/*****************************************************************************
* OpenDecoder: probe the decoder and return score
*****************************************************************************/
static int OpenDecoder( vlc_object_t *p_this )
{
decoder_t *p_dec = ( decoder_t* )p_this;
decoder_sys_t *p_sys;
GstStateChangeReturn i_ret;
gboolean b_ret;
sink_src_caps_t caps = { NULL, NULL };
GstStructure *p_str;
GstAppSrcCallbacks cb;
int i_rval = VLC_SUCCESS;
GList *p_list;
bool dbin;
#define VLC_GST_CHECK( r, v, s, t ) \
{ if( r == v ){ msg_Err( p_dec, s ); i_rval = t; goto fail; } }
if( !vlc_gst_init( ))
{
msg_Err( p_dec, "failed to register vlcvideosink" );
return VLC_EGENERIC;
}
p_str = vlc_to_gst_fmt( &p_dec->fmt_in );
if( !p_str )
return VLC_EGENERIC;
/* Allocate the memory needed to store the decoder's structure */
p_sys = p_dec->p_sys = calloc( 1, sizeof( *p_sys ) );
if( p_sys == NULL )
{
gst_structure_free( p_str );
return VLC_ENOMEM;
}
dbin = var_CreateGetBool( p_dec, "use-decodebin" );
msg_Dbg( p_dec, "Using decodebin? %s", dbin ? "yes ":"no" );
caps.p_sinkcaps = gst_caps_new_empty( );
gst_caps_append_structure( caps.p_sinkcaps, p_str );
/* Currently supports only system memory raw output format */
caps.p_srccaps = gst_caps_new_empty_simple( "video/x-raw" );
/* Get the list of all the available gstreamer decoders */
p_list = gst_element_factory_list_get_elements(
GST_ELEMENT_FACTORY_TYPE_DECODER, GST_RANK_MARGINAL );
VLC_GST_CHECK( p_list, NULL, "no decoder list found", VLC_ENOMOD );
if( !dbin )
{
GList *p_l;
/* Sort them as per ranks */
p_list = g_list_sort( p_list, gst_plugin_feature_rank_compare_func );
VLC_GST_CHECK( p_list, NULL, "failed to sort decoders list",
VLC_ENOMOD );
p_l = g_list_find_custom( p_list, &caps, find_decoder_func );
VLC_GST_CHECK( p_l, NULL, "no suitable decoder found",
VLC_ENOMOD );
/* create the decoder with highest rank */
p_sys->p_decode_in = gst_element_factory_create(
( GstElementFactory* )p_l->data, NULL );
VLC_GST_CHECK( p_sys->p_decode_in, NULL,
"failed to create decoder", VLC_ENOMOD );
}
else
{
GList *p_l;
/* Just check if any suitable decoder exists, rest will be
* handled by decodebin */
p_l = g_list_find_custom( p_list, &caps, find_decoder_func );
VLC_GST_CHECK( p_l, NULL, "no suitable decoder found",
VLC_ENOMOD );
}
gst_plugin_feature_list_free( p_list );
p_list = NULL;
gst_caps_unref( caps.p_srccaps );
caps.p_srccaps = NULL;
p_sys->b_prerolled = false;
p_sys->b_running = false;
/* Queue: GStreamer thread will dump buffers into this queue,
* DecodeBlock() will pop out the buffers from the queue */
p_sys->p_que = gst_atomic_queue_new( 0 );
VLC_GST_CHECK( p_sys->p_que, NULL, "failed to create queue",
VLC_ENOMEM );
p_sys->p_decode_src = gst_element_factory_make( "appsrc", NULL );
VLC_GST_CHECK( p_sys->p_decode_src, NULL, "appsrc not found",
VLC_ENOMOD );
g_object_set( G_OBJECT( p_sys->p_decode_src ), "caps", caps.p_sinkcaps,
"emit-signals", TRUE, "format", GST_FORMAT_BYTES,
"stream-type", GST_APP_STREAM_TYPE_SEEKABLE,
/* Making DecodeBlock() to block on appsrc with max queue size of 1 byte.
* This will make the push_buffer() tightly coupled with the buffer
* flow from appsrc -> decoder. push_buffer() will only return when
* the same buffer it just fed to appsrc has also been fed to the
* decoder element as well */
"block", TRUE, "max-bytes", ( guint64 )1, NULL );
gst_caps_unref( caps.p_sinkcaps );
caps.p_sinkcaps = NULL;
cb.enough_data = NULL;
cb.need_data = NULL;
cb.seek_data = seek_data_cb;
gst_app_src_set_callbacks( GST_APP_SRC( p_sys->p_decode_src ),
&cb, p_dec, NULL );
if( dbin )
{
p_sys->p_decode_in = gst_element_factory_make( "decodebin", NULL );
VLC_GST_CHECK( p_sys->p_decode_in, NULL, "decodebin not found",
VLC_ENOMOD );
//g_object_set( G_OBJECT( p_sys->p_decode_in ),
//"max-size-buffers", 2, NULL );
//g_signal_connect( G_OBJECT( p_sys->p_decode_in ), "no-more-pads",
//G_CALLBACK( no_more_pads_cb ), p_dec );
g_signal_connect( G_OBJECT( p_sys->p_decode_in ), "pad-added",
G_CALLBACK( pad_added_cb ), p_dec );
}
/* videosink: will emit signal for every available buffer */
p_sys->p_decode_out = gst_element_factory_make( "vlcvideosink", NULL );
VLC_GST_CHECK( p_sys->p_decode_out, NULL, "vlcvideosink not found",
VLC_ENOMOD );
p_sys->p_allocator = gst_vlc_picture_plane_allocator_new(
(gpointer) p_dec );
g_object_set( G_OBJECT( p_sys->p_decode_out ), "sync", FALSE, "allocator",
p_sys->p_allocator, "id", (gpointer) p_dec, NULL );
g_signal_connect( G_OBJECT( p_sys->p_decode_out ), "new-buffer",
G_CALLBACK( frame_handoff_cb ), p_dec );
//FIXME: caps_signal
#if 0
g_signal_connect( G_OBJECT( p_sys->p_decode_out ), "new-caps",
G_CALLBACK( caps_handoff_cb ), p_dec );
#else
GST_VLC_VIDEO_SINK( p_sys->p_decode_out )->new_caps = caps_handoff_cb;
#endif
p_sys->p_decoder = GST_ELEMENT( gst_bin_new( "decoder" ) );
VLC_GST_CHECK( p_sys->p_decoder, NULL, "bin not found", VLC_ENOMOD );
p_sys->p_bus = gst_bus_new( );
VLC_GST_CHECK( p_sys->p_bus, NULL, "failed to create bus",
VLC_ENOMOD );
gst_element_set_bus( p_sys->p_decoder, p_sys->p_bus );
gst_bin_add_many( GST_BIN( p_sys->p_decoder ),
p_sys->p_decode_src, p_sys->p_decode_in,
p_sys->p_decode_out, NULL );
gst_object_ref( p_sys->p_decode_src );
gst_object_ref( p_sys->p_decode_in );
gst_object_ref( p_sys->p_decode_out );
b_ret = gst_element_link( p_sys->p_decode_src, p_sys->p_decode_in );
VLC_GST_CHECK( b_ret, FALSE, "failed to link src <-> in",
VLC_EGENERIC );
if( !dbin )
{
b_ret = gst_element_link( p_sys->p_decode_in, p_sys->p_decode_out );
VLC_GST_CHECK( b_ret, FALSE, "failed to link in <-> out",
VLC_EGENERIC );
}
p_dec->fmt_out.i_cat = p_dec->fmt_in.i_cat;
/* set the pipeline to playing */
i_ret = gst_element_set_state( p_sys->p_decoder, GST_STATE_PLAYING );
VLC_GST_CHECK( i_ret, GST_STATE_CHANGE_FAILURE,
"set state failure", VLC_EGENERIC );
p_sys->b_running = true;
/* Set callbacks */
p_dec->pf_decode_video = DecodeBlock;
p_dec->pf_flush = Flush;
return VLC_SUCCESS;
fail:
if( caps.p_sinkcaps )
gst_caps_unref( caps.p_sinkcaps );
if( caps.p_srccaps )
gst_caps_unref( caps.p_srccaps );
if( p_list )
gst_plugin_feature_list_free( p_list );
CloseDecoder( ( vlc_object_t* )p_dec );
return i_rval;
}
bool CvCapture_GStreamer::open( int type, const char* filename )
{
close();
CV_FUNCNAME("cvCaptureFromCAM_GStreamer");
__BEGIN__;
gst_initializer::init();
// if(!isInited) {
// printf("gst_init\n");
// gst_init (NULL, NULL);
// gst_debug_set_active(TRUE);
// gst_debug_set_colored(TRUE);
// gst_debug_set_default_threshold(GST_LEVEL_WARNING);
// isInited = true;
// }
bool stream = false;
bool manualpipeline = false;
char *uri = NULL;
uridecodebin = NULL;
if(type != CV_CAP_GSTREAMER_FILE) {
close();
return false;
}
if(!gst_uri_is_valid(filename)) {
uri = realpath(filename, NULL);
stream=false;
if(uri) {
uri = g_filename_to_uri(uri, NULL, NULL);
if(!uri) {
CV_WARN("GStreamer: Error opening file\n");
close();
return false;
}
} else {
GError *err = NULL;
//uridecodebin = gst_parse_bin_from_description(filename, FALSE, &err);
uridecodebin = gst_parse_launch(filename, &err);
if(!uridecodebin) {
CV_WARN("GStreamer: Error opening bin\n");
close();
return false;
}
stream = true;
manualpipeline = true;
}
} else {
stream = true;
uri = g_strdup(filename);
}
if(!uridecodebin) {
uridecodebin = gst_element_factory_make ("uridecodebin", NULL);
g_object_set(G_OBJECT(uridecodebin),"uri",uri, NULL);
if(!uridecodebin) {
CV_WARN("GStreamer: Failed to create uridecodebin\n");
close();
return false;
}
}
if(manualpipeline) {
GstIterator *it = gst_bin_iterate_sinks(GST_BIN(uridecodebin));
if(gst_iterator_next(it, (gpointer *)&sink) != GST_ITERATOR_OK) {
CV_ERROR(CV_StsError, "GStreamer: cannot find appsink in manual pipeline\n");
return false;
}
pipeline = uridecodebin;
} else {
pipeline = gst_pipeline_new (NULL);
color = gst_element_factory_make("ffmpegcolorspace", NULL);
sink = gst_element_factory_make("appsink", NULL);
gst_bin_add_many(GST_BIN(pipeline), uridecodebin, color, sink, NULL);
g_signal_connect(uridecodebin, "pad-added", G_CALLBACK(newPad), color);
if(!gst_element_link(color, sink)) {
CV_ERROR(CV_StsError, "GStreamer: cannot link color -> sink\n");
gst_object_unref(pipeline);
return false;
}
}
gst_app_sink_set_max_buffers (GST_APP_SINK(sink), 1);
gst_app_sink_set_drop (GST_APP_SINK(sink), stream);
caps = gst_caps_new_simple("video/x-raw-rgb",
"red_mask", G_TYPE_INT, 0x0000FF,
"green_mask", G_TYPE_INT, 0x00FF00,
"blue_mask", G_TYPE_INT, 0xFF0000,
NULL);
gst_app_sink_set_caps(GST_APP_SINK(sink), caps);
gst_caps_unref(caps);
if(gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_READY) ==
GST_STATE_CHANGE_FAILURE) {
CV_WARN("GStreamer: unable to set pipeline to ready\n");
gst_object_unref(pipeline);
return false;
}
if(gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING) ==
GST_STATE_CHANGE_FAILURE) {
gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_NULL);
CV_WARN("GStreamer: unable to set pipeline to playing\n");
gst_object_unref(pipeline);
return false;
}
handleMessage();
__END__;
return true;
}
bool CvCapture_GStreamer::setProperty( int propId, double value )
{
GstFormat format;
GstSeekFlags flags;
if(!pipeline) {
CV_WARN("GStreamer: no pipeline");
return false;
}
switch(propId) {
case CV_CAP_PROP_POS_MSEC:
format = GST_FORMAT_TIME;
flags = (GstSeekFlags) (GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE);
if(!gst_element_seek_simple(GST_ELEMENT(pipeline), format,
flags, (gint64) (value * GST_MSECOND))) {
CV_WARN("GStreamer: unable to seek");
}
break;
case CV_CAP_PROP_POS_FRAMES:
format = GST_FORMAT_DEFAULT;
flags = (GstSeekFlags) (GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE);
if(!gst_element_seek_simple(GST_ELEMENT(pipeline), format,
flags, (gint64) value)) {
CV_WARN("GStreamer: unable to seek");
}
break;
case CV_CAP_PROP_POS_AVI_RATIO:
format = GST_FORMAT_PERCENT;
flags = (GstSeekFlags) (GST_SEEK_FLAG_FLUSH|GST_SEEK_FLAG_ACCURATE);
if(!gst_element_seek_simple(GST_ELEMENT(pipeline), format,
flags, (gint64) (value * GST_FORMAT_PERCENT_MAX))) {
CV_WARN("GStreamer: unable to seek");
}
break;
case CV_CAP_PROP_FRAME_WIDTH:
if(value > 0)
setFilter("width", G_TYPE_INT, (int) value, 0);
else
removeFilter("width");
break;
case CV_CAP_PROP_FRAME_HEIGHT:
