/* FIXME, the duration query should reflect how long you will produce
* data, that is the amount of stream time until you will emit EOS.
*
* For synchronized mixing this is always the max of all the durations
* of upstream since we emit EOS when all of them finished.
*
* We don't do synchronized mixing so this really depends on where the
* streams where punched in and what their relative offsets are against
* eachother which we can get from the first timestamps we see.
*
* When we add a new stream (or remove a stream) the duration might
* also become invalid again and we need to post a new DURATION
* message to notify this fact to the parent.
* For now we take the max of all the upstream elements so the simple
* cases work at least somewhat.
*/
static gboolean
gst_audio_aggregator_query_duration (GstAudioAggregator * aagg,
GstQuery * query)
{
gint64 max;
gboolean res;
GstFormat format;
GstIterator *it;
gboolean done;
GValue item = { 0, };
/* parse format */
gst_query_parse_duration (query, &format, NULL);
max = -1;
res = TRUE;
done = FALSE;
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (aagg));
while (!done) {
GstIteratorResult ires;
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
GstPad *pad = g_value_get_object (&item);
gint64 duration;
/* ask sink peer for duration */
res &= gst_pad_peer_query_duration (pad, format, &duration);
/* take max from all valid return values */
if (res) {
/* valid unknown length, stop searching */
if (duration == -1) {
max = duration;
done = TRUE;
}
/* else see if bigger than current max */
else if (duration > max)
max = duration;
}
g_value_reset (&item);
break;
}
case GST_ITERATOR_RESYNC:
max = -1;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
g_value_unset (&item);
gst_iterator_free (it);
if (res) {
/* and store the max */
GST_DEBUG_OBJECT (aagg, "Total duration in format %s: %"
GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
gst_query_set_duration (query, format, max);
}
return res;
}
static void
gst_raw_parse_loop (GstElement * element)
{
GstRawParse *rp = GST_RAW_PARSE (element);
GstRawParseClass *rp_class = GST_RAW_PARSE_GET_CLASS (rp);
GstFlowReturn ret;
GstBuffer *buffer;
gint size;
if (G_UNLIKELY (rp->push_stream_start)) {
gchar *stream_id;
GstEvent *event;
stream_id =
gst_pad_create_stream_id (rp->srcpad, GST_ELEMENT_CAST (rp), NULL);
event = gst_event_new_stream_start (stream_id);
gst_event_set_group_id (event, gst_util_group_id_next ());
GST_DEBUG_OBJECT (rp, "Pushing STREAM_START");
gst_pad_push_event (rp->srcpad, event);
rp->push_stream_start = FALSE;
g_free (stream_id);
}
if (!gst_raw_parse_set_src_caps (rp))
goto no_caps;
if (rp->start_segment) {
GST_DEBUG_OBJECT (rp, "sending start segment");
gst_pad_push_event (rp->srcpad, rp->start_segment);
rp->start_segment = NULL;
}
if (rp_class->multiple_frames_per_buffer && rp->framesize < 4096)
size = 4096 - (4096 % rp->framesize);
else
size = rp->framesize;
if (rp->segment.rate >= 0) {
if (rp->offset + size > rp->upstream_length) {
GstFormat fmt = GST_FORMAT_BYTES;
if (!gst_pad_peer_query_duration (rp->sinkpad, fmt, &rp->upstream_length)) {
GST_WARNING_OBJECT (rp,
"Could not get upstream duration, trying to pull frame by frame");
size = rp->framesize;
} else if (rp->upstream_length < rp->offset + rp->framesize) {
ret = GST_FLOW_EOS;
goto pause;
