / 100, /* periodicity in 100s of ns */
(WAVEFORMATEX *) & format, NULL);
if (hr != S_OK) {
GST_ELEMENT_ERROR (element, RESOURCE, OPEN_READ, (NULL),
("IAudioClient::Initialize () failed: %s",
gst_wasapi_util_hresult_to_string (hr)));
goto beach;
}
hr = IAudioClient_GetStreamLatency (client, &latency_rt);
if (hr != S_OK) {
GST_ERROR_OBJECT (element, "IAudioClient::GetStreamLatency () failed");
goto beach;
}
GST_INFO_OBJECT (element, "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);
IUnknown_AddRef (client);
*ret_client = client;
*ret_latency = latency_rt * 100;
res = TRUE;
beach:
if (client != NULL)
IUnknown_Release (client);
if (device != NULL)
IUnknown_Release (device);
if (enumerator != NULL)
IUnknown_Release (enumerator);
return res;
}
#if 0
static WAVEFORMATEXTENSIBLE *
gst_wasapi_src_probe_device_format (GstWasapiSrc * self, IMMDevice * device)
{
HRESULT hr;
IPropertyStore *props = NULL;
PROPVARIANT format_prop;
WAVEFORMATEXTENSIBLE *format = NULL;
hr = IMMDevice_OpenPropertyStore (device, STGM_READ, &props);
if (hr != S_OK)
goto beach;
PropVariantInit (&format_prop);
hr = IPropertyStore_GetValue (props, &PKEY_AudioEngine_DeviceFormat,
&format_prop);
if (hr != S_OK)
goto beach;
format = (WAVEFORMATEXTENSIBLE *) format_prop.blob.pBlobData;
/* hmm: HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\CurrentVersion\MMDevices\Audio\Capture\{64adb8b7-9716-4c02-8929-96e53f5642da}\Properties */
beach:
if (props != NULL)
IUnknown_Release (props);
return format;
}
#endif

41
sys/wasapi/gstwasapiutil.h Normal file
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@ -0,0 +1,41 @@
/*
* 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.
*/
#ifndef __GST_WASAPI_UTIL_H__
#define __GST_WASAPI_UTIL_H__
#include <gst/gst.h>
#include <audioclient.h>
const gchar *
gst_wasapi_util_hresult_to_string (HRESULT hr);
gboolean
gst_wasapi_util_get_default_device_client (GstElement * element,
gboolean capture,
guint rate,
GstClockTime buffer_time,
GstClockTime period_time,
DWORD flags,
IAudioClient ** ret_client,
GstClockTime * ret_latency);
#endif /* __GST_WASAPI_UTIL_H__ */