if(value > 0)
setFilter("height", G_TYPE_INT, (int) value, 0);
else
removeFilter("height");
break;
case CV_CAP_PROP_FPS:
if(value > 0) {
int num, denom;
num = (int) value;
if(value != num) { // FIXME this supports only fractions x/1 and x/2
num = (int) (value * 2);
denom = 2;
} else
denom = 1;
setFilter("framerate", GST_TYPE_FRACTION, num, denom);
} else
removeFilter("framerate");
break;
case CV_CAP_PROP_FOURCC:
case CV_CAP_PROP_FRAME_COUNT:
case CV_CAP_PROP_FORMAT:
case CV_CAP_PROP_MODE:
case CV_CAP_PROP_BRIGHTNESS:
case CV_CAP_PROP_CONTRAST:
case CV_CAP_PROP_SATURATION:
case CV_CAP_PROP_HUE:
case CV_CAP_PROP_GAIN:
case CV_CAP_PROP_CONVERT_RGB:
break;
case CV_CAP_GSTREAMER_QUEUE_LENGTH:
if(!sink)
break;
gst_app_sink_set_max_buffers(GST_APP_SINK(sink), (guint) value);
break;
default:
CV_WARN("GStreamer: unhandled property");
}
return false;
}
/* Return the possible output caps based on inputs and downstream prefs */
static GstCaps *
_update_caps (GstVideoAggregator * vagg, GstCaps * caps, GstCaps * filter)
{
GstGLStereoMix *mix = GST_GL_STEREO_MIX (vagg);
GList *l;
gint best_width = -1, best_height = -1;
gdouble best_fps = -1, cur_fps;
gint best_fps_n = 0, best_fps_d = 1;
GstVideoInfo *mix_info;
GstCaps *blend_caps, *tmp_caps;
GstCaps *out_caps;
GST_OBJECT_LOCK (vagg);
for (l = GST_ELEMENT (vagg)->sinkpads; l; l = l->next) {
GstVideoAggregatorPad *pad = l->data;
GstVideoInfo tmp = pad->info;
gint this_width, this_height;
gint fps_n, fps_d;
if (!pad->info.finfo)
continue;
/* This can happen if we release a pad and another pad hasn't been negotiated_caps yet */
if (GST_VIDEO_INFO_FORMAT (&pad->info) == GST_VIDEO_FORMAT_UNKNOWN)
continue;
/* Convert to per-view width/height for unpacked forms */
gst_video_multiview_video_info_change_mode (&tmp,
GST_VIDEO_MULTIVIEW_MODE_SEPARATED, GST_VIDEO_MULTIVIEW_FLAGS_NONE);
this_width = GST_VIDEO_INFO_WIDTH (&tmp);
this_height = GST_VIDEO_INFO_HEIGHT (&tmp);
fps_n = GST_VIDEO_INFO_FPS_N (&tmp);
fps_d = GST_VIDEO_INFO_FPS_D (&tmp);
GST_INFO_OBJECT (vagg, "Input pad %" GST_PTR_FORMAT
" w %u h %u", pad, this_width, this_height);
if (this_width == 0 || this_height == 0)
continue;
if (best_width < this_width)
best_width = this_width;
if (best_height < this_height)
best_height = this_height;
if (fps_d == 0)
cur_fps = 0.0;
else
gst_util_fraction_to_double (fps_n, fps_d, &cur_fps);
if (best_fps < cur_fps) {
best_fps = cur_fps;
best_fps_n = fps_n;
best_fps_d = fps_d;
}
/* FIXME: Preserve PAR for at least one input when different sized inputs */
}
GST_OBJECT_UNLOCK (vagg);
mix_info = &mix->mix_info;
gst_video_info_set_format (mix_info, GST_VIDEO_FORMAT_RGBA, best_width,
best_height);
GST_VIDEO_INFO_FPS_N (mix_info) = best_fps_n;
GST_VIDEO_INFO_FPS_D (mix_info) = best_fps_d;
GST_VIDEO_INFO_MULTIVIEW_MODE (mix_info) = GST_VIDEO_MULTIVIEW_MODE_SEPARATED;
GST_VIDEO_INFO_VIEWS (mix_info) = 2;
/* FIXME: If input is marked as flipped or flopped, preserve those flags */
GST_VIDEO_INFO_MULTIVIEW_FLAGS (mix_info) = GST_VIDEO_MULTIVIEW_FLAGS_NONE;
/* Choose our output format based on downstream preferences */
blend_caps = gst_video_info_to_caps (mix_info);
gst_caps_set_features (blend_caps, 0,
gst_caps_features_from_string (GST_CAPS_FEATURE_MEMORY_GL_MEMORY));
tmp_caps = get_converted_caps (GST_GL_STEREO_MIX (vagg), blend_caps);
gst_caps_unref (blend_caps);
out_caps = gst_caps_intersect (caps, tmp_caps);
gst_caps_unref (tmp_caps);
GST_DEBUG_OBJECT (vagg, "Possible output caps %" GST_PTR_FORMAT, out_caps);
return out_caps;
}
static GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
const guint8 *data;
GstAudioChannelPosition pos[64];
const GstAudioChannelPosition *posn = NULL;
GstMapInfo map;
if (!gst_opus_header_is_id_header (buf)) {
GST_ERROR_OBJECT (dec, "Header is not an Opus ID header");
return GST_FLOW_ERROR;
}
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data;
if (!(dec->n_channels == 0 || dec->n_channels == data[9])) {
gst_buffer_unmap (buf, &map);
GST_ERROR_OBJECT (dec, "Opus ID header has invalid channels");
return GST_FLOW_ERROR;
}
dec->n_channels = data[9];
dec->sample_rate = GST_READ_UINT32_LE (data + 12);
dec->pre_skip = GST_READ_UINT16_LE (data + 10);
dec->r128_gain = GST_READ_UINT16_LE (data + 16);
dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
GST_INFO_OBJECT (dec,
"Found pre-skip of %u samples, R128 gain %d (volume %f)",
dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
dec->channel_mapping_family = data[18];
if (dec->channel_mapping_family == 0) {
/* implicit mapping */
GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
dec->n_streams = dec->n_stereo_streams = 1;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
} else {
dec->n_streams = data[19];
dec->n_stereo_streams = data[20];
memcpy (dec->channel_mapping, data + 21, dec->n_channels);
if (dec->channel_mapping_family == 1) {
GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
switch (dec->n_channels) {
case 1:
case 2:
/* nothing */
break;
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
posn = gst_opus_channel_positions[dec->n_channels - 1];
break;
default:{
gint i;
GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < dec->n_channels; i++)
pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
posn = pos;
}
}
} else {
GST_INFO_OBJECT (dec, "Channel mapping family %d",
dec->channel_mapping_family);
}
}
gst_opus_dec_negotiate (dec, posn);
gst_buffer_unmap (buf, &map);
return GST_FLOW_OK;
}
GstElement* webkitVideoSinkNew(WebCore::GStreamerGWorld* gstGWorld)
{
GstElement* element = GST_ELEMENT(g_object_new(WEBKIT_TYPE_VIDEO_SINK, 0));
WEBKIT_VIDEO_SINK(element)->priv->gstGWorld = gstGWorld;
return element;
}
static void
gst_curl_smtp_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstCurlSmtpSink *sink;
GstState cur_state;
g_return_if_fail (GST_IS_CURL_SMTP_SINK (object));
sink = GST_CURL_SMTP_SINK (object);
gst_element_get_state (GST_ELEMENT (sink), &cur_state, NULL, 0);
if (cur_state != GST_STATE_PLAYING && cur_state != GST_STATE_PAUSED) {
GST_OBJECT_LOCK (sink);
switch (prop_id) {
case PROP_MAIL_RCPT:
g_free (sink->mail_rcpt);
sink->mail_rcpt = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "mail-rcpt set to %s", sink->mail_rcpt);
break;
case PROP_MAIL_FROM:
g_free (sink->mail_from);
sink->mail_from = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "mail-from set to %s", sink->mail_from);
break;
case PROP_SUBJECT:
g_free (sink->subject);
sink->subject = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "subject set to %s", sink->subject);
break;
case PROP_MESSAGE_BODY:
g_free (sink->message_body);
sink->message_body = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "message-body set to %s", sink->message_body);
break;
case PROP_CONTENT_TYPE:
g_free (sink->content_type);
sink->content_type = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "content-type set to %s", sink->content_type);
break;
case PROP_USE_SSL:
sink->use_ssl = g_value_get_boolean (value);
GST_DEBUG_OBJECT (sink, "use-ssl set to %d", sink->use_ssl);
break;
case PROP_NBR_ATTACHMENTS:
sink->nbr_attachments = g_value_get_int (value);
sink->nbr_attachments_left = sink->nbr_attachments;
GST_DEBUG_OBJECT (sink, "nbr-attachments set to %d",
sink->nbr_attachments);
break;
case PROP_POP_USER_NAME:
g_free (sink->pop_user);
sink->pop_user = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "pop-user set to %s", sink->pop_user);
break;
case PROP_POP_USER_PASSWD:
g_free (sink->pop_passwd);
sink->pop_passwd = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "pop-passwd set to %s", sink->pop_passwd);
break;
case PROP_POP_LOCATION:
g_free (sink->pop_location);
sink->pop_location = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "pop-location set to %s", sink->pop_location);
break;
default:
GST_DEBUG_OBJECT (sink, "invalid property id %d", prop_id);
break;
}
GST_OBJECT_UNLOCK (sink);
return;
}
/* in PLAYING or PAUSED state */
GST_OBJECT_LOCK (sink);
switch (prop_id) {
case PROP_CONTENT_TYPE:
g_free (sink->content_type);
sink->content_type = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "content type set to %s", sink->content_type);
break;
default:
GST_WARNING_OBJECT (sink, "cannot set property when PLAYING");
break;
}
GST_OBJECT_UNLOCK (sink);
}
static void
gst_curl_http_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstCurlHttpSink *sink;
GstState cur_state;
g_return_if_fail (GST_IS_CURL_HTTP_SINK (object));
sink = GST_CURL_HTTP_SINK (object);
gst_element_get_state (GST_ELEMENT (sink), &cur_state, NULL, 0);
if (cur_state != GST_STATE_PLAYING && cur_state != GST_STATE_PAUSED) {
GST_OBJECT_LOCK (sink);
switch (prop_id) {
case PROP_PROXY:
g_free (sink->proxy);
sink->proxy = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "proxy set to %s", sink->proxy);
break;
case PROP_PROXY_PORT:
sink->proxy_port = g_value_get_int (value);
GST_DEBUG_OBJECT (sink, "proxy port set to %d", sink->proxy_port);
break;
case PROP_PROXY_USER_NAME:
g_free (sink->proxy_user);
sink->proxy_user = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "proxy user set to %s", sink->proxy_user);
break;
case PROP_PROXY_USER_PASSWD:
g_free (sink->proxy_passwd);
sink->proxy_passwd = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "proxy password set to %s", sink->proxy_passwd);
break;
case PROP_USE_CONTENT_LENGTH:
sink->use_content_length = g_value_get_boolean (value);
GST_DEBUG_OBJECT (sink, "use_content_length set to %d",
sink->use_content_length);
break;
case PROP_CONTENT_TYPE:
g_free (sink->content_type);
sink->content_type = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "content type set to %s", sink->content_type);
break;
default:
GST_DEBUG_OBJECT (sink, "invalid property id %d", prop_id);
break;
}
GST_OBJECT_UNLOCK (sink);
return;
}
/* in PLAYING or PAUSED state */
GST_OBJECT_LOCK (sink);
switch (prop_id) {
case PROP_CONTENT_TYPE:
g_free (sink->content_type);
sink->content_type = g_value_dup_string (value);
GST_DEBUG_OBJECT (sink, "content type set to %s", sink->content_type);
break;
default:
GST_WARNING_OBJECT (sink, "cannot set property when PLAYING");
break;
}
GST_OBJECT_UNLOCK (sink);
}
static GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
const guint8 *data = GST_BUFFER_DATA (buf);
GstCaps *caps;