} else if (rp->offset + size > rp->upstream_length) {
size = rp->upstream_length - rp->offset;
size -= size % rp->framesize;
}
}
} else {
if (rp->offset == 0) {
ret = GST_FLOW_EOS;
goto pause;
} else if (rp->offset < size) {
size -= rp->offset;
}
rp->offset -= size;
}
buffer = NULL;
ret = gst_pad_pull_range (rp->sinkpad, rp->offset, size, &buffer);
if (ret != GST_FLOW_OK) {
GST_DEBUG_OBJECT (rp, "pull_range (%" G_GINT64_FORMAT ", %u) "
"failed, flow: %s", rp->offset, size, gst_flow_get_name (ret));
buffer = NULL;
goto pause;
}
if (gst_buffer_get_size (buffer) < size) {
GST_DEBUG_OBJECT (rp, "Short read at offset %" G_GINT64_FORMAT
", got only %" G_GSIZE_FORMAT " of %u bytes", rp->offset,
gst_buffer_get_size (buffer), size);
if (size > rp->framesize) {
gst_buffer_set_size (buffer, gst_buffer_get_size (buffer) -
gst_buffer_get_size (buffer) % rp->framesize);
} else {
gst_buffer_unref (buffer);
buffer = NULL;
ret = GST_FLOW_EOS;
goto pause;
}
}
ret = gst_raw_parse_push_buffer (rp, buffer);
if (ret != GST_FLOW_OK)
goto pause;
return;
/* ERRORS */
no_caps:
{
GST_ERROR_OBJECT (rp, "could not negotiate caps");
ret = GST_FLOW_NOT_NEGOTIATED;
goto pause;
}
pause:
{
const gchar *reason = gst_flow_get_name (ret);
GST_LOG_OBJECT (rp, "pausing task, reason %s", reason);
gst_pad_pause_task (rp->sinkpad);
if (ret == GST_FLOW_EOS) {
if (rp->segment.flags & GST_SEEK_FLAG_SEGMENT) {
GstClockTime stop;
GST_LOG_OBJECT (rp, "Sending segment done");
if ((stop = rp->segment.stop) == -1)
stop = rp->segment.duration;
gst_element_post_message (GST_ELEMENT_CAST (rp),
gst_message_new_segment_done (GST_OBJECT_CAST (rp),
rp->segment.format, stop));
gst_pad_push_event (rp->srcpad,
gst_event_new_segment_done (rp->segment.format, stop));
} else {
GST_LOG_OBJECT (rp, "Sending EOS, at end of stream");
gst_pad_push_event (rp->srcpad, gst_event_new_eos ());
}
} else if (ret == GST_FLOW_NOT_LINKED || ret < GST_FLOW_EOS) {
GST_ELEMENT_ERROR (rp, STREAM, FAILED,
("Internal data stream error."),
("stream stopped, reason %s", reason));
gst_pad_push_event (rp->srcpad, gst_event_new_eos ());
}
return;
}
}
static gboolean
gst_hls_demux_change_playlist (GstHLSDemux * demux, guint max_bitrate,
gboolean * changed)
{
GList *previous_variant, *current_variant;
gint old_bandwidth, new_bandwidth;
GstAdaptiveDemux *adaptive_demux = GST_ADAPTIVE_DEMUX_CAST (demux);
GstAdaptiveDemuxStream *stream;
g_return_val_if_fail (adaptive_demux->streams != NULL, FALSE);
stream = adaptive_demux->streams->data;
previous_variant = demux->client->main->current_variant;
current_variant = gst_m3u8_client_get_playlist_for_bitrate (demux->client,
max_bitrate);
GST_M3U8_CLIENT_LOCK (demux->client);
retry_failover_protection:
old_bandwidth = GST_M3U8 (previous_variant->data)->bandwidth;
new_bandwidth = GST_M3U8 (current_variant->data)->bandwidth;
/* Don't do anything else if the playlist is the same */
if (new_bandwidth == old_bandwidth) {
GST_M3U8_CLIENT_UNLOCK (demux->client);
return TRUE;
}
demux->client->main->current_variant = current_variant;
GST_M3U8_CLIENT_UNLOCK (demux->client);
gst_m3u8_client_set_current (demux->client, current_variant->data);
GST_INFO_OBJECT (demux, "Client was on %dbps, max allowed is %dbps, switching"
" to bitrate %dbps", old_bandwidth, max_bitrate, new_bandwidth);
stream->discont = TRUE;
if (gst_hls_demux_update_playlist (demux, FALSE, NULL)) {
