static void
gst_videorate_update_duration (GstVideoRate * videorate)
{
GstMessage *m;
m = gst_message_new_duration_changed (GST_OBJECT (videorate));
gst_element_post_message (GST_ELEMENT (videorate), m);
}
static void
avrcp_metadata_cb (GstAvrcpConnection * avrcp, GstTagList * taglist,
gpointer user_data)
{
GstAvdtpSrc *src = GST_AVDTP_SRC (user_data);
guint64 duration;
if (gst_tag_list_get_uint64 (taglist, GST_TAG_DURATION, &duration)) {
src->duration = duration;
gst_element_post_message (GST_ELEMENT (src),
gst_message_new_duration_changed (GST_OBJECT (src)));
}
gst_pad_push_event (GST_BASE_SRC_PAD (src),
gst_event_new_tag (gst_tag_list_copy (taglist)));
gst_element_post_message (GST_ELEMENT (src),
gst_message_new_tag (GST_OBJECT (src), taglist));
}
static GstFlowReturn
gst_app_src_create (GstBaseSrc * bsrc, guint64 offset, guint size,
GstBuffer ** buf)
{
GstAppSrc *appsrc = GST_APP_SRC_CAST (bsrc);
GstAppSrcPrivate *priv = appsrc->priv;
GstFlowReturn ret;
GST_OBJECT_LOCK (appsrc);
if (G_UNLIKELY (priv->size != bsrc->segment.duration &&
bsrc->segment.format == GST_FORMAT_BYTES)) {
GST_DEBUG_OBJECT (appsrc,
"Size changed from %" G_GINT64_FORMAT " to %" G_GINT64_FORMAT,
bsrc->segment.duration, priv->size);
bsrc->segment.duration = priv->size;
GST_OBJECT_UNLOCK (appsrc);
gst_element_post_message (GST_ELEMENT (appsrc),
gst_message_new_duration_changed (GST_OBJECT (appsrc)));
} else {
GST_OBJECT_UNLOCK (appsrc);
}
g_mutex_lock (&priv->mutex);
/* check flushing first */
if (G_UNLIKELY (priv->flushing))
goto flushing;
if (priv->stream_type == GST_APP_STREAM_TYPE_RANDOM_ACCESS) {
/* if we are dealing with a random-access stream, issue a seek if the offset
* changed. */
if (G_UNLIKELY (priv->offset != offset)) {
gboolean res;
/* do the seek */
res = gst_app_src_emit_seek (appsrc, offset);
if (G_UNLIKELY (!res))
/* failing to seek is fatal */
goto seek_error;
priv->offset = offset;
priv->is_eos = FALSE;
}
}
while (TRUE) {
/* return data as long as we have some */
if (!g_queue_is_empty (priv->queue)) {
guint buf_size;
if (priv->new_caps) {
gst_app_src_do_negotiate (bsrc);
priv->new_caps = FALSE;
}
*buf = g_queue_pop_head (priv->queue);
buf_size = gst_buffer_get_size (*buf);
GST_DEBUG_OBJECT (appsrc, "we have buffer %p of size %u", *buf, buf_size);
priv->queued_bytes -= buf_size;
/* only update the offset when in random_access mode */
if (priv->stream_type == GST_APP_STREAM_TYPE_RANDOM_ACCESS)
priv->offset += buf_size;
/* signal that we removed an item */
g_cond_broadcast (&priv->cond);
/* see if we go lower than the empty-percent */
if (priv->min_percent && priv->max_bytes) {
if (priv->queued_bytes * 100 / priv->max_bytes <= priv->min_percent)
/* ignore flushing state, we got a buffer and we will return it now.