const GstAudioChannelPosition *pos = NULL;
if (!gst_opus_header_is_id_header (buf)) {
GST_ERROR_OBJECT (dec, "Header is not an Opus ID header");
return GST_FLOW_ERROR;
}
if (!(dec->n_channels == 0 || dec->n_channels == data[9])) {
GST_ERROR_OBJECT (dec, "Opus ID header has invalid channels");
return GST_FLOW_ERROR;
}
dec->n_channels = data[9];
dec->pre_skip = GST_READ_UINT16_LE (data + 10);
dec->r128_gain = GST_READ_UINT16_LE (data + 16);
dec->r128_gain_volume = gst_opus_dec_get_r128_volume (dec->r128_gain);
GST_INFO_OBJECT (dec,
"Found pre-skip of %u samples, R128 gain %d (volume %f)",
dec->pre_skip, dec->r128_gain, dec->r128_gain_volume);
dec->channel_mapping_family = data[18];
if (dec->channel_mapping_family == 0) {
/* implicit mapping */
GST_INFO_OBJECT (dec, "Channel mapping family 0, implicit mapping");
dec->n_streams = dec->n_stereo_streams = 1;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
} else {
dec->n_streams = data[19];
dec->n_stereo_streams = data[20];
memcpy (dec->channel_mapping, data + 21, dec->n_channels);
if (dec->channel_mapping_family == 1) {
GST_INFO_OBJECT (dec, "Channel mapping family 1, Vorbis mapping");
switch (dec->n_channels) {
case 1:
case 2:
/* nothing */
break;
case 3:
case 4:
case 5:
case 6:
case 7:
case 8:
pos = gst_opus_channel_positions[dec->n_channels - 1];
break;
default:{
gint i;
GstAudioChannelPosition *posn =
g_new (GstAudioChannelPosition, dec->n_channels);
GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < dec->n_channels; i++)
posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
pos = posn;
}
}
} else {
GST_INFO_OBJECT (dec, "Channel mapping family %d",
dec->channel_mapping_family);
}
}
caps = gst_opus_dec_negotiate (dec);
if (pos) {
GST_DEBUG_OBJECT (dec, "Setting channel positions on caps");
gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
}
if (dec->n_channels > 8) {
g_free ((GstAudioChannelPosition *) pos);
}
GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
gst_caps_unref (caps);
return GST_FLOW_OK;
}
int GStreamer_init(const char *mplayer)
{
GError* error;
GstBus *bus;
GstElement *videosink, *audiosink;
int err;
if (g_initialized)
g_error("GStreamer: already initialized, call destroy first!\n");
g_state_callback = NULL;
g_duration = 0;
g_position = 0;
/* pthread synchronization */
pthread_mutex_init(&g_mutex, NULL);
err = pthread_cond_init(&g_main_cond, NULL);
if (err) {
g_error("GStreamer: failed to initialize main condition %s\n",
strerror(errno));
return -1;
}
/* init gstreamer library */
if (!gst_init_check(NULL, NULL, &error)) {
g_error("GStreamer: failed to initialize gstreamer library: [%d] %s\n",
error->code, error->message);
g_error_free(error);
return -1;
}
/* create pipeline */
g_pipeline = gst_pipeline_new("pipeline");
g_pipeline_name = gst_element_get_name(GST_ELEMENT(g_pipeline));
/* register callback */
bus = gst_pipeline_get_bus(GST_PIPELINE(g_pipeline));
gst_bus_add_watch(bus, my_bus_callback, NULL);
gst_object_unref(bus);
#if 0
/* TODO unlinked when removed from pipeline */
/* hardcode audio/video sink */
g_videosink = create_video_sink();
g_audiosink = create_audio_sink();
if (!g_videosink || !g_audiosink) {
/* TODO memory leak */
g_error("GStreamer: failed to create sink elements\n");
return -1;
}
#endif
/* prepare http/file src */
g_filesrc = gst_element_factory_make ("filesrc", "filesrc");
g_httpsrc = gst_element_factory_make ("souphttpsrc", "httpsrc");
if (!g_filesrc || !g_httpsrc) {
/* TODO memory leak */
g_error("GStreamer: failed to create src elements %x %x\n", g_filesrc, g_httpsrc);
return -1;
}
g_object_ref(g_filesrc);
g_object_ref(g_httpsrc);
/* initialize pipeline */
/* TODO do for audio/video pipe separately */
if (gst_element_set_state(g_pipeline, GST_STATE_READY) ==
GST_STATE_CHANGE_FAILURE) {
g_error("GStreamer: could not set pipeline to ready\n");
}
/* start main loop */
g_main_loop = g_main_loop_new(NULL, FALSE);
err = pthread_create(&g_reader_thread, NULL, main_thread_proc, NULL);
if (err) {
g_error("GStreamer: failed to launch gstreamer main thread %s\n",
strerror(errno));
goto err_pthread;
}
g_print("GStreamer: SUCCESSFULLY INITIALIZED\n");
g_initialized = 1;
return 0;
err_pthread:
pthread_cond_destroy(&g_main_cond);
pthread_mutex_destroy(&g_mutex);
return err;
}
static GstElement *
gst_auto_video_sink_find_best (GstAutoVideoSink * sink)
{
GList *list, *item;
GstElement *choice = NULL;
GstMessage *message = NULL;
GSList *errors = NULL;
GstBus *bus = gst_bus_new ();
GstPad *el_pad = NULL;
GstCaps *el_caps = NULL;
gboolean no_match = TRUE;
list = gst_registry_feature_filter (gst_registry_get (),
(GstPluginFeatureFilter) gst_auto_video_sink_factory_filter, FALSE, sink);
list = g_list_sort (list, (GCompareFunc) gst_auto_video_sink_compare_ranks);
GST_LOG_OBJECT (sink, "Trying to find usable video devices ...");
for (item = list; item != NULL; item = item->next) {
GstElementFactory *f = GST_ELEMENT_FACTORY (item->data);
GstElement *el;
if ((el = gst_auto_video_sink_create_element_with_pretty_name (sink, f))) {
GstStateChangeReturn ret;
GST_DEBUG_OBJECT (sink, "Testing %s", GST_OBJECT_NAME (f));
/* If autovideosink has been provided with filter caps,
* accept only sinks that match with the filter caps */
if (sink->filter_caps) {
el_pad = gst_element_get_static_pad (GST_ELEMENT (el), "sink");
el_caps = gst_pad_query_caps (el_pad, NULL);
gst_object_unref (el_pad);
GST_DEBUG_OBJECT (sink,
"Checking caps: %" GST_PTR_FORMAT " vs. %" GST_PTR_FORMAT,
sink->filter_caps, el_caps);
no_match = !gst_caps_can_intersect (sink->filter_caps, el_caps);
gst_caps_unref (el_caps);
if (no_match) {
GST_DEBUG_OBJECT (sink, "Incompatible caps");
gst_object_unref (el);
continue;
} else {
GST_DEBUG_OBJECT (sink, "Found compatible caps");
}
}
gst_element_set_bus (el, bus);
ret = gst_element_set_state (el, GST_STATE_READY);
if (ret == GST_STATE_CHANGE_SUCCESS) {
GST_DEBUG_OBJECT (sink, "This worked!");
choice = el;
break;
}
/* collect all error messages */
while ((message = gst_bus_pop_filtered (bus, GST_MESSAGE_ERROR))) {
GST_DEBUG_OBJECT (sink, "error message %" GST_PTR_FORMAT, message);
errors = g_slist_append (errors, message);
}
gst_element_set_state (el, GST_STATE_NULL);
gst_object_unref (el);
}
}
GST_DEBUG_OBJECT (sink, "done trying");
if (!choice) {
if (errors) {
/* FIXME: we forward the first error for now; but later on it might make
* sense to actually analyse them */
gst_message_ref (GST_MESSAGE (errors->data));
GST_DEBUG_OBJECT (sink, "reposting message %p", errors->data);
gst_element_post_message (GST_ELEMENT (sink), GST_MESSAGE (errors->data));
} else {
/* send warning message to application and use a fakesink */
GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
("Failed to find a usable video sink"));
choice = gst_element_factory_make ("fakesink", "fake-video-sink");
if (g_object_class_find_property (G_OBJECT_GET_CLASS (choice), "sync"))
g_object_set (choice, "sync", TRUE, NULL);
gst_element_set_state (choice, GST_STATE_READY);
}
}
gst_object_unref (bus);
gst_plugin_feature_list_free (list);
g_slist_foreach (errors, (GFunc) gst_mini_object_unref, NULL);
g_slist_free (errors);
return choice;
}
GstElement *create_video_sink()
{
GstElement *bin, *decoder = NULL;
GstIterator *iter;
GstIteratorResult res;
GError *error = NULL;
GstPad *pad;
gpointer element = NULL;
const char* decoder_name;
#ifndef DESKTOP
/* create pipeline */
decoder_name = "tividdec20";
bin = gst_parse_launch_full("TIViddec2 genTimeStamps=FALSE \
engineName=decode \
codecName=h264dec numFrames=-1 \
! videoscale method=0 \
! video/x-raw-yuv, format=(fourcc)I420, width=320, height=240 \
! ffmpegcolorspace \
! video/x-raw-rgb, bpp=16 \
! TIDmaiVideoSink displayStd=fbdev displayDevice=/dev/fb0 videoStd=QVGA \
videoOutput=LCD resizer=FALSE accelFrameCopy=TRUE",
NULL, 0, &error);
#else
decoder_name = "decodebin";
bin = gst_parse_launch_full("decodebin \
! videoscale method=0 \
! video/x-raw-yuv, format=(fourcc)I420, width=320, height=240 \
! xvimagesink",
NULL, 0, &error);
#endif
if (!bin) {
g_error("GStreamer: failed to parse video sink pipeline\n");
return NULL;
}
gst_object_set_name(GST_OBJECT(bin), "video-sink");
iter = gst_bin_iterate_elements(GST_BIN(bin));
res = gst_iterator_next (iter, &element);
while (res == GST_ITERATOR_OK) {
gchar *name;
name = gst_object_get_name(GST_OBJECT (element));
if (name) {
if (!strncmp(name, decoder_name, strlen(decoder_name))) {
decoder = GST_ELEMENT(element);
}
g_printf("GS: video sink element: %s \n", name);
g_free (name);
}
gst_object_unref (element);
element = NULL;
res = gst_iterator_next (iter, &element);
}
gst_iterator_free (iter);
if (!decoder) {
/* mem leak */
g_printf("decoder element not found\n");
return NULL;
}
/* add ghostpad */
pad = gst_element_get_static_pad (decoder, "sink");
gst_element_add_pad(bin, gst_ghost_pad_new("sink", pad));
gst_object_unref(GST_OBJECT(pad));
return bin;
}
/* http://<xxx>/manual/html/section-bus-message-types.html */
static gboolean my_bus_callback(GstBus *bus, GstMessage *msg,
gpointer user_data)
{
GstMessageType msgType;
GstObject *msgSrc;
gchar *msgSrcName;
/* used in switch */
/* error message */
gchar *debug;
GError *err;
GstState oldstate, newstate, pending;
/* stream status */
GstElement *owner;
msgType = GST_MESSAGE_TYPE(msg);
msgSrc = GST_MESSAGE_SRC(msg);
msgSrcName = GST_OBJECT_NAME(msgSrc);
switch (GST_MESSAGE_TYPE(msg)) {
case GST_MESSAGE_EOS:
g_print("GStreamer: end-of-stream\n");
pthread_mutex_lock(&g_mutex);
gst_element_set_state(GST_ELEMENT(g_pipeline), GST_STATE_NULL);
trigger_callback(GST_STATE_NULL);
pthread_mutex_unlock(&g_mutex);
break;
case GST_MESSAGE_ERROR:
gst_message_parse_error(msg, &err, &debug);
g_free (debug);
g_error("GStreamer: error: [%d] %s\n", err->code, err->message);
g_error_free(err);
/* TODO no sleep in callback */
pthread_mutex_lock(&g_mutex);
/* setting state to null flushes pipeline */
gst_element_set_state(GST_ELEMENT(g_pipeline), GST_STATE_NULL);
trigger_callback(GST_STATE_NULL);
pthread_mutex_unlock(&g_mutex);
break;
case GST_MESSAGE_STATE_CHANGED:
gst_message_parse_state_changed(msg, &oldstate, &newstate, &pending);
#if 0 /* noisy */
g_print("GStreamer: %s: State change: OLD: '%s', NEW: '%s', PENDING: '%s'\n",
msgSrcName,
gststate_get_name(oldstate),
gststate_get_name(newstate),