gchar *uri;
gchar *main_uri;
uri = gst_m3u8_client_get_current_uri (demux->client);
main_uri = gst_m3u8_client_get_uri (demux->client);
gst_element_post_message (GST_ELEMENT_CAST (demux),
gst_message_new_element (GST_OBJECT_CAST (demux),
gst_structure_new (GST_ADAPTIVE_DEMUX_STATISTICS_MESSAGE_NAME,
"manifest-uri", G_TYPE_STRING,
main_uri, "uri", G_TYPE_STRING,
uri, "bitrate", G_TYPE_INT, new_bandwidth, NULL)));
g_free (uri);
g_free (main_uri);
if (changed)
*changed = TRUE;
} else {
GList *failover = NULL;
GST_INFO_OBJECT (demux, "Unable to update playlist. Switching back");
GST_M3U8_CLIENT_LOCK (demux->client);
failover = g_list_previous (current_variant);
if (failover && new_bandwidth == GST_M3U8 (failover->data)->bandwidth) {
current_variant = failover;
goto retry_failover_protection;
}
demux->client->main->current_variant = previous_variant;
GST_M3U8_CLIENT_UNLOCK (demux->client);
gst_m3u8_client_set_current (demux->client, previous_variant->data);
/* Try a lower bitrate (or stop if we just tried the lowest) */
if (GST_M3U8 (previous_variant->data)->iframe && new_bandwidth ==
GST_M3U8 (g_list_first (demux->client->main->iframe_lists)->data)->
bandwidth)
return FALSE;
else if (!GST_M3U8 (previous_variant->data)->iframe && new_bandwidth ==
GST_M3U8 (g_list_first (demux->client->main->lists)->data)->bandwidth)
return FALSE;
else
return gst_hls_demux_change_playlist (demux, new_bandwidth - 1, changed);
}
/* Force typefinding since we might have changed media type */
demux->do_typefind = TRUE;
return TRUE;
}
static gboolean
mpegtsmux_src_event (GstPad * pad, GstEvent * event)
{
MpegTsMux *mux = GST_MPEG_TSMUX (gst_pad_get_parent (pad));
gboolean res = TRUE;
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CUSTOM_UPSTREAM:
{
GstIterator *iter;
GstIteratorResult iter_ret;
GstPad *sinkpad;
GstClockTime running_time;
gboolean all_headers, done;
guint count;
if (!gst_video_event_is_force_key_unit (event))
break;
gst_video_event_parse_upstream_force_key_unit (event,
&running_time, &all_headers, &count);
GST_INFO_OBJECT (mux, "received upstream force-key-unit event, "
"seqnum %d running_time %" GST_TIME_FORMAT " all_headers %d count %d",
gst_event_get_seqnum (event), GST_TIME_ARGS (running_time),
all_headers, count);
if (!all_headers)
break;
mux->pending_key_unit_ts = running_time;
gst_event_replace (&mux->force_key_unit_event, event);
iter = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (mux));
done = FALSE;
while (!done) {
gboolean res = FALSE, tmp;
iter_ret = gst_iterator_next (iter, (gpointer *) & sinkpad);
switch (iter_ret) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
GST_INFO_OBJECT (mux, "forwarding to %s",
gst_pad_get_name (sinkpad));
tmp = gst_pad_push_event (sinkpad, gst_event_ref (event));
GST_INFO_OBJECT (mux, "result %d", tmp);
/* succeed if at least one pad succeeds */
res |= tmp;
gst_object_unref (sinkpad);
break;
case GST_ITERATOR_ERROR:
done = TRUE;
break;
case GST_ITERATOR_RESYNC:
break;
}
}
gst_event_unref (event);
break;
}
default:
res = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (mux);
return res;
}
static void
paused_mode_task (gpointer data)
{
GstMimEnc *mimenc = GST_MIMENC (data);
GstClockTime now;
GstClockTimeDiff diff;
GstFlowReturn ret;
if (!GST_ELEMENT_CLOCK (mimenc)) {
GST_ERROR_OBJECT (mimenc, "Element has no clock");
