gstreamer/sys/wasapi/gstwasapisrc.c
Ole André Vadla Ravnås 69fad589ac sys/: New plugin for audio capture and playback using Windows Audio Session
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.
2008-09-30 11:19:10 +00:00

443 lines
12 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-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;
}