* Errors will be handled in the next round */
gst_app_src_emit_need_data (appsrc, size);
}
ret = GST_FLOW_OK;
break;
} else {
gst_app_src_emit_need_data (appsrc, size);
/* we can be flushing now because we released the lock above */
if (G_UNLIKELY (priv->flushing))
goto flushing;
/* if we have a buffer now, continue the loop and try to return it. In
* random-access mode (where a buffer is normally pushed in the above
* signal) we can still be empty because the pushed buffer got flushed or
* when the application pushes the requested buffer later, we support both
* possibilities. */
if (!g_queue_is_empty (priv->queue))
continue;
/* no buffer yet, maybe we are EOS, if not, block for more data. */
}
/* check EOS */
if (G_UNLIKELY (priv->is_eos))
goto eos;
/* nothing to return, wait a while for new data or flushing. */
g_cond_wait (&priv->cond, &priv->mutex);
}
g_mutex_unlock (&priv->mutex);
return ret;
/* ERRORS */
flushing:
{
GST_DEBUG_OBJECT (appsrc, "we are flushing");
g_mutex_unlock (&priv->mutex);
return GST_FLOW_FLUSHING;
}
eos:
{
GST_DEBUG_OBJECT (appsrc, "we are EOS");
g_mutex_unlock (&priv->mutex);
return GST_FLOW_EOS;
}
seek_error:
{
g_mutex_unlock (&priv->mutex);
GST_ELEMENT_ERROR (appsrc, RESOURCE, READ, ("failed to seek"),
GST_ERROR_SYSTEM);
return GST_FLOW_ERROR;
}
}
static GstFlowReturn
gst_test_http_src_create (GstBaseSrc * basesrc, guint64 offset,
guint length, GstBuffer ** retbuf)
{
GstTestHTTPSrc *src = GST_TEST_HTTP_SRC (basesrc);
guint bytes_read;
GstFlowReturn ret = GST_FLOW_OK;
guint blocksize;
fail_unless (gst_test_http_src_callbacks != NULL);
fail_unless (gst_test_http_src_callbacks->src_create != NULL);
GST_OBJECT_LOCK (src);
blocksize = basesrc->blocksize;
GST_OBJECT_UNLOCK (src);
g_mutex_lock (&src->mutex);
GST_DEBUG ("gst_test_http_src_create feeding from %" G_GUINT64_FORMAT,
src->position);
if (src->uri == NULL) {
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), GST_ERROR_SYSTEM);
g_mutex_unlock (&src->mutex);
return GST_FLOW_ERROR;
}
if (src->input.status_code < 200 || src->input.status_code >= 300) {
GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND, ("%s",
"Generated requested error"), ("%s (%d), URL: %s, Redirect to: %s",
"Generated requested error", src->input.status_code, src->uri,
GST_STR_NULL (NULL)));
g_mutex_unlock (&src->mutex);
return GST_FLOW_ERROR;
}
if (src->http_method == METHOD_INVALID) {
GST_ELEMENT_ERROR (src, RESOURCE, READ, ("%s",
"Invalid HTTP method"), ("%s (%s), URL: %s",
"Invalid HTTP method", src->http_method_name, src->uri));
g_mutex_unlock (&src->mutex);
return GST_FLOW_ERROR;
} else if (src->http_method == METHOD_HEAD) {
ret = GST_FLOW_EOS;
goto http_events;
}
fail_unless_equals_uint64 (offset, src->position);
bytes_read = MIN ((src->segment_end - src->position), blocksize);
if (bytes_read == 0) {
ret = GST_FLOW_EOS;
goto http_events;
}
ret = gst_test_http_src_callbacks->src_create (src,
offset, bytes_read, retbuf,
src->input.context, gst_test_http_src_callback_user_data);
if (ret != GST_FLOW_OK) {
goto http_events;
}
GST_BUFFER_OFFSET (*retbuf) = src->position;
GST_BUFFER_OFFSET_END (*retbuf) = src->position + bytes_read;
src->position += bytes_read;
http_events:
if (src->http_headers_event) {
gst_pad_push_event (GST_BASE_SRC_PAD (src), src->http_headers_event);
src->http_headers_event = NULL;
}
if (src->duration_changed) {
src->duration_changed = FALSE;
gst_element_post_message (GST_ELEMENT (src),
gst_message_new_duration_changed (GST_OBJECT (src)));
}
g_mutex_unlock (&src->mutex);
return ret;
}
void PlaybackPipeline::notifyDurationChanged()
{
gst_element_post_message(GST_ELEMENT(m_webKitMediaSrc.get()), gst_message_new_duration_changed(GST_OBJECT(m_webKitMediaSrc.get())));
// WebKitMediaSrc will ask MediaPlayerPrivateGStreamerMSE for the new duration later, when somebody asks for it.
}