gst_pad_pause_task (mimenc->srcpad);
return;
}
GST_OBJECT_LOCK (mimenc);
if (mimenc->stop_paused_mode) {
GST_OBJECT_UNLOCK (mimenc);
goto stop_task;
}
now = gst_clock_get_time (GST_ELEMENT_CLOCK (mimenc));
diff = now - GST_ELEMENT_CAST (mimenc)->base_time - mimenc->last_buffer;
if (diff < 0)
diff = 0;
if (diff > 3.95 * GST_SECOND) {
GstBuffer *buffer = gst_mimenc_create_tcp_header (mimenc, 0,
mimenc->last_buffer + 4 * GST_SECOND, FALSE, TRUE);
GstEvent *event = NULL;
mimenc->last_buffer += 4 * GST_SECOND;
if (mimenc->need_newsegment) {
event = gst_event_new_new_segment (FALSE, 1.0, GST_FORMAT_TIME, 0, -1, 0);
mimenc->need_newsegment = FALSE;
}
GST_OBJECT_UNLOCK (mimenc);
GST_LOG_OBJECT (mimenc, "Haven't had an incoming buffer in 4 seconds,"
" sending out a pause frame");
if (event) {
if (!gst_pad_push_event (mimenc->srcpad, event))
GST_WARNING_OBJECT (mimenc, "Failed to push NEWSEGMENT event");
}
ret = gst_pad_push (mimenc->srcpad, buffer);
if (ret < 0) {
GST_WARNING_OBJECT (mimenc, "Error pushing paused header: %s",
gst_flow_get_name (ret));
goto stop_task;
}
} else {
GstClockTime next_stop;
GstClockID id;
next_stop = now + (4 * GST_SECOND - MIN (diff, 4 * GST_SECOND));
id = gst_clock_new_single_shot_id (GST_ELEMENT_CLOCK (mimenc), next_stop);
if (mimenc->stop_paused_mode) {
GST_OBJECT_UNLOCK (mimenc);
goto stop_task;
}
mimenc->clock_id = id;
GST_OBJECT_UNLOCK (mimenc);
gst_clock_id_wait (id, NULL);
GST_OBJECT_LOCK (mimenc);
mimenc->clock_id = NULL;
GST_OBJECT_UNLOCK (mimenc);
gst_clock_id_unref (id);
}
return;
stop_task:
gst_pad_pause_task (mimenc->srcpad);
}
static GstStateChangeReturn
gst_mimenc_change_state (GstElement * element, GstStateChange transition)
{
GstMimEnc *mimenc = GST_MIMENC (element);
GstStateChangeReturn ret;
gboolean paused_mode;
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
GST_OBJECT_LOCK (element);
if (mimenc->enc != NULL) {
mimic_close (mimenc->enc);
mimenc->enc = NULL;
mimenc->buffer_size = -1;
mimenc->frames = 0;
}
GST_OBJECT_UNLOCK (element);
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_OBJECT_LOCK (mimenc);
gst_segment_init (&mimenc->segment, GST_FORMAT_UNDEFINED);
mimenc->last_buffer = GST_CLOCK_TIME_NONE;
mimenc->need_newsegment = TRUE;
GST_OBJECT_UNLOCK (mimenc);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
GST_OBJECT_LOCK (mimenc);
if (mimenc->clock_id)
gst_clock_id_unschedule (mimenc->clock_id);
mimenc->stop_paused_mode = TRUE;
GST_OBJECT_UNLOCK (mimenc);
gst_pad_pause_task (mimenc->srcpad);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_OBJECT_LOCK (mimenc);
mimenc->stop_paused_mode = FALSE;
paused_mode = mimenc->paused_mode;
if (paused_mode) {
if (!GST_ELEMENT_CLOCK (mimenc)) {
GST_OBJECT_UNLOCK (mimenc);
GST_ELEMENT_ERROR (mimenc, RESOURCE, FAILED,
("Using paused-mode requires a clock, but no clock was provided"
" to the element"), (NULL));
return GST_STATE_CHANGE_FAILURE;
}
if (mimenc->last_buffer == GST_CLOCK_TIME_NONE)
mimenc->last_buffer = gst_clock_get_time (GST_ELEMENT_CLOCK (mimenc))
- GST_ELEMENT_CAST (mimenc)->base_time;
}
GST_OBJECT_UNLOCK (mimenc);
if (paused_mode) {
if (!gst_pad_start_task (mimenc->srcpad, paused_mode_task, mimenc)) {
ret = GST_STATE_CHANGE_FAILURE;
GST_ERROR_OBJECT (mimenc, "Can not start task");
}
}
break;
default:
break;
}
return ret;
}
static GstStateChangeReturn
gst_decklink_audio_src_change_state (GstElement * element,
GstStateChange transition)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (element);
GstStateChangeReturn ret;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!gst_decklink_audio_src_open (self)) {
ret = GST_STATE_CHANGE_FAILURE;
goto out;
}
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:{
GstElement *videosrc = NULL;
// Check if there is a video src for this input too and if it
// is actually in the same pipeline
g_mutex_lock (&self->input->lock);
if (self->input->videosrc)
videosrc = GST_ELEMENT_CAST (gst_object_ref (self->input->videosrc));
g_mutex_unlock (&self->input->lock);
if (!videosrc) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL), ("Audio src needs a video src for its operation"));
ret = GST_STATE_CHANGE_FAILURE;
goto out;
}
// FIXME: This causes deadlocks sometimes
#if 0
else if (!in_same_pipeline (GST_ELEMENT_CAST (self), videosrc)) {
GST_ELEMENT_ERROR (self, STREAM, FAILED,
(NULL),
("Audio src and video src need to be in the same pipeline"));
ret = GST_STATE_CHANGE_FAILURE;
gst_object_unref (videosrc);
goto out;
}
#endif
if (videosrc)
gst_object_unref (videosrc);
self->flushing = FALSE;
self->next_offset = -1;
break;
}
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
if (ret == GST_STATE_CHANGE_FAILURE)
return ret;
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_decklink_audio_src_stop (self);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
gst_decklink_audio_src_close (self);
break;
default:
break;
}
out:
return ret;
}
static GstFlowReturn
gst_udpsrc_create (GstPushSrc * psrc, GstBuffer ** buf)
{
GstUDPSrc *udpsrc;
GstBuffer *outbuf = NULL;
GSocketAddress *saddr = NULL;
GSocketAddress **p_saddr;
gint flags = G_SOCKET_MSG_NONE;
gboolean try_again;
GError *err = NULL;
gssize res;
gsize offset;
udpsrc = GST_UDPSRC_CAST (psrc);
if (!gst_udpsrc_ensure_mem (udpsrc))
goto memory_alloc_error;
/* Retrieve sender address unless we've been configured not to do so */
p_saddr = (udpsrc->retrieve_sender_address) ? &saddr : NULL;
retry:
do {
gint64 timeout;
try_again = FALSE;
if (udpsrc->timeout)
timeout = udpsrc->timeout / 1000;
else
timeout = -1;
GST_LOG_OBJECT (udpsrc, "doing select, timeout %" G_GINT64_FORMAT, timeout);
if (!g_socket_condition_timed_wait (udpsrc->used_socket, G_IO_IN | G_IO_PRI,
timeout, udpsrc->cancellable, &err)) {
if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_BUSY)
|| g_error_matches (err, G_IO_ERROR, G_IO_ERROR_CANCELLED)) {
goto stopped;
} else if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_TIMED_OUT)) {
g_clear_error (&err);
/* timeout, post element message */
gst_element_post_message (GST_ELEMENT_CAST (udpsrc),
gst_message_new_element (GST_OBJECT_CAST (udpsrc),
gst_structure_new ("GstUDPSrcTimeout",
"timeout", G_TYPE_UINT64, udpsrc->timeout, NULL)));
} else {
goto select_error;
}
try_again = TRUE;
}
} while (G_UNLIKELY (try_again));
if (saddr != NULL) {
g_object_unref (saddr);
saddr = NULL;
}
res =
g_socket_receive_message (udpsrc->used_socket, p_saddr, udpsrc->vec, 2,
NULL, NULL, &flags, udpsrc->cancellable, &err);
if (G_UNLIKELY (res < 0)) {
/* G_IO_ERROR_HOST_UNREACHABLE for a UDP socket means that a packet sent
* with udpsink generated a "port unreachable" ICMP response. We ignore
* that and try again.
* On Windows we get G_IO_ERROR_CONNECTION_CLOSED instead */
#if GLIB_CHECK_VERSION(2,44,0)
if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_HOST_UNREACHABLE) ||
g_error_matches (err, G_IO_ERROR, G_IO_ERROR_CONNECTION_CLOSED)) {
#else
if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_HOST_UNREACHABLE)) {
#endif
g_clear_error (&err);
goto retry;
}
goto receive_error;
}
/* remember maximum packet size */
if (res > udpsrc->max_size)
udpsrc->max_size = res;
outbuf = gst_buffer_new ();
/* append first memory chunk to buffer */
gst_buffer_append_memory (outbuf, udpsrc->mem);
/* if the packet didn't fit into the first chunk, add second one as well */
if (res > udpsrc->map.size) {
gst_buffer_append_memory (outbuf, udpsrc->mem_max);
gst_memory_unmap (udpsrc->mem_max, &udpsrc->map_max);
udpsrc->vec[1].buffer = NULL;
udpsrc->vec[1].size = 0;
udpsrc->mem_max = NULL;
}
/* make sure we allocate a new chunk next time (we do this only here because
* we look at map.size to see if the second memory chunk is needed above) */
gst_memory_unmap (udpsrc->mem, &udpsrc->map);
udpsrc->vec[0].buffer = NULL;
udpsrc->vec[0].size = 0;
udpsrc->mem = NULL;
offset = udpsrc->skip_first_bytes;
if (G_UNLIKELY (offset > 0 && res < offset))
goto skip_error;
gst_buffer_resize (outbuf, offset, res - offset);
/* use buffer metadata so receivers can also track the address */
if (saddr) {
gst_buffer_add_net_address_meta (outbuf, saddr);
g_object_unref (saddr);
saddr = NULL;
}
GST_LOG_OBJECT (udpsrc, "read packet of %d bytes", (int) res);
*buf = GST_BUFFER_CAST (outbuf);
return GST_FLOW_OK;
/* ERRORS */
memory_alloc_error:
{
GST_ELEMENT_ERROR (udpsrc, RESOURCE, READ, (NULL),
("Failed to allocate or map memory"));
return GST_FLOW_ERROR;
}
select_error:
{
GST_ELEMENT_ERROR (udpsrc, RESOURCE, READ, (NULL),
("select error: %s", err->message));
g_clear_error (&err);
return GST_FLOW_ERROR;
}
stopped:
{
GST_DEBUG ("stop called");
g_clear_error (&err);
return GST_FLOW_FLUSHING;
}
receive_error:
{
if (g_error_matches (err, G_IO_ERROR, G_IO_ERROR_BUSY) ||
g_error_matches (err, G_IO_ERROR, G_IO_ERROR_CANCELLED)) {
g_clear_error (&err);
return GST_FLOW_FLUSHING;
} else {
GST_ELEMENT_ERROR (udpsrc, RESOURCE, READ, (NULL),
("receive error %" G_GSSIZE_FORMAT ": %s", res, err->message));
g_clear_error (&err);
return GST_FLOW_ERROR;
}
}
skip_error:
{
gst_buffer_unref (outbuf);
GST_ELEMENT_ERROR (udpsrc, STREAM, DECODE, (NULL),
("UDP buffer to small to skip header"));
return GST_FLOW_ERROR;
}
}
static gboolean
gst_udpsrc_set_uri (GstUDPSrc * src, const gchar * uri, GError ** error)
{
gchar *address;
guint16 port;
if (!gst_udp_parse_uri (uri, &address, &port))
goto wrong_uri;
if (port == (guint16) - 1)
port = UDP_DEFAULT_PORT;
g_free (src->address);
src->address = address;
src->port = port;
g_free (src->uri);
src->uri = g_strdup (uri);
return TRUE;
/* ERRORS */
wrong_uri:
{
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL),
("error parsing uri %s", uri));
g_set_error_literal (error, GST_URI_ERROR, GST_URI_ERROR_BAD_URI,
"Could not parse UDP URI");
return FALSE;
}
}
static GstFlowReturn
vorbis_handle_comment_packet (GstVorbisDec * vd, ogg_packet * packet)
{
guint bitrate = 0;
gchar *encoder = NULL;
GstTagList *list, *old_list;
GstBuffer *buf;
GST_DEBUG_OBJECT (vd, "parsing comment packet");
buf = gst_buffer_new ();
GST_BUFFER_DATA (buf) = gst_ogg_packet_data (packet);
GST_BUFFER_SIZE (buf) = gst_ogg_packet_size (packet);
list =
gst_tag_list_from_vorbiscomment_buffer (buf, (guint8 *) "\003vorbis", 7,
&encoder);
old_list = vd->taglist;
vd->taglist = gst_tag_list_merge (vd->taglist, list, GST_TAG_MERGE_REPLACE);
if (old_list)
gst_tag_list_free (old_list);
gst_tag_list_free (list);
gst_buffer_unref (buf);
if (!vd->taglist) {
GST_ERROR_OBJECT (vd, "couldn't decode comments");
vd->taglist = gst_tag_list_new ();
}
if (encoder) {
if (encoder[0])
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER, encoder, NULL);
g_free (encoder);
}
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_ENCODER_VERSION, vd->vi.version,
GST_TAG_AUDIO_CODEC, "Vorbis", NULL);
if (vd->vi.bitrate_nominal > 0 && vd->vi.bitrate_nominal <= 0x7FFFFFFF) {
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_NOMINAL_BITRATE, (guint) vd->vi.bitrate_nominal, NULL);
bitrate = vd->vi.bitrate_nominal;
}
if (vd->vi.bitrate_upper > 0 && vd->vi.bitrate_upper <= 0x7FFFFFFF) {
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_MAXIMUM_BITRATE, (guint) vd->vi.bitrate_upper, NULL);
if (!bitrate)
bitrate = vd->vi.bitrate_upper;
}
if (vd->vi.bitrate_lower > 0 && vd->vi.bitrate_lower <= 0x7FFFFFFF) {
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_MINIMUM_BITRATE, (guint) vd->vi.bitrate_lower, NULL);
if (!bitrate)
bitrate = vd->vi.bitrate_lower;
}
if (bitrate) {
gst_tag_list_add (vd->taglist, GST_TAG_MERGE_REPLACE,
GST_TAG_BITRATE, (guint) bitrate, NULL);
}
if (vd->initialized) {
gst_element_found_tags_for_pad (GST_ELEMENT_CAST (vd), vd->srcpad,
vd->taglist);
vd->taglist = NULL;
} else {
/* Only post them as messages for the time being. *
* They will be pushed on the pad once the decoder is initialized */
gst_element_post_message (GST_ELEMENT_CAST (vd),
gst_message_new_tag (GST_OBJECT (vd), gst_tag_list_copy (vd->taglist)));
}
return GST_FLOW_OK;
}
static gboolean
gst_decklink_audio_src_set_caps (GstBaseSrc * bsrc, GstCaps * caps)
{
GstDecklinkAudioSrc *self = GST_DECKLINK_AUDIO_SRC_CAST (bsrc);
BMDAudioSampleType sample_depth;
GstCaps *current_caps;
HRESULT ret;
BMDAudioConnection conn = (BMDAudioConnection) - 1;
GST_DEBUG_OBJECT (self, "Setting caps %" GST_PTR_FORMAT, caps);
if ((current_caps = gst_pad_get_current_caps (GST_BASE_SRC_PAD (bsrc)))) {
GstCaps *curcaps_cp;
GstStructure *cur_st, *caps_st;
GST_DEBUG_OBJECT (self, "Pad already has caps %" GST_PTR_FORMAT, caps);
curcaps_cp = gst_caps_make_writable (current_caps);
cur_st = gst_caps_get_structure (curcaps_cp, 0);
caps_st = gst_caps_get_structure (caps, 0);
gst_structure_remove_field (cur_st, "channel-mask");
if (!gst_structure_can_intersect (caps_st, cur_st)) {
GST_ERROR_OBJECT (self, "New caps are not compatible with old caps");
gst_caps_unref (current_caps);
gst_caps_unref (curcaps_cp);
return FALSE;
} else {
gst_caps_unref (current_caps);
gst_caps_unref (curcaps_cp);
return TRUE;
}
}
if (!gst_audio_info_from_caps (&self->info, caps))
return FALSE;
if (self->info.finfo->format == GST_AUDIO_FORMAT_S16LE) {
sample_depth = bmdAudioSampleType16bitInteger;
} else {
sample_depth = bmdAudioSampleType32bitInteger;
}
switch (self->connection) {
case GST_DECKLINK_AUDIO_CONNECTION_AUTO:{
GstElement *videosrc = NULL;
GstDecklinkConnectionEnum vconn;
// Try to get the connection from the videosrc and try
// to select a sensible audio connection based on that
g_mutex_lock (&self->input->lock);
if (self->input->videosrc)
videosrc = GST_ELEMENT_CAST (gst_object_ref (self->input->videosrc));
g_mutex_unlock (&self->input->lock);
if (videosrc) {
g_object_get (videosrc, "connection", &vconn, NULL);
gst_object_unref (videosrc);
switch (vconn) {
case GST_DECKLINK_CONNECTION_SDI:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_CONNECTION_HDMI:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_CONNECTION_OPTICAL_SDI:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_CONNECTION_COMPONENT:
conn = bmdAudioConnectionAnalog;
break;
case GST_DECKLINK_CONNECTION_COMPOSITE:
conn = bmdAudioConnectionAnalog;
break;
case GST_DECKLINK_CONNECTION_SVIDEO:
conn = bmdAudioConnectionAnalog;
break;
default:
// Use default
break;
}
}
break;
}
case GST_DECKLINK_AUDIO_CONNECTION_EMBEDDED:
conn = bmdAudioConnectionEmbedded;
break;
case GST_DECKLINK_AUDIO_CONNECTION_AES_EBU:
conn = bmdAudioConnectionAESEBU;
break;
case GST_DECKLINK_AUDIO_CONNECTION_ANALOG:
conn = bmdAudioConnectionAnalog;
break;
case GST_DECKLINK_AUDIO_CONNECTION_ANALOG_XLR:
conn = bmdAudioConnectionAnalogXLR;
break;
case GST_DECKLINK_AUDIO_CONNECTION_ANALOG_RCA:
conn = bmdAudioConnectionAnalogRCA;
break;
default:
g_assert_not_reached ();
break;
}
if (conn != (BMDAudioConnection) - 1) {
ret =
self->input->config->SetInt (bmdDeckLinkConfigAudioInputConnection,
conn);
if (ret != S_OK) {
GST_ERROR ("set configuration (audio input connection): 0x%08x", ret);
return FALSE;
}
}
ret = self->input->input->EnableAudioInput (bmdAudioSampleRate48kHz,
sample_depth, 2);
if (ret != S_OK) {
GST_WARNING_OBJECT (self, "Failed to enable audio input: 0x%08x", ret);
return FALSE;
}
g_mutex_lock (&self->input->lock);
self->input->audio_enabled = TRUE;
if (self->input->start_streams && self->input->videosrc)
self->input->start_streams (self->input->videosrc);
g_mutex_unlock (&self->input->lock);
return TRUE;
}
static GstFlowReturn
gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
GstBuffer ** outbuf)
{
GstBaseAudioSrc *src = GST_BASE_AUDIO_SRC (bsrc);
GstBuffer *buf;
guchar *data;
guint samples, total_samples;
guint64 sample;
gint bps;
GstRingBuffer *ringbuffer;
GstRingBufferSpec *spec;
guint read;
GstClockTime timestamp, duration;
GstClock *clock;
ringbuffer = src->ringbuffer;
spec = &ringbuffer->spec;
if (G_UNLIKELY (!gst_ring_buffer_is_acquired (ringbuffer)))
goto wrong_state;
bps = spec->bytes_per_sample;
if ((length == 0 && bsrc->blocksize == 0) || length == -1)
/* no length given, use the default segment size */
length = spec->segsize;
else
/* make sure we round down to an integral number of samples */
length -= length % bps;
/* figure out the offset in the ringbuffer */
if (G_UNLIKELY (offset != -1)) {
sample = offset / bps;
/* if a specific offset was given it must be the next sequential
* offset we expect or we fail for now. */
if (src->next_sample != -1 && sample != src->next_sample)
goto wrong_offset;
} else {
/* calculate the sