gstreamer/sys/dshowsrcwrapper/gstdshowaudiosrc.cpp
Stéphane Cerveau add7878e14 bad: use of g_value_dup_string
Use helper method to get string from GValue.
2019-12-30 14:13:03 +00:00

806 lines
22 KiB
C++

/* GStreamer
* Copyright (C) 2007 Sebastien Moutte <sebastien@moutte.net>
*
* gstdshowaudiosrc.c:
*
* 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.
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstdshowaudiosrc.h"
GST_DEBUG_CATEGORY_STATIC (dshowaudiosrc_debug);
#define GST_CAT_DEFAULT dshowaudiosrc_debug
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string){ "
GST_AUDIO_NE (S16) ", "
GST_AUDIO_NE (U16) ", "
GST_AUDIO_NE (S8) ", "
GST_AUDIO_NE (U8)
" }, "
"rate = " GST_AUDIO_RATE_RANGE ", "
"channels = (int) [ 1, 2 ]")
);
G_DEFINE_TYPE(GstDshowAudioSrc, gst_dshowaudiosrc, GST_TYPE_AUDIO_SRC);
enum
{
PROP_0,
PROP_DEVICE,
PROP_DEVICE_NAME,
PROP_DEVICE_INDEX
};
#define DEFAULT_PROP_DEVICE_INDEX 0
static void gst_dshowaudiosrc_dispose (GObject * gobject);
static void gst_dshowaudiosrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_dshowaudiosrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstCaps *gst_dshowaudiosrc_get_caps (GstBaseSrc * src, GstCaps * filter);
static GstStateChangeReturn gst_dshowaudiosrc_change_state (GstElement *
element, GstStateChange transition);
static gboolean gst_dshowaudiosrc_open (GstAudioSrc * asrc);
static gboolean gst_dshowaudiosrc_prepare (GstAudioSrc * asrc,
GstAudioRingBufferSpec * spec);
static gboolean gst_dshowaudiosrc_unprepare (GstAudioSrc * asrc);
static gboolean gst_dshowaudiosrc_close (GstAudioSrc * asrc);
static guint gst_dshowaudiosrc_read (GstAudioSrc * asrc, gpointer data,
guint length, GstClockTime *timestamp);
static guint gst_dshowaudiosrc_delay (GstAudioSrc * asrc);
static void gst_dshowaudiosrc_reset (GstAudioSrc * asrc);
/* utils */
static GstCaps *gst_dshowaudiosrc_getcaps_from_streamcaps (GstDshowAudioSrc *
src, IPin * pin, IAMStreamConfig * streamcaps);
static gboolean gst_dshowaudiosrc_push_buffer (guint8 * buffer, guint size,
gpointer src_object, GstClockTime duration);
static void
gst_dshowaudiosrc_class_init (GstDshowAudioSrcClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
GstBaseSrcClass *gstbasesrc_class;
GstAudioSrcClass *gstaudiosrc_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
gstbasesrc_class = (GstBaseSrcClass *) klass;
gstaudiosrc_class = (GstAudioSrcClass *) klass;
gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_dispose);
gobject_class->set_property =
GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_set_property);
gobject_class->get_property =
GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_get_property);
gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_get_caps);
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_change_state);
gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_open);
gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_prepare);
gstaudiosrc_class->unprepare =
GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_unprepare);
gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_close);
gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_read);
gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_delay);
gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_dshowaudiosrc_reset);
g_object_class_install_property
(gobject_class, PROP_DEVICE,
g_param_spec_string ("device", "Device",
"Directshow device reference (classID/name)", NULL,
static_cast < GParamFlags > (G_PARAM_READWRITE)));
g_object_class_install_property
(gobject_class, PROP_DEVICE_NAME,
g_param_spec_string ("device-name", "Device name",
"Human-readable name of the sound device", NULL,
static_cast < GParamFlags > (G_PARAM_READWRITE)));
g_object_class_install_property
(gobject_class, PROP_DEVICE_INDEX,
g_param_spec_int ("device-index", "Device index",
"Index of the enumerated audio device", 0, G_MAXINT,
DEFAULT_PROP_DEVICE_INDEX,
static_cast < GParamFlags > (G_PARAM_READWRITE)));
gst_element_class_add_static_pad_template (gstelement_class, &src_template);
gst_element_class_set_static_metadata (gstelement_class,
"Directshow audio capture source", "Source/Audio",
"Receive data from a directshow audio capture graph",
"Sebastien Moutte <sebastien@moutte.net>");
GST_DEBUG_CATEGORY_INIT (dshowaudiosrc_debug, "dshowaudiosrc", 0,
"Directshow audio source");
}
static void
gst_dshowaudiosrc_init (GstDshowAudioSrc * src)
{
src->device = NULL;
src->device_name = NULL;
src->device_index = DEFAULT_PROP_DEVICE_INDEX;
src->audio_cap_filter = NULL;
src->dshow_fakesink = NULL;
src->media_filter = NULL;
src->filter_graph = NULL;
src->caps = NULL;
src->pins_mediatypes = NULL;
src->gbarray = g_byte_array_new ();
g_mutex_init(&src->gbarray_lock);
src->is_running = FALSE;
CoInitializeEx (NULL, COINIT_MULTITHREADED);
}
static void
gst_dshowaudiosrc_dispose (GObject * gobject)
{
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (gobject);
if (src->device) {
g_free (src->device);
src->device = NULL;
}
if (src->device_name) {
g_free (src->device_name);
src->device_name = NULL;
}
if (src->caps) {
gst_caps_unref (src->caps);
src->caps = NULL;
}
if (src->pins_mediatypes) {
gst_dshow_free_pins_mediatypes (src->pins_mediatypes);
src->pins_mediatypes = NULL;
}
if (src->gbarray) {
g_byte_array_free (src->gbarray, TRUE);
src->gbarray = NULL;
}
g_mutex_clear(&src->gbarray_lock);
/* clean dshow */
if (src->audio_cap_filter)
src->audio_cap_filter->Release ();
CoUninitialize ();
G_OBJECT_CLASS (gst_dshowaudiosrc_parent_class)->dispose (gobject);
}
static void
gst_dshowaudiosrc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (object);
switch (prop_id) {
case PROP_DEVICE:
{
if (src->device) {
g_free (src->device);
src->device = NULL;
}
if (g_value_get_string (value)) {
src->device = g_value_dup_string (value);;
}
break;
}
case PROP_DEVICE_NAME:
{
if (src->device_name) {
g_free (src->device_name);
src->device_name = NULL;
}
if (g_value_get_string (value)) {
src->device_name = g_value_dup_string (value);;
}
break;
}
case PROP_DEVICE_INDEX:
{
src->device_index = g_value_get_int (value);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_dshowaudiosrc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstDshowAudioSrc *src;
g_return_if_fail (GST_IS_DSHOWAUDIOSRC (object));
src = GST_DSHOWAUDIOSRC (object);
switch (prop_id) {
case PROP_DEVICE:
g_value_set_string (value, src->device);
break;
case PROP_DEVICE_NAME:
g_value_set_string (value, src->device_name);
break;
case PROP_DEVICE_INDEX:
g_value_set_int (value, src->device_index);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstCaps *
gst_dshowaudiosrc_get_caps (GstBaseSrc * basesrc, GstCaps * filter)
{
HRESULT hres = S_OK;
IBindCtx *lpbc = NULL;
IMoniker *audiom = NULL;
DWORD dwEaten;
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (basesrc);
gunichar2 *unidevice = NULL;
if (src->device) {
g_free (src->device);
src->device = NULL;
}
src->device =
gst_dshow_getdevice_from_devicename (&CLSID_AudioInputDeviceCategory,
&src->device_name, &src->device_index);
if (!src->device) {
GST_ERROR ("No audio device found.");
return NULL;
}
unidevice =
g_utf8_to_utf16 (src->device, strlen (src->device), NULL, NULL, NULL);
if (!src->audio_cap_filter) {
hres = CreateBindCtx (0, &lpbc);
if (SUCCEEDED (hres)) {
hres =
MkParseDisplayName (lpbc, (LPCOLESTR) unidevice, &dwEaten, &audiom);
if (SUCCEEDED (hres)) {
hres = audiom->BindToObject (lpbc, NULL, IID_IBaseFilter,
(LPVOID *) & src->audio_cap_filter);
audiom->Release ();
}
lpbc->Release ();
}
}
if (src->audio_cap_filter && !src->caps) {
/* get the capture pins supported types */
IPin *capture_pin = NULL;
IEnumPins *enumpins = NULL;
HRESULT hres;
hres = src->audio_cap_filter->EnumPins (&enumpins);
if (SUCCEEDED (hres)) {
while (enumpins->Next (1, &capture_pin, NULL) == S_OK) {
IKsPropertySet *pKs = NULL;
hres =
capture_pin->QueryInterface (IID_IKsPropertySet, (LPVOID *) & pKs);
if (SUCCEEDED (hres) && pKs) {
DWORD cbReturned;
GUID pin_category;
RPC_STATUS rpcstatus;
hres =
pKs->Get (AMPROPSETID_Pin,
AMPROPERTY_PIN_CATEGORY, NULL, 0, &pin_category, sizeof (GUID),
&cbReturned);
/* we only want capture pins */
if (UuidCompare (&pin_category, (UUID *) & PIN_CATEGORY_CAPTURE,
&rpcstatus) == 0) {
IAMStreamConfig *streamcaps = NULL;
if (SUCCEEDED (capture_pin->QueryInterface (IID_IAMStreamConfig,
(LPVOID *) & streamcaps))) {
src->caps =
gst_dshowaudiosrc_getcaps_from_streamcaps (src, capture_pin,
streamcaps);
streamcaps->Release ();
}
}
pKs->Release ();
}
capture_pin->Release ();
}
enumpins->Release ();
}
}
if (unidevice) {
g_free (unidevice);
}
if (src->caps) {
GstCaps *caps;
if (filter) {
caps = gst_caps_intersect_full (filter, src->caps, GST_CAPS_INTERSECT_FIRST);
} else {
caps = gst_caps_ref (src->caps);
}
return caps;
}
return NULL;
}
static GstStateChangeReturn
gst_dshowaudiosrc_change_state (GstElement * element, GstStateChange transition)
{
HRESULT hres = S_FALSE;
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
if (src->media_filter) {
src->is_running = TRUE;
hres = src->media_filter->Run (0);
}
if (hres != S_OK) {
GST_ERROR ("Can't RUN the directshow capture graph (error=0x%x)", hres);
src->is_running = FALSE;
return GST_STATE_CHANGE_FAILURE;
}
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
if (src->media_filter)
hres = src->media_filter->Stop ();
if (hres != S_OK) {
GST_ERROR ("Can't STOP the directshow capture graph (error=0x%x)",
hres);
return GST_STATE_CHANGE_FAILURE;
}
src->is_running = FALSE;
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return GST_ELEMENT_CLASS(gst_dshowaudiosrc_parent_class)->change_state(element, transition);
}
static gboolean
gst_dshowaudiosrc_open (GstAudioSrc * asrc)
{
HRESULT hres = S_FALSE;
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
hres = CoCreateInstance (CLSID_FilterGraph, NULL, CLSCTX_INPROC,
IID_IFilterGraph, (LPVOID *) & src->filter_graph);
if (hres != S_OK || !src->filter_graph) {
GST_ERROR
("Can't create an instance of the directshow graph manager (error=0x%x)",
hres);
goto error;
}
hres =
src->filter_graph->QueryInterface (IID_IMediaFilter,
(LPVOID *) & src->media_filter);
if (hres != S_OK || !src->media_filter) {
GST_ERROR
("Can't get IMediacontrol interface from the graph manager (error=0x%x)",
hres);
goto error;
}
src->dshow_fakesink = new CDshowFakeSink;
src->dshow_fakesink->AddRef ();
hres = src->filter_graph->AddFilter (src->audio_cap_filter, L"capture");
if (hres != S_OK) {
GST_ERROR
("Can't add the directshow capture filter to the graph (error=0x%x)",
hres);
goto error;
}
hres = src->filter_graph->AddFilter (src->dshow_fakesink, L"fakesink");
if (hres != S_OK) {
GST_ERROR ("Can't add our fakesink filter to the graph (error=0x%x)", hres);
goto error;
}
return TRUE;
error:
if (src->dshow_fakesink) {
src->dshow_fakesink->Release ();
src->dshow_fakesink = NULL;
}
if (src->media_filter) {
src->media_filter->Release ();
src->media_filter = NULL;
}
if (src->filter_graph) {
src->filter_graph->Release ();
src->filter_graph = NULL;
}
return FALSE;
}
static gboolean
gst_dshowaudiosrc_prepare (GstAudioSrc * asrc, GstAudioRingBufferSpec * spec)
{
HRESULT hres;
IPin *input_pin = NULL;
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
GstCaps *current_caps = gst_pad_get_current_caps (GST_BASE_SRC_PAD (asrc));
if (current_caps) {
if (gst_caps_is_equal (spec->caps, current_caps)) {
gst_caps_unref (current_caps);
return TRUE;
}
gst_caps_unref (current_caps);
}
/* In 1.0, prepare() seems to be called in the PLAYING state. Most
of the time you can't do much on a running graph. */
gboolean was_running = src->is_running;
if (was_running) {
HRESULT hres = src->media_filter->Stop ();
if (hres != S_OK) {
GST_ERROR("Can't STOP the directshow capture graph for preparing (error=0x%x)", hres);
return FALSE;
}
src->is_running = FALSE;
}
/* search the negotiated caps in our caps list to get its index and the corresponding mediatype */
if (gst_caps_is_subset (spec->caps, src->caps)) {
guint i = 0;
gint res = -1;
for (; i < gst_caps_get_size (src->caps) && res == -1; i++) {
GstCaps *capstmp = gst_caps_copy_nth (src->caps, i);
if (gst_caps_is_subset (spec->caps, capstmp)) {
res = i;
}
gst_caps_unref (capstmp);
}
if (res != -1 && src->pins_mediatypes) {
/*get the corresponding media type and build the dshow graph */
GstCapturePinMediaType *pin_mediatype = NULL;
GList *type = g_list_nth (src->pins_mediatypes, res);
if (type) {
pin_mediatype = (GstCapturePinMediaType *) type->data;
src->dshow_fakesink->gst_set_media_type (pin_mediatype->mediatype);
src->dshow_fakesink->gst_set_buffer_callback (
(push_buffer_func) gst_dshowaudiosrc_push_buffer, src);
gst_dshow_get_pin_from_filter (src->dshow_fakesink, PINDIR_INPUT,
&input_pin);
if (!input_pin) {
GST_ERROR ("Can't get input pin from our directshow fakesink filter");
goto error;
}
spec->segsize = (gint) (spec->info.bpf * spec->info.rate * spec->latency_time /
GST_MSECOND);
spec->segtotal = (gint) ((gfloat) spec->buffer_time /
(gfloat) spec->latency_time + 0.5);
if (!gst_dshow_configure_latency (pin_mediatype->capture_pin,
spec->segsize))
{
GST_WARNING ("Could not change capture latency");
spec->segsize = spec->info.rate * spec->info.channels;
spec->segtotal = 2;
};
GST_INFO ("Configuring with segsize:%d segtotal:%d", spec->segsize, spec->segtotal);
if (gst_dshow_is_pin_connected (pin_mediatype->capture_pin)) {
GST_DEBUG_OBJECT (src,
"capture_pin already connected, disconnecting");
src->filter_graph->Disconnect (pin_mediatype->capture_pin);
}
if (gst_dshow_is_pin_connected (input_pin)) {
GST_DEBUG_OBJECT (src, "input_pin already connected, disconnecting");
src->filter_graph->Disconnect (input_pin);
}
hres = src->filter_graph->ConnectDirect (pin_mediatype->capture_pin,
input_pin, NULL);
input_pin->Release ();
if (hres != S_OK) {
GST_ERROR
("Can't connect capture filter with fakesink filter (error=0x%x)",
hres);
goto error;
}
}
}
}
if (was_running) {
HRESULT hres = src->media_filter->Run (0);
if (hres != S_OK) {
GST_ERROR("Can't RUN the directshow capture graph after prepare (error=0x%x)", hres);
return FALSE;
}
src->is_running = TRUE;
}
return TRUE;
error:
/* Don't restart the graph, we're out anyway. */
return FALSE;
}
static gboolean
gst_dshowaudiosrc_unprepare (GstAudioSrc * asrc)
{
IPin *input_pin = NULL, *output_pin = NULL;
HRESULT hres = S_FALSE;
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
/* disconnect filters */
gst_dshow_get_pin_from_filter (src->audio_cap_filter, PINDIR_OUTPUT,
&output_pin);
if (output_pin) {
hres = src->filter_graph->Disconnect (output_pin);
output_pin->Release ();
}
gst_dshow_get_pin_from_filter (src->dshow_fakesink, PINDIR_INPUT, &input_pin);
if (input_pin) {
hres = src->filter_graph->Disconnect (input_pin);
input_pin->Release ();
}
return TRUE;
}
static gboolean
gst_dshowaudiosrc_close (GstAudioSrc * asrc)
{
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
if (!src->filter_graph)
return TRUE;
/*remove filters from the graph */
src->filter_graph->RemoveFilter (src->audio_cap_filter);
src->filter_graph->RemoveFilter (src->dshow_fakesink);
/*release our gstreamer dshow sink */
src->dshow_fakesink->Release ();
src->dshow_fakesink = NULL;
/*release media filter interface */
src->media_filter->Release ();
src->media_filter = NULL;
/*release the filter graph manager */
src->filter_graph->Release ();
src->filter_graph = NULL;
return TRUE;
}
static guint
gst_dshowaudiosrc_read (GstAudioSrc * asrc, gpointer data, guint length, GstClockTime *timestamp)
{
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
guint ret = 0;
if (!src->is_running)
return -1;
if (src->gbarray) {
test:
if (src->gbarray->len >= length) {
g_mutex_lock (&src->gbarray_lock);
memcpy (data, src->gbarray->data + (src->gbarray->len - length), length);
g_byte_array_remove_range (src->gbarray, src->gbarray->len - length,
length);
ret = length;
g_mutex_unlock (&src->gbarray_lock);
} else {
if (src->is_running) {
Sleep (GST_AUDIO_BASE_SRC(src)->ringbuffer->spec.latency_time /
GST_MSECOND / 10);
goto test;
}
}
}
return ret;
}
static guint
gst_dshowaudiosrc_delay (GstAudioSrc * asrc)
{
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
guint ret = 0;
if (src->gbarray) {
g_mutex_lock (&src->gbarray_lock);
if (src->gbarray->len) {
ret = src->gbarray->len / 4;
}
g_mutex_unlock (&src->gbarray_lock);
}
return ret;
}
static void
gst_dshowaudiosrc_reset (GstAudioSrc * asrc)
{
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (asrc);
g_mutex_lock (&src->gbarray_lock);
GST_DEBUG ("byte array size= %d", src->gbarray->len);
if (src->gbarray->len > 0)
g_byte_array_remove_range (src->gbarray, 0, src->gbarray->len);
g_mutex_unlock (&src->gbarray_lock);
}
static GstCaps *
gst_dshowaudiosrc_getcaps_from_streamcaps (GstDshowAudioSrc * src, IPin * pin,
IAMStreamConfig * streamcaps)
{
GstCaps *caps = NULL;
HRESULT hres = S_OK;
int icount = 0;
int isize = 0;
AUDIO_STREAM_CONFIG_CAPS ascc;
int i = 0;
if (!streamcaps)
return NULL;
streamcaps->GetNumberOfCapabilities (&icount, &isize);
if (isize != sizeof (ascc))
return NULL;
for (; i < icount; i++) {
GstCapturePinMediaType *pin_mediatype = g_new0 (GstCapturePinMediaType, 1);
pin->AddRef ();
pin_mediatype->capture_pin = pin;
hres = streamcaps->GetStreamCaps (i, &pin_mediatype->mediatype,
(BYTE *) & ascc);
if (hres == S_OK && pin_mediatype->mediatype) {
GstCaps *mediacaps = NULL;
if (!caps)
caps = gst_caps_new_empty ();
if (gst_dshow_check_mediatype (pin_mediatype->mediatype, MEDIASUBTYPE_PCM,
FORMAT_WaveFormatEx)) {
GstAudioFormat format = GST_AUDIO_FORMAT_UNKNOWN;
WAVEFORMATEX *wavformat =
(WAVEFORMATEX *) pin_mediatype->mediatype->pbFormat;
switch (wavformat->wFormatTag) {
case WAVE_FORMAT_PCM:
format = gst_audio_format_build_integer (TRUE, G_BYTE_ORDER, wavformat->wBitsPerSample, wavformat->wBitsPerSample);
break;
default:
break;
}
if (format != GST_AUDIO_FORMAT_UNKNOWN) {
GstAudioInfo info;
gst_audio_info_init(&info);
gst_audio_info_set_format(&info,
format,
wavformat->nSamplesPerSec,
wavformat->nChannels,
NULL);
mediacaps = gst_audio_info_to_caps(&info);
}
if (mediacaps) {
src->pins_mediatypes =
g_list_append (src->pins_mediatypes, pin_mediatype);
gst_caps_append (caps, mediacaps);
} else {
gst_dshow_free_pin_mediatype (pin_mediatype);
}
} else {
gst_dshow_free_pin_mediatype (pin_mediatype);
}
} else {
gst_dshow_free_pin_mediatype (pin_mediatype);
}
}
if (caps && gst_caps_is_empty (caps)) {
gst_caps_unref (caps);
caps = NULL;
}
return caps;
}
static gboolean
gst_dshowaudiosrc_push_buffer (guint8 * buffer, guint size, gpointer src_object,
GstClockTime duration)
{
GstDshowAudioSrc *src = GST_DSHOWAUDIOSRC (src_object);
if (!buffer || size == 0 || !src) {
return FALSE;
}
g_mutex_lock (&src->gbarray_lock);
g_byte_array_prepend (src->gbarray, buffer, size);
g_mutex_unlock (&src->gbarray_lock);
return TRUE;
}