gststate_get_name(pending));
#endif
if (!strcmp(msgSrcName, g_pipeline_name))
trigger_callback(newstate); /* TODO GstState != GStreamer_state */
break;
case GST_MESSAGE_WARNING:
case GST_MESSAGE_INFO:
/* TODO */
break;
case GST_MESSAGE_APPLICATION: /* marshal information into the main thread */
case GST_MESSAGE_ASYNC_START:
case GST_MESSAGE_ASYNC_DONE:
case GST_MESSAGE_BUFFERING: /* caching of network streams */
case GST_MESSAGE_CLOCK_LOST:
case GST_MESSAGE_CLOCK_PROVIDE:
case GST_MESSAGE_ELEMENT: /* custom message, e.g. qtdemux redirect */
case GST_MESSAGE_LATENCY:
case GST_MESSAGE_NEW_CLOCK:
case GST_MESSAGE_REQUEST_STATE:
case GST_MESSAGE_SEGMENT_DONE:
case GST_MESSAGE_SEGMENT_START:
case GST_MESSAGE_STATE_DIRTY:
case GST_MESSAGE_STEP_DONE:
case GST_MESSAGE_STRUCTURE_CHANGE:
case GST_MESSAGE_TAG: /* meta data: artist, title */
/* ignore */
break;
case GST_MESSAGE_DURATION:
default:
g_print("GStreamer: BUS_CALL %s %d\n",
gst_message_type_get_name(GST_MESSAGE_TYPE(msg)),
GST_MESSAGE_TYPE(msg));
break;
}
return 1;
}
int
main (int argc, char *argv[])
{
GThread *input_thread;
gint numbuffers = -1;
gchar device[128] = { '\0' };
gchar input[128] = { '\0' };
gulong frequency = 0;
GstBus *bus;
/* see for input option */
int c;
while (1) {
static char long_options_desc[][64] = {
{"Number of buffers to output before sending EOS"},
{"Device location. Common in /dev/video0"},
{"input/output (channel) to switch to"},
{"frequency to tune to (in Hz)"},
{0, 0, 0, 0}
};
static struct option long_options[] = {
{"numbuffers", 1, 0, 'n'},
{"device", 1, 0, 'd'},
{"input", 1, 0, 'i'},
{"frequency", 1, 0, 'f'},
{0, 0, 0, 0}
};
/* getopt_long stores the option index here. */
int option_index = 0;
c = getopt_long (argc, argv, "n:d:i:f:h", long_options, &option_index);
/* Detect the end of the options. */
if (c == -1) {
printf ("tip: use -h to see help message.\n");
break;
}
switch (c) {
case 0:
/* If this option set a flag, do nothing else now. */
if (long_options[option_index].flag != 0)
break;
printf ("option %s", long_options[option_index].name);
if (optarg)
printf (" with arg %s", optarg);
printf ("\n");
break;
case 'n':
numbuffers = atoi (optarg);
break;
case 'd':
strncpy (device, optarg, sizeof (device) / sizeof (device[0]));
break;
case 'i':
strncpy (input, optarg, sizeof (input) / sizeof (input[0]));
break;
case 'f':
frequency = atol (optarg);
break;
case 'h':
printf ("Usage: v4l2src-test [OPTION]...\n");
for (c = 0; long_options[c].name; ++c) {
printf ("-%c, --%s\r\t\t\t\t%s\n", long_options[c].val,
long_options[c].name, long_options_desc[c]);
}
exit (0);
break;
case '?':
/* getopt_long already printed an error message. */
printf ("Use -h to see help message.\n");
break;
default:
abort ();
}
}
/* Print any remaining command line arguments (not options). */
if (optind < argc) {
printf ("Use -h to see help message.\n" "non-option ARGV-elements: ");
while (optind < argc)
printf ("%s ", argv[optind++]);
putchar ('\n');
}
/* init */
gst_init (&argc, &argv);
/* create elements */
if (!(pipeline = gst_pipeline_new ("my_pipeline"))) {
fprintf (stderr, "error: gst_pipeline_new return NULL");
return -1;
}
if (!(source = gst_element_factory_make ("v4l2src", NULL))) {
fprintf (stderr,
"error: gst_element_factory_make (\"v4l2src\", NULL) return NULL");
return -1;
}
if (!(sink = gst_element_factory_make ("xvimagesink", NULL))) {
fprintf (stderr,
"error: gst_element_factory_make (\"xvimagesink\", NULL) return NULL");
return -1;
}
if (numbuffers > -1) {
g_object_set (source, "num-buffers", numbuffers, NULL);
}
if (device[0]) {
g_object_set (source, "device", device, NULL);
}
if (input[0]) {
g_object_set (source, "input", input, NULL);
}
if (frequency) {
g_object_set (source, "frequency", frequency, NULL);
}
/* you would normally check that the elements were created properly */
bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
gst_bus_add_watch (bus, my_bus_callback, NULL);
/* put together a pipeline */
gst_bin_add_many (GST_BIN (pipeline), source, sink, NULL);
gst_element_link_pads (source, "src", sink, "sink");
/* start the pipeline */
gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_PLAYING);
loop = g_main_loop_new (NULL, FALSE);
input_thread = g_thread_try_new ("v4l2src-test", read_user, source, NULL);
if (input_thread == NULL) {
fprintf (stderr, "error: g_thread_try_new() failed");
return -1;
}
g_main_loop_run (loop);
g_thread_join (input_thread);
gst_element_set_state (GST_ELEMENT (pipeline), GST_STATE_NULL);
gst_object_unref (bus);
gst_object_unref (pipeline);
gst_deinit ();
return 0;
}
static gboolean
gst_hls_demux_cache_fragments (GstHLSDemux * demux)
{
gint i;
/* If this playlist is a variant playlist, select the first one
* and update it */
if (gst_m3u8_client_has_variant_playlist (demux->client)) {
GstM3U8 *child = NULL;
if (demux->connection_speed == 0) {
GST_M3U8_CLIENT_LOCK (demux->client);
child = demux->client->main->current_variant->data;
GST_M3U8_CLIENT_UNLOCK (demux->client);
} else {
GList *tmp = gst_m3u8_client_get_playlist_for_bitrate (demux->client,
demux->connection_speed);
child = GST_M3U8 (tmp->data);
}
gst_m3u8_client_set_current (demux->client, child);
if (!gst_hls_demux_update_playlist (demux, FALSE)) {
GST_ERROR_OBJECT (demux, "Could not fetch the child playlist %s",
child->uri);
return FALSE;
}
}
if (!gst_m3u8_client_is_live (demux->client)) {
GstClockTime duration = gst_m3u8_client_get_duration (demux->client);
GST_DEBUG_OBJECT (demux, "Sending duration message : %" GST_TIME_FORMAT,
GST_TIME_ARGS (duration));
if (duration != GST_CLOCK_TIME_NONE)
gst_element_post_message (GST_ELEMENT (demux),
gst_message_new_duration (GST_OBJECT (demux),
GST_FORMAT_TIME, duration));
}
/* Cache the first fragments */
for (i = 0; i < demux->fragments_cache; i++) {
gst_element_post_message (GST_ELEMENT (demux),
gst_message_new_buffering (GST_OBJECT (demux),
100 * i / demux->fragments_cache));
g_get_current_time (&demux->next_update);
if (!gst_hls_demux_get_next_fragment (demux, TRUE)) {
if (demux->end_of_playlist)
break;
if (!demux->cancelled)
GST_ERROR_OBJECT (demux, "Error caching the first fragments");
return FALSE;
}
/* make sure we stop caching fragments if something cancelled it */
if (demux->cancelled)
return FALSE;
gst_hls_demux_switch_playlist (demux);
}
gst_element_post_message (GST_ELEMENT (demux),
gst_message_new_buffering (GST_OBJECT (demux), 100));
g_get_current_time (&demux->next_update);
demux->need_cache = FALSE;
return TRUE;
}
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
GstFlowReturn res = GST_FLOW_OK;
gint size;
guint8 *data;
GstBuffer *outbuf;
gint16 *out_data;
int n, err;
int samples;
unsigned int packet_size;
GstBuffer *buf;
if (dec->state == NULL) {
/* If we did not get any headers, default to 2 channels */
if (dec->n_channels == 0) {
GstCaps *caps;
GST_INFO_OBJECT (dec, "No header, assuming single stream");
dec->n_channels = 2;
dec->sample_rate = 48000;
caps = gst_opus_dec_negotiate (dec);
GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
gst_caps_unref (caps);
/* default stereo mapping */
dec->channel_mapping_family = 0;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
dec->n_streams = 1;
dec->n_stereo_streams = 1;
}
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
dec->n_channels, dec->sample_rate);
#ifndef GST_DISABLE_DEBUG
gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
"Mapping table", dec->n_channels, dec->channel_mapping);
#endif
GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
dec->n_stereo_streams);
dec->state =
opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
if (!dec->state || err != OPUS_OK)
goto creation_failed;
}
if (buffer) {
GST_DEBUG_OBJECT (dec, "Received buffer of size %u",
GST_BUFFER_SIZE (buffer));
} else {
GST_DEBUG_OBJECT (dec, "Received missing buffer");
}
/* if using in-band FEC, we introdude one extra frame's delay as we need
to potentially wait for next buffer to decode a missing buffer */
if (dec->use_inband_fec && !dec->primed) {
GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
gst_buffer_replace (&dec->last_buffer, buffer);
dec->primed = TRUE;
goto done;
}
/* That's the buffer we'll be sending to the opus decoder. */
buf = (dec->use_inband_fec
&& GST_BUFFER_SIZE (dec->last_buffer) > 0) ? dec->last_buffer : buffer;
if (buf && GST_BUFFER_SIZE (buf) > 0) {
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
GST_DEBUG_OBJECT (dec, "Using buffer of size %u", size);
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "Using NULL buffer");
data = NULL;
size = 0;
}
/* use maximum size (120 ms) as the number of returned samples is
not constant over the stream. */
samples = 120 * dec->sample_rate / 1000;
packet_size = samples * dec->n_channels * 2;
res = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec),
GST_BUFFER_OFFSET_NONE, packet_size,
GST_PAD_CAPS (GST_AUDIO_DECODER_SRC_PAD (dec)), &outbuf);
if (res != GST_FLOW_OK) {
GST_DEBUG_OBJECT (dec, "buf alloc flow: %s", gst_flow_get_name (res));
return res;
}
out_data = (gint16 *) GST_BUFFER_DATA (outbuf);
if (dec->use_inband_fec) {
if (dec->last_buffer) {
/* normal delayed decode */
GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
} else {
/* FEC reconstruction decode */
GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
1);
}
} else {
/* normal decode */
GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0);
}
if (n < 0) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
gst_buffer_unref (outbuf);
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
GST_BUFFER_SIZE (outbuf) = n * 2 * dec->n_channels;
/* Skip any samples that need skipping */
if (dec->pre_skip > 0) {
guint scaled_pre_skip = dec->pre_skip * dec->sample_rate / 48000;
guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
GST_BUFFER_SIZE (outbuf) -= skip * 2 * dec->n_channels;
GST_BUFFER_DATA (outbuf) += skip * 2 * dec->n_channels;
dec->pre_skip -= scaled_skip;
GST_INFO_OBJECT (dec,
"Skipping %u samples (%u at 48000 Hz, %u left to skip)", skip,
scaled_skip, dec->pre_skip);
}
if (GST_BUFFER_SIZE (outbuf) == 0) {
gst_buffer_unref (outbuf);
outbuf = NULL;
}
/* Apply gain */
/* Would be better off leaving this to a volume element, as this is
a naive conversion that does too many int/float conversions.
However, we don't have control over the pipeline...
So make it optional if the user program wants to use a volume,
but do it by default so the correct volume goes out by default */
if (dec->apply_gain && outbuf && dec->r128_gain) {
unsigned int i, nsamples = GST_BUFFER_SIZE (outbuf) / 2;
double volume = dec->r128_gain_volume;
gint16 *samples = (gint16 *) GST_BUFFER_DATA (outbuf);
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
for (i = 0; i < nsamples; ++i) {
int sample = (int) (samples[i] * volume + 0.5);
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
}
}
if (dec->use_inband_fec) {
gst_buffer_replace (&dec->last_buffer, buffer);
}
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
if (res != GST_FLOW_OK)
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
done:
return res;
creation_failed:
GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
return GST_FLOW_ERROR;
}
/**
* gst_aggregator_iterate_sinkpads:
* @self: The #GstAggregator
* @func: The function to call.
* @user_data: The data to pass to @func.
*
* Iterate the sinkpads of aggregator to call a function on them.
*
* This method guarantees that @func will be called only once for each
* sink pad.
*/
gboolean
gst_aggregator_iterate_sinkpads (GstAggregator * self,
GstAggregatorPadForeachFunc func, gpointer user_data)
{
gboolean result = FALSE;
GstIterator *iter;
gboolean done = FALSE;
GValue item = { 0, };
GList *seen_pads = NULL;
iter = gst_element_iterate_sink_pads (GST_ELEMENT (self));
if (!iter)
goto no_iter;
while (!done) {
switch (gst_iterator_next (iter, &item)) {
case GST_ITERATOR_OK:
{
GstPad *pad;
pad = g_value_get_object (&item);
/* if already pushed, skip. FIXME, find something faster to tag pads */
if (pad == NULL || g_list_find (seen_pads, pad)) {
g_value_reset (&item);
break;
}
GST_LOG_OBJECT (self, "calling function on pad %s:%s",
GST_DEBUG_PAD_NAME (pad));
result = func (self, pad, user_data);
done = !result;
seen_pads = g_list_prepend (seen_pads, pad);
g_value_reset (&item);
break;
}
case GST_ITERATOR_RESYNC:
gst_iterator_resync (iter);
break;
case GST_ITERATOR_ERROR:
GST_ERROR_OBJECT (self,
"Could not iterate over internally linked pads");
done = TRUE;
break;
case GST_ITERATOR_DONE:
done = TRUE;
break;
}
}
g_value_unset (&item);
gst_iterator_free (iter);
if (seen_pads == NULL) {
GST_DEBUG_OBJECT (self, "No pad seen");
return FALSE;
}
g_list_free (seen_pads);
no_iter:
return result;
}
static GstFlowReturn
theora_handle_data_packet (GstTheoraDec * dec, ogg_packet * packet,
GstClockTime outtime, GstClockTime outdur)
{
/* normal data packet */
th_ycbcr_buffer buf;
GstBuffer *out;
gboolean keyframe;
GstFlowReturn result;
ogg_int64_t gp;
if (G_UNLIKELY (!dec->have_header))
goto not_initialized;
/* get timestamp and durations */
if (outtime == -1)
outtime = dec->last_timestamp;
if (outdur == -1)
outdur = gst_util_uint64_scale_int (GST_SECOND, dec->info.fps_denominator,
dec->info.fps_numerator);
/* calculate expected next timestamp */
if (outtime != -1 && outdur != -1)
dec->last_timestamp = outtime + outdur;
/* the second most significant bit of the first data byte is cleared
* for keyframes. We can only check it if it's not a zero-length packet. */
keyframe = packet->bytes && ((packet->packet[0] & 0x40) == 0);
if (G_UNLIKELY (keyframe)) {
GST_DEBUG_OBJECT (dec, "we have a keyframe");
dec->need_keyframe = FALSE;
} else if (G_UNLIKELY (dec->need_keyframe)) {
goto dropping;
}
GST_DEBUG_OBJECT (dec, "parsing data packet");
/* this does the decoding */
if (G_UNLIKELY (th_decode_packetin (dec->decoder, packet, &gp) < 0))
goto decode_error;
if (outtime != -1) {
gboolean need_skip;
GstClockTime running_time;
GstClockTime earliest_time;
gdouble proportion;
/* qos needs to be done on running time */
running_time = gst_segment_to_running_time (&dec->segment, GST_FORMAT_TIME,
outtime);
GST_OBJECT_LOCK (dec);
proportion = dec->proportion;
earliest_time = dec->earliest_time;
/* check for QoS, don't perform the last steps of getting and
* pushing the buffers that are known to be late. */
need_skip = earliest_time != -1 && running_time <= earliest_time;
GST_OBJECT_UNLOCK (dec);
if (need_skip) {
GstMessage *qos_msg;
guint64 stream_time;
gint64 jitter;
GST_DEBUG_OBJECT (dec, "skipping decoding: qostime %"
GST_TIME_FORMAT " <= %" GST_TIME_FORMAT,
GST_TIME_ARGS (running_time), GST_TIME_ARGS (earliest_time));
dec->dropped++;
stream_time =
gst_segment_to_stream_time (&dec->segment, GST_FORMAT_TIME, outtime);
jitter = GST_CLOCK_DIFF (running_time, earliest_time);
qos_msg =
gst_message_new_qos (GST_OBJECT_CAST (dec), FALSE, running_time,
stream_time, outtime, outdur);
gst_message_set_qos_values (qos_msg, jitter, proportion, 1000000);
gst_message_set_qos_stats (qos_msg, GST_FORMAT_BUFFERS,
dec->processed, dec->dropped);
gst_element_post_message (GST_ELEMENT_CAST (dec), qos_msg);
goto dropping_qos;
}
}
/* this does postprocessing and set up the decoded frame
* pointers in our yuv variable */
if (G_UNLIKELY (th_decode_ycbcr_out (dec->decoder, buf) < 0))
goto no_yuv;
if (G_UNLIKELY ((buf[0].width != dec->info.frame_width)
|| (buf[0].height != dec->info.frame_height)))
goto wrong_dimensions;
result = theora_handle_image (dec, buf, &out);
if (result != GST_FLOW_OK)
return result;
GST_BUFFER_OFFSET (out) = dec->frame_nr;
if (dec->frame_nr != -1)
dec->frame_nr++;
GST_BUFFER_OFFSET_END (out) = dec->frame_nr;
GST_BUFFER_TIMESTAMP (out) = outtime;
GST_BUFFER_DURATION (out) = outdur;
dec->processed++;
if (dec->segment.rate >= 0.0)
result = theora_dec_push_forward (dec, out);
else
result = theora_dec_push_reverse (dec, out);
return result;
/* ERRORS */
not_initialized:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("no header sent yet"));
return GST_FLOW_ERROR;
}
dropping:
{
GST_WARNING_OBJECT (dec, "dropping frame because we need a keyframe");
dec->discont = TRUE;
return GST_FLOW_OK;
}
dropping_qos:
{
if (dec->frame_nr != -1)
dec->frame_nr++;
dec->discont = TRUE;
GST_WARNING_OBJECT (dec, "dropping frame because of QoS");
return GST_FLOW_OK;
}
decode_error:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("theora decoder did not decode data packet"));
return GST_FLOW_ERROR;
}
no_yuv:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("couldn't read out YUV image"));
return GST_FLOW_ERROR;
}
wrong_dimensions:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, FORMAT,
(NULL), ("dimensions of image do not match header"));
return GST_FLOW_ERROR;
}
}
GstElement* webkitVideoSinkNew()
{
return GST_ELEMENT(g_object_new(WEBKIT_TYPE_VIDEO_SINK, 0));
}
static void
gst_wrapper_camera_bin_reset_video_src_caps (GstWrapperCameraBinSrc * self,
GstCaps * caps)
{
GstClock *clock;
gint64 base_time;
GST_DEBUG_OBJECT (self, "Resetting src caps to %" GST_PTR_FORMAT, caps);
if (self->src_vid_src) {
GstCaps *old_caps;
g_object_get (G_OBJECT (self->src_filter), "caps", &old_caps, NULL);
if (gst_caps_is_equal (caps, old_caps)) {
GST_DEBUG_OBJECT (self, "old and new caps are same, do not reset it");
if (old_caps)
gst_caps_unref (old_caps);
return;
}
if (old_caps)
gst_caps_unref (old_caps);
clock = gst_element_get_clock (self->src_vid_src);
base_time = gst_element_get_base_time (self->src_vid_src);
/* Ideally, we should only need to get the source to READY here,
* but it seems v4l2src isn't happy with this. Putting to NULL makes
* it work.
*
* TODO fix this in v4l2src
*/
gst_element_set_state (self->src_vid_src, GST_STATE_NULL);
set_capsfilter_caps (self, caps);
self->drop_newseg = TRUE;
GST_DEBUG_OBJECT (self, "Bringing source up");
if (!gst_element_sync_state_with_parent (self->src_vid_src)) {
GST_WARNING_OBJECT (self, "Failed to reset source caps");
gst_element_set_state (self->src_vid_src, GST_STATE_NULL);
}
if (clock) {
gst_element_set_clock (self->src_vid_src, clock);
gst_element_set_base_time (self->src_vid_src, base_time);
if (GST_IS_BIN (self->src_vid_src)) {
GstIterator *it =
gst_bin_iterate_elements (GST_BIN (self->src_vid_src));
gpointer item = NULL;
gboolean done = FALSE;
while (!done) {
switch (gst_iterator_next (it, &item)) {
case GST_ITERATOR_OK:
gst_element_set_base_time (GST_ELEMENT (item), base_time);
gst_object_unref (item);
break;
case GST_ITERATOR_RESYNC:
gst_iterator_resync (it);
break;
case GST_ITERATOR_ERROR:
done = TRUE;
break;
case GST_ITERATOR_DONE:
done = TRUE;
break;
}
}
gst_iterator_free (it);
}
gst_object_unref (clock);
}
}
}
static void
gst_gnome_vfs_sink_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstGnomeVFSSink *sink;
GstState cur_state;
sink = GST_GNOME_VFS_SINK (object);
gst_element_get_state (GST_ELEMENT (sink), &cur_state, NULL, 0);
if (cur_state == GST_STATE_PLAYING || cur_state == GST_STATE_PAUSED) {
GST_WARNING_OBJECT (sink, "cannot set property when PAUSED or PLAYING");
return;
}
GST_OBJECT_LOCK (sink);
switch (prop_id) {
case ARG_LOCATION:{
const gchar *new_location;
if (sink->uri) {
gnome_vfs_uri_unref (sink->uri);
sink->uri = NULL;
}
if (sink->uri_name) {
g_free (sink->uri_name);
sink->uri_name = NULL;
}
new_location = g_value_get_string (value);
if (new_location) {
sink->uri_name = gst_gnome_vfs_location_to_uri_string (new_location);
sink->uri = gnome_vfs_uri_new (sink->uri_name);
}
break;
}
case ARG_URI:{
if (sink->uri) {
gnome_vfs_uri_unref (sink->uri);
sink->uri = NULL;
}
if (sink->uri_name) {
g_free (sink->uri_name);
sink->uri_name = NULL;
}
if (g_value_get_boxed (value)) {
sink->uri = (GnomeVFSURI *) g_value_dup_boxed (value);
sink->uri_name = gnome_vfs_uri_to_string (sink->uri, 0);
}
break;
}
case ARG_HANDLE:{
if (sink->uri) {
gnome_vfs_uri_unref (sink->uri);
sink->uri = NULL;
}
if (sink->uri_name) {
g_free (sink->uri_name);
sink->uri_name = NULL;
}
sink->handle = g_value_get_boxed (value);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (sink);
}
static GstElement *find_color_balance_element() {
GstIterator *iterator = gst_bin_iterate_all_by_interface(
GST_BIN(pipeline), GST_TYPE_COLOR_BALANCE);
GstElement *color_balance_element = NULL;
gboolean done = FALSE, hardware = FALSE;
#if GST_CHECK_VERSION(1, 0, 0)
GValue item = G_VALUE_INIT;
#else
gpointer item;
#endif
while (!done) {
switch (gst_iterator_next(iterator, &item)) {
case GST_ITERATOR_OK : {
#if GST_CHECK_VERSION(1, 0, 0)
GstElement *element = g_value_get_object(&item);
#else
GstElement *element = GST_ELEMENT(item);
#endif
if (is_valid_color_balance_element(element)) {
if (!color_balance_element) {
color_balance_element = GST_ELEMENT_CAST(
gst_object_ref(element));
hardware =
(gst_color_balance_get_balance_type(GST_COLOR_BALANCE
(element)) == GST_COLOR_BALANCE_HARDWARE);
}
else if (!hardware) {
gboolean tmp =
(gst_color_balance_get_balance_type(GST_COLOR_BALANCE
(element)) == GST_COLOR_BALANCE_HARDWARE);
if (tmp) {
if (color_balance_element)
gst_object_unref(color_balance_element);
color_balance_element =
GST_ELEMENT_CAST(gst_object_ref(element));
hardware = TRUE;
}
}
}
#if GST_CHECK_VERSION(1, 0, 0)
g_value_reset(&item);
#endif
if (hardware && color_balance_element)
done = TRUE;
break;
}
case GST_ITERATOR_RESYNC :
gst_iterator_resync(iterator);
done = FALSE;
hardware = FALSE;
if (color_balance_element)
gst_object_unref(color_balance_element);
color_balance_element = NULL;
break;
case GST_ITERATOR_DONE:
case GST_ITERATOR_ERROR:
default:
done = TRUE;
}
}
#if GST_CHECK_VERSION(1, 0, 0)
g_value_unset(&item);
#endif
gst_iterator_free(iterator);
return color_balance_element;
}
/* called with the object lock held */
static gboolean
gst_gl_stereo_mix_process_frames (GstGLStereoMix * mixer)
{
GstVideoAggregator *vagg = GST_VIDEO_AGGREGATOR (mixer);
GstBuffer *converted_buffer, *inbuf;
GstVideoInfo *out_info = &vagg->info;
#ifndef G_DISABLE_ASSERT
gint n;
#endif
gint v, views;
gint valid_views = 0;
GList *walk;
inbuf = gst_buffer_new ();
walk = GST_ELEMENT (mixer)->sinkpads;
while (walk) {
GstGLStereoMixPad *pad = walk->data;
GstMemory *in_mem;
GST_LOG_OBJECT (mixer, "Handling frame %d", valid_views);
if (!pad || !pad->current_buffer) {
GST_DEBUG ("skipping texture, null frame");
walk = g_list_next (walk);
continue;
}
in_mem = gst_buffer_get_memory (pad->current_buffer, 0);
GST_LOG_OBJECT (mixer,
"Appending memory %" GST_PTR_FORMAT " to intermediate buffer", in_mem);
/* Appending the memory to a 2nd buffer locks it
* exclusive a 2nd time, which will mark it for
* copy-on-write. The ref will keep the memory
* alive but we add a parent_buffer_meta to also
* prevent the input buffer from returning to any buffer
* pool it might belong to
*/
gst_buffer_append_memory (inbuf, in_mem);
/* Use parent buffer meta to keep input buffer alive */
gst_buffer_add_parent_buffer_meta (inbuf, pad->current_buffer);
valid_views++;
walk = g_list_next (walk);
}
if (mixer->mix_info.views != valid_views) {
GST_WARNING_OBJECT (mixer, "Not enough input views to process");
return FALSE;
}
if (GST_VIDEO_INFO_MULTIVIEW_MODE (out_info) ==
GST_VIDEO_MULTIVIEW_MODE_SEPARATED)
views = out_info->views;
else
views = 1;
if (gst_gl_view_convert_submit_input_buffer (mixer->viewconvert,
FALSE, inbuf) != GST_FLOW_OK)
return FALSE;
/* Clear any existing buffers, just in case */
gst_buffer_replace (&mixer->primary_out, NULL);
gst_buffer_replace (&mixer->auxilliary_out, NULL);
if (gst_gl_view_convert_get_output (mixer->viewconvert,
&mixer->primary_out) != GST_FLOW_OK)
return FALSE;
if (GST_VIDEO_INFO_MULTIVIEW_MODE (out_info) ==
GST_VIDEO_MULTIVIEW_MODE_FRAME_BY_FRAME) {
if (gst_gl_view_convert_get_output (mixer->viewconvert,
&mixer->auxilliary_out) != GST_FLOW_OK)
return FALSE;
}
if (mixer->primary_out == NULL)
return FALSE;
converted_buffer = mixer->primary_out;
#ifndef G_DISABLE_ASSERT
n = gst_buffer_n_memory (converted_buffer);
g_assert (n == GST_VIDEO_INFO_N_PLANES (out_info) * views);
#endif
for (v = 0; v < views; v++) {
gst_buffer_add_video_meta_full (converted_buffer, v,
GST_VIDEO_INFO_FORMAT (out_info),
GST_VIDEO_INFO_WIDTH (out_info),
GST_VIDEO_INFO_HEIGHT (out_info),
GST_VIDEO_INFO_N_PLANES (out_info), out_info->offset, out_info->stride);
if (mixer->auxilliary_out) {
gst_buffer_add_video_meta_full (mixer->auxilliary_out, v,
GST_VIDEO_INFO_FORMAT (out_info),
GST_VIDEO_INFO_WIDTH (out_info),
GST_VIDEO_INFO_HEIGHT (out_info),
GST_VIDEO_INFO_N_PLANES (out_info), out_info->offset,
out_info->stride);
}
}
return TRUE;
}
static void
gst_nle_source_pad_added_cb (GstElement * element, GstPad * pad,
GstNleSource * nlesrc)
{
GstCaps *caps;
const GstStructure *s;
const gchar *mime;
GstElement *appsink = NULL;
GstPad *sink_pad;
GstAppSinkCallbacks appsink_cbs;
GstNleSrcItem *item;
item = (GstNleSrcItem *) g_list_nth_data (nlesrc->queue, nlesrc->index);
caps = gst_pad_get_caps_reffed (pad);
s = gst_caps_get_structure (caps, 0);
mime = gst_structure_get_name (s);
GST_DEBUG_OBJECT (nlesrc, "Found mime type: %s", mime);
if (g_strrstr (mime, "video") && !nlesrc->video_linked) {
appsink = gst_element_factory_make ("appsink", NULL);
memset (&appsink_cbs, 0, sizeof (appsink_cbs));
appsink_cbs.eos = gst_nle_source_on_video_eos;
appsink_cbs.new_preroll = gst_nle_source_on_preroll_buffer;
appsink_cbs.new_buffer = gst_nle_source_on_video_buffer;
nlesrc->video_linked = TRUE;
if (!nlesrc->video_srcpad_added) {
gst_pad_set_active (nlesrc->video_srcpad, TRUE);
gst_element_add_pad (GST_ELEMENT (nlesrc),
gst_object_ref (nlesrc->video_srcpad));
nlesrc->video_srcpad_added = TRUE;
}
gst_pad_add_event_probe (GST_BASE_SINK_PAD (GST_BASE_SINK (appsink)),
(GCallback) gst_nle_source_video_pad_probe_cb, nlesrc);
nlesrc->video_eos = FALSE;
} else if (g_strrstr (mime, "audio") && nlesrc->with_audio
&& !nlesrc->audio_linked && (item ? item->rate == 1.0 : TRUE)) {
appsink = gst_element_factory_make ("appsink", NULL);
memset (&appsink_cbs, 0, sizeof (appsink_cbs));
appsink_cbs.eos = gst_nle_source_on_audio_eos;
appsink_cbs.new_preroll = gst_nle_source_on_preroll_buffer;
appsink_cbs.new_buffer = gst_nle_source_on_audio_buffer;
nlesrc->audio_linked = TRUE;
if (!nlesrc->audio_srcpad_added) {
gst_pad_set_active (nlesrc->audio_srcpad, TRUE);
gst_element_add_pad (GST_ELEMENT (nlesrc),
gst_object_ref (nlesrc->audio_srcpad));
nlesrc->audio_srcpad_added = TRUE;
}
gst_pad_add_event_probe (GST_BASE_SINK_PAD (GST_BASE_SINK (appsink)),
(GCallback) gst_nle_source_audio_pad_probe_cb, nlesrc);
nlesrc->audio_eos = FALSE;
}
if (appsink != NULL) {
g_object_set (appsink, "sync", FALSE, NULL);
gst_app_sink_set_callbacks (GST_APP_SINK (appsink), &appsink_cbs, nlesrc,
NULL);
gst_bin_add (GST_BIN (nlesrc->decoder), appsink);
sink_pad = gst_element_get_static_pad (appsink, "sink");
gst_pad_link (pad, sink_pad);
gst_element_sync_state_with_parent (appsink);
gst_object_unref (sink_pad);
}
}
static GstFlowReturn
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
GstFlowReturn res = GST_FLOW_OK;
gsize size;
guint8 *data;
GstBuffer *outbuf;
gint16 *out_data;
int n, err;
int samples;
unsigned int packet_size;
GstBuffer *buf;
GstMapInfo map, omap;
if (dec->state == NULL) {
/* If we did not get any headers, default to 2 channels */
if (dec->n_channels == 0) {
GST_INFO_OBJECT (dec, "No header, assuming single stream");
dec->n_channels = 2;
dec->sample_rate = 48000;
/* default stereo mapping */
dec->channel_mapping_family = 0;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
dec->n_streams = 1;
dec->n_stereo_streams = 1;
gst_opus_dec_negotiate (dec, NULL);
}
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
dec->n_channels, dec->sample_rate);
#ifndef GST_DISABLE_GST_DEBUG
gst_opus_common_log_channel_mapping_table (GST_ELEMENT (dec), opusdec_debug,
"Mapping table", dec->n_channels, dec->channel_mapping);
#endif
GST_DEBUG_OBJECT (dec, "%d streams, %d stereo", dec->n_streams,
dec->n_stereo_streams);
dec->state =
opus_multistream_decoder_create (dec->sample_rate, dec->n_channels,
dec->n_streams, dec->n_stereo_streams, dec->channel_mapping, &err);
if (!dec->state || err != OPUS_OK)
goto creation_failed;
}
if (buffer) {
GST_DEBUG_OBJECT (dec, "Received buffer of size %" G_GSIZE_FORMAT,
gst_buffer_get_size (buffer));
} else {
GST_DEBUG_OBJECT (dec, "Received missing buffer");
}
/* if using in-band FEC, we introdude one extra frame's delay as we need
to potentially wait for next buffer to decode a missing buffer */
if (dec->use_inband_fec && !dec->primed) {
GST_DEBUG_OBJECT (dec, "First buffer received in FEC mode, early out");
gst_buffer_replace (&dec->last_buffer, buffer);
dec->primed = TRUE;
goto done;
}
/* That's the buffer we'll be sending to the opus decoder. */
buf = (dec->use_inband_fec
&& gst_buffer_get_size (dec->last_buffer) >
0) ? dec->last_buffer : buffer;
if (buf && gst_buffer_get_size (buf) > 0) {
gst_buffer_map (buf, &map, GST_MAP_READ);
data = map.data;
size = map.size;
GST_DEBUG_OBJECT (dec, "Using buffer of size %" G_GSIZE_FORMAT, size);
} else {
/* concealment data, pass NULL as the bits parameters */
GST_DEBUG_OBJECT (dec, "Using NULL buffer");
data = NULL;
size = 0;
}
if (gst_buffer_get_size (buffer) == 0) {
GstClockTime const opus_plc_alignment = 2500 * GST_USECOND;
GstClockTime aligned_missing_duration;
GstClockTime missing_duration = GST_BUFFER_DURATION (buffer);
GST_DEBUG_OBJECT (dec,
"missing buffer, doing PLC duration %" GST_TIME_FORMAT
" plus leftover %" GST_TIME_FORMAT, GST_TIME_ARGS (missing_duration),
GST_TIME_ARGS (dec->leftover_plc_duration));
/* add the leftover PLC duration to that of the buffer */
missing_duration += dec->leftover_plc_duration;
/* align the combined buffer and leftover PLC duration to multiples
* of 2.5ms, always rounding down, and store excess duration for later */
aligned_missing_duration =
(missing_duration / opus_plc_alignment) * opus_plc_alignment;
dec->leftover_plc_duration = missing_duration - aligned_missing_duration;
/* Opus' PLC cannot operate with less than 2.5ms; skip PLC
* and accumulate the missing duration in the leftover_plc_duration
* for the next PLC attempt */
if (aligned_missing_duration < opus_plc_alignment) {
GST_DEBUG_OBJECT (dec,
"current duration %" GST_TIME_FORMAT
" of missing data not enough for PLC (minimum needed: %"
GST_TIME_FORMAT ") - skipping", GST_TIME_ARGS (missing_duration),
GST_TIME_ARGS (opus_plc_alignment));
goto done;
}
/* convert the duration (in nanoseconds) to sample count */
samples =
gst_util_uint64_scale_int (aligned_missing_duration, dec->sample_rate,
GST_SECOND);
GST_DEBUG_OBJECT (dec,
"calculated PLC frame length: %" GST_TIME_FORMAT
" num frame samples: %d new leftover: %" GST_TIME_FORMAT,
GST_TIME_ARGS (aligned_missing_duration), samples,
GST_TIME_ARGS (dec->leftover_plc_duration));
} else {
/* use maximum size (120 ms) as the number of returned samples is
not constant over the stream. */
samples = 120 * dec->sample_rate / 1000;
}
packet_size = samples * dec->n_channels * 2;
outbuf =
gst_audio_decoder_allocate_output_buffer (GST_AUDIO_DECODER (dec),
packet_size);
if (!outbuf) {
goto buffer_failed;
}
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = (gint16 *) omap.data;
if (dec->use_inband_fec) {
if (gst_buffer_get_size (dec->last_buffer) > 0) {
/* normal delayed decode */
GST_LOG_OBJECT (dec, "FEC enabled, decoding last delayed buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
0);
} else {
/* FEC reconstruction decode */
GST_LOG_OBJECT (dec, "FEC enabled, reconstructing last buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples,
1);
}
} else {
/* normal decode */
GST_LOG_OBJECT (dec, "FEC disabled, decoding buffer");
n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0);
}
gst_buffer_unmap (outbuf, &omap);
if (data != NULL)
gst_buffer_unmap (buf, &map);
if (n < 0) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
gst_buffer_unref (outbuf);
return GST_FLOW_ERROR;
}
GST_DEBUG_OBJECT (dec, "decoded %d samples", n);
gst_buffer_set_size (outbuf, n * 2 * dec->n_channels);
/* Skip any samples that need skipping */
if (dec->pre_skip > 0) {
guint scaled_pre_skip = dec->pre_skip * dec->sample_rate / 48000;
guint skip = scaled_pre_skip > n ? n : scaled_pre_skip;
guint scaled_skip = skip * 48000 / dec->sample_rate;
gst_buffer_resize (outbuf, skip * 2 * dec->n_channels, -1);
dec->pre_skip -= scaled_skip;
GST_INFO_OBJECT (dec,
"Skipping %u samples (%u at 48000 Hz, %u left to skip)", skip,
scaled_skip, dec->pre_skip);
}
if (gst_buffer_get_size (outbuf) == 0) {
gst_buffer_unref (outbuf);
outbuf = NULL;
} else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
dec->n_channels, dec->opus_pos, dec->info.position);
}
/* Apply gain */
/* Would be better off leaving this to a volume element, as this is
a naive conversion that does too many int/float conversions.
However, we don't have control over the pipeline...
So make it optional if the user program wants to use a volume,
but do it by default so the correct volume goes out by default */
if (dec->apply_gain && outbuf && dec->r128_gain) {
gsize rsize;
unsigned int i, nsamples;
double volume = dec->r128_gain_volume;
gint16 *samples;
gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
samples = (gint16 *) omap.data;
rsize = omap.size;
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
nsamples = rsize / 2;
for (i = 0; i < nsamples; ++i) {
int sample = (int) (samples[i] * volume + 0.5);
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
}
gst_buffer_unmap (outbuf, &omap);
}
if (dec->use_inband_fec) {
gst_buffer_replace (&dec->last_buffer, buffer);
}
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
if (res != GST_FLOW_OK)
GST_DEBUG_OBJECT (dec, "flow: %s", gst_flow_get_name (res));
done:
return res;
creation_failed:
GST_ERROR_OBJECT (dec, "Failed to create Opus decoder: %d", err);
return GST_FLOW_ERROR;
buffer_failed:
GST_ERROR_OBJECT (dec, "Failed to create %u byte buffer", packet_size);
return GST_FLOW_ERROR;
}
static void
gst_nle_source_next (GstNleSource * nlesrc)
{
GstNleSrcItem *item;
GstStateChangeReturn ret;
GstElement *uridecodebin;
GstBus *bus;
GstState state;
nlesrc->index++;
if (nlesrc->index >= g_list_length (nlesrc->queue)) {
gst_nle_source_push_eos (nlesrc);
return;
}
if (nlesrc->source != NULL) {
gst_object_unref (nlesrc->source);
nlesrc->source = NULL;
}
if (nlesrc->decoder != NULL) {
gst_element_set_state (GST_ELEMENT (nlesrc->decoder), GST_STATE_NULL);
gst_element_get_state (GST_ELEMENT (nlesrc->decoder), NULL, NULL, 0);
gst_object_unref (nlesrc->decoder);
}
nlesrc->decoder = gst_pipeline_new ("decoder");
uridecodebin = gst_element_factory_make ("uridecodebin", NULL);
/* Connect signal to recover source element for queries in bytes */
g_signal_connect (uridecodebin, "source-setup",
G_CALLBACK (gst_nle_source_on_source_setup), nlesrc);
gst_bin_add (GST_BIN (nlesrc->decoder), uridecodebin);
g_signal_connect (uridecodebin, "autoplug-select",
G_CALLBACK (lgm_filter_video_decoders), nlesrc);
g_signal_connect (uridecodebin, "pad-added",
G_CALLBACK (gst_nle_source_pad_added_cb), nlesrc);
g_signal_connect (uridecodebin, "no-more-pads",
G_CALLBACK (gst_nle_source_no_more_pads), nlesrc);
bus = GST_ELEMENT_BUS (nlesrc->decoder);
gst_bus_add_signal_watch (bus);
g_signal_connect (bus, "message", G_CALLBACK (gst_nle_source_bus_message),
nlesrc);
item = (GstNleSrcItem *) g_list_nth_data (nlesrc->queue, nlesrc->index);
GST_INFO_OBJECT (nlesrc, "Starting next item with uri:%s", item->file_path);
GST_INFO_OBJECT (nlesrc, "start:%" GST_TIME_FORMAT " stop:%"
GST_TIME_FORMAT " rate:%f", GST_TIME_ARGS (item->start),
GST_TIME_ARGS (item->stop), item->rate);
g_object_set (uridecodebin, "uri", item->file_path, NULL);
nlesrc->seek_done = FALSE;
if (GST_CLOCK_TIME_IS_VALID (item->stop)) {
nlesrc->video_seek_done = FALSE;
nlesrc->audio_seek_done = FALSE;
} else {
nlesrc->video_seek_done = TRUE;
nlesrc->audio_seek_done = TRUE;
}
nlesrc->audio_eos = TRUE;
nlesrc->video_eos = TRUE;
nlesrc->audio_ts = 0;
nlesrc->video_ts = 0;
nlesrc->start_ts = nlesrc->accu_time;
nlesrc->video_linked = FALSE;
nlesrc->audio_linked = FALSE;
nlesrc->item_setup = FALSE;
nlesrc->cached_duration = 0;
GST_DEBUG_OBJECT (nlesrc, "Start ts:%" GST_TIME_FORMAT,
GST_TIME_ARGS (nlesrc->start_ts));
gst_element_set_state (nlesrc->decoder, GST_STATE_PLAYING);
ret = gst_element_get_state (nlesrc->decoder, &state, NULL, 5 * GST_SECOND);
if (ret == GST_STATE_CHANGE_FAILURE) {
GST_WARNING_OBJECT (nlesrc, "Error changing state, selecting next item.");
gst_nle_source_check_eos (nlesrc);
return;
}
nlesrc->seek_done = TRUE;
if (!item->still_picture && GST_CLOCK_TIME_IS_VALID (item->stop)) {
GST_DEBUG_OBJECT (nlesrc, "Sending seek event");
gst_element_seek (nlesrc->decoder, 1, GST_FORMAT_TIME,
GST_SEEK_FLAG_ACCURATE,
GST_SEEK_TYPE_SET, item->start, GST_SEEK_TYPE_SET, item->stop);
}
}
bool CvVideoWriter_GStreamer::open( const char * filename, int fourcc,
double fps, CvSize frameSize, bool is_color )
{
CV_FUNCNAME("CvVideoWriter_GStreamer::open");
__BEGIN__;
//actually doesn't support fourcc parameter and encode an avi with jpegenc
//we need to find a common api between backend to support fourcc for avi
//but also to choose in a common way codec and container format (ogg,dirac,matroska)
// check arguments
assert (filename);
assert (fps > 0);
assert (frameSize.width > 0 && frameSize.height > 0);
std::map<int,char*>::iterator encit;
encit=encs.find(fourcc);
if (encit==encs.end())
CV_ERROR( CV_StsUnsupportedFormat,"Gstreamer Opencv backend doesn't support this codec acutally.");
// if(!isInited) {
// gst_init (NULL, NULL);
// isInited = true;
// }
gst_initializer::init();
close();
source=gst_element_factory_make("appsrc",NULL);
file=gst_element_factory_make("filesink", NULL);
enc=gst_element_factory_make(encit->second, NULL);
mux=gst_element_factory_make("avimux", NULL);
color = gst_element_factory_make("ffmpegcolorspace", NULL);
if (!enc)
CV_ERROR( CV_StsUnsupportedFormat, "Your version of Gstreamer doesn't support this codec acutally or needed plugin missing.");
g_object_set(G_OBJECT(file), "location", filename, NULL);
pipeline = gst_pipeline_new (NULL);
GstCaps* caps;
if (is_color) {
input_pix_fmt=1;
caps= gst_video_format_new_caps(GST_VIDEO_FORMAT_BGR,
frameSize.width,
frameSize.height,
(int) (fps * 1000),
1000,
1,
1);
}
else {
input_pix_fmt=0;
caps= gst_caps_new_simple("video/x-raw-gray",
"width", G_TYPE_INT, frameSize.width,
"height", G_TYPE_INT, frameSize.height,
"framerate", GST_TYPE_FRACTION, int(fps),1,
"bpp",G_TYPE_INT,8,
"depth",G_TYPE_INT,8,
NULL);
}
gst_app_src_set_caps(GST_APP_SRC(source), caps);
if (fourcc==CV_FOURCC_DEFAULT) {
gst_bin_add_many(GST_BIN(pipeline), source, color,mux, file, NULL);
if(!gst_element_link_many(source,color,enc,mux,file,NULL)) {
CV_ERROR(CV_StsError, "GStreamer: cannot link elements\n");
}
}
else {
gst_bin_add_many(GST_BIN(pipeline), source, color,enc,mux, file, NULL);
if(!gst_element_link_many(source,color,enc,mux,file,NULL)) {
CV_ERROR(CV_StsError, "GStreamer: cannot link elements\n");
}
}
if(gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING) ==
GST_STATE_CHANGE_FAILURE) {
CV_ERROR(CV_StsError, "GStreamer: cannot put pipeline to play\n");
}
__END__;
return true;
}
int
main (int argc, char *argv[])
{
GstElement *filesrc, *osssink, *queue, *parse, *decode;
GstElement *bin;
GstElement *thread;
gst_init (&argc, &argv);
if (argc != 2) {
g_print ("usage: %s <filename>\n", argv[0]);
exit (-1);
}
/* create a new thread to hold the elements */
thread = gst_thread_new ("thread");
g_assert (thread != NULL);
/* create a new bin to hold the elements */
bin = gst_bin_new ("bin");
g_assert (bin != NULL);
/* create a disk reader */
filesrc = gst_element_factory_make ("filesrc", "disk_source");
g_assert (filesrc != NULL);
g_object_set (G_OBJECT (filesrc), "location", argv[1], NULL);
g_signal_connect (G_OBJECT (filesrc), "eos", G_CALLBACK (eos), thread);
queue = gst_element_factory_make ("queue", "queue");
/* and an audio sink */
osssink = gst_element_factory_make ("osssink", "play_audio");
g_assert (osssink != NULL);
parse = gst_element_factory_make ("mp3parse", "parse");
decode = gst_element_factory_make ("mpg123", "decode");
/* add objects to the main bin */
gst_bin_add (GST_BIN (bin), filesrc);
gst_bin_add (GST_BIN (bin), queue);
gst_bin_add (GST_BIN (thread), parse);
gst_bin_add (GST_BIN (thread), decode);
gst_bin_add (GST_BIN (thread), osssink);
gst_element_link_many (filesrc, queue, parse, decode, osssink, NULL);
/* make it ready */
gst_element_set_state (GST_ELEMENT (bin), GST_STATE_READY);
/* start playing */
gst_element_set_state (GST_ELEMENT (bin), GST_STATE_PLAYING);
playing = TRUE;
while (playing) {
gst_bin_iterate (GST_BIN (bin));
}
gst_element_set_state (GST_ELEMENT (bin), GST_STATE_NULL);
exit (0);
}
static GstFlowReturn
daala_handle_data_packet (GstDaalaDec * dec, ogg_packet * packet,
GstVideoCodecFrame * frame)
{
/* normal data packet */
od_img img;
gboolean keyframe;
GstFlowReturn result;
if (G_UNLIKELY (!dec->have_header))
goto not_initialized;
/* the second most significant bit of the first data byte is cleared
* for keyframes. We can only check it if it's not a zero-length packet. */
keyframe = packet->bytes && ((packet->packet[0] & 0x40));
if (G_UNLIKELY (keyframe)) {
GST_DEBUG_OBJECT (dec, "we have a keyframe");
dec->need_keyframe = FALSE;
} else if (G_UNLIKELY (dec->need_keyframe)) {
goto dropping;
}
GST_DEBUG_OBJECT (dec, "parsing data packet");
/* this does the decoding */
if (G_UNLIKELY (daala_decode_packet_in (dec->decoder, &img, packet) < 0))
goto decode_error;
if (frame &&
(gst_video_decoder_get_max_decode_time (GST_VIDEO_DECODER (dec),
frame) < 0))
goto dropping_qos;
if (G_UNLIKELY ((img.width != dec->info.pic_width
|| img.height != dec->info.pic_height)))
goto wrong_dimensions;
result = daala_handle_image (dec, &img, frame);
return result;
/* ERRORS */
not_initialized:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("no header sent yet"));
return GST_FLOW_ERROR;
}
dropping:
{
GST_WARNING_OBJECT (dec, "dropping frame because we need a keyframe");
return GST_CUSTOM_FLOW_DROP;
}
dropping_qos:
{
GST_WARNING_OBJECT (dec, "dropping frame because of QoS");
return GST_CUSTOM_FLOW_DROP;
}
decode_error:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("daala decoder did not decode data packet"));
return GST_FLOW_ERROR;
}
wrong_dimensions:
{
GST_ELEMENT_ERROR (GST_ELEMENT (dec), STREAM, FORMAT,
(NULL), ("dimensions of image do not match header"));
return GST_FLOW_ERROR;
}
}
bool VideoSender::enableSending(bool enable)
{
GstElement *sink;
#define USE_TEE 0
#if USE_TEE
GstElement *ob;
#endif
GError *error = NULL;
qDebug() << "In" << __FUNCTION__ << ", Enable:" << enable;
// Disable video sending
if (enable == false) {
qDebug() << "Stopping video encoding";
if (pipeline) {
gst_element_set_state(pipeline, GST_STATE_NULL);
}
qDebug() << "Deleting pipeline";
if (pipeline) {
gst_object_unref(GST_OBJECT(pipeline));
pipeline = NULL;
}
encoder = NULL;
ODdata[OB_VIDEO_PARAM_CONTINUE] = 0;
if (ODprocess) {
ODprocess->write((const char *)ODdata, sizeof(ODdata));
}
return true;
}
if (pipeline) {
// Do nothing as the pipeline has already been created and is
// probably running
qCritical("Pipeline exists already, doing nothing");
return true;
}
// Initialisation. We don't pass command line arguments here
if (!gst_init_check(NULL, NULL, NULL)) {
qCritical("Failed to init GST");
return false;
}
if (!hardware) {
qCritical("No hardware plugin");
return false;
}
QString pipelineString = "";
pipelineString.append(videoSource + " name=source");
pipelineString.append(" ! ");
pipelineString.append("capsfilter caps=\"video/x-raw,format=(string)I420,framerate=(fraction)30/1,");
switch(quality) {
default:
case 0:
pipelineString.append("width=(int)320,height=(int)240");
break;
case 1:
pipelineString.append("width=(int)640,height=(int)480");
break;
case 2:
pipelineString.append("width=(int)800,height=(int)600");
break;
}
pipelineString.append("\"");
#if USE_TEE
pipelineString.append(" ! ");
pipelineString.append("tee name=scripttee");
// FIXME: does this case latency?
pipelineString.append(" ! ");
pipelineString.append("queue");
#endif
pipelineString.append(" ! ");
pipelineString.append(hardware->getEncodingPipeline());
pipelineString.append(" ! ");
pipelineString.append("rtph264pay name=rtppay config-interval=1 mtu=500");
pipelineString.append(" ! ");
pipelineString.append("appsink name=sink sync=false max-buffers=1 drop=true");
#if USE_TEE
// Tee (branch) frames for external components
pipelineString.append(" scripttee. ");
// TODO: downscale to 320x240?
pipelineString.append(" ! ");
pipelineString.append("appsink name=ob sync=false max-buffers=1 drop=true");
#endif
qDebug() << "Using pipeline:" << pipelineString;
// Create encoding video pipeline
pipeline = gst_parse_launch(pipelineString.toUtf8(), &error);
if (!pipeline) {
qCritical("Failed to parse pipeline: %s", error->message);
g_error_free(error);
return false;
}
encoder = gst_bin_get_by_name(GST_BIN(pipeline), "encoder");
if (!encoder) {
qCritical("Failed to get encoder");
return false;
}
// Assuming here that X86 uses x264enc
if (hardware->getHardwareName() == "generic_x86") {
g_object_set(G_OBJECT(encoder), "speed-preset", 1, NULL); // ultrafast
g_object_set(G_OBJECT(encoder), "tune", 0x00000004, NULL); // zerolatency
}
if (hardware->getHardwareName() == "tegrak1" ||
hardware->getHardwareName() == "tegrax1") {
//g_object_set(G_OBJECT(encoder), "input-buffers", 2, NULL); // not valid on 1.0
//g_object_set(G_OBJECT(encoder), "output-buffers", 2, NULL); // not valid on 1.0
//g_object_set(G_OBJECT(encoder), "quality-level", 0, NULL);
//g_object_set(G_OBJECT(encoder), "rc-mode", 0, NULL);
}
if (hardware->getHardwareName() == "tegrax2") {
g_object_set(G_OBJECT(encoder), "preset-level", 0, NULL); // 0 == UltraFastPreset for high perf
}
setBitrate(bitrate);
{
GstElement *source;
source = gst_bin_get_by_name(GST_BIN(pipeline), "source");
if (!source) {
qCritical("Failed to get source");
return false;
}
g_object_set(G_OBJECT(source), "do-timestamp", true, NULL);
if (videoSource == "videotestsrc") {
g_object_set(G_OBJECT(source), "is-live", true, NULL);
} else if (videoSource == "v4l2src") {
//g_object_set(G_OBJECT(source), "always-copy", false, NULL);
const char *camera = "/dev/video0";
QByteArray env_camera = qgetenv("PLECO_SLAVE_CAMERA");
if (!env_camera.isNull()) {
camera = env_camera.data();
}
g_object_set(G_OBJECT(source), "device", camera, NULL);
}
if (hardware->getHardwareName() == "tegrak1" ||
hardware->getHardwareName() == "tegrax1") {
g_object_set(G_OBJECT(source), "io-mode", 1, NULL);
}
}
sink = gst_bin_get_by_name(GST_BIN(pipeline), "sink");
if (!sink) {
qCritical("Failed to get sink");
return false;
}
// Set appsink callbacks
GstAppSinkCallbacks appSinkCallbacks;
appSinkCallbacks.eos = NULL;
appSinkCallbacks.new_preroll = NULL;
appSinkCallbacks.new_sample = &newBufferCB;
gst_app_sink_set_callbacks(GST_APP_SINK(sink), &appSinkCallbacks, this, NULL);
#if USE_TEE
// Callbacks for the OB process appsink
ob = gst_bin_get_by_name(GST_BIN(pipeline), "ob");
if (!ob) {
qCritical("Failed to get ob appsink");
return false;
}
// Set appsink callbacks
GstAppSinkCallbacks obCallbacks;
obCallbacks.eos = NULL;
obCallbacks.new_preroll = NULL;
obCallbacks.new_sample = &newBufferOBCB;
gst_app_sink_set_callbacks(GST_APP_SINK(ob), &obCallbacks, this, NULL);
#endif
// Start running
gst_element_set_state(GST_ELEMENT(pipeline), GST_STATE_PLAYING);
launchObjectDetection();
return true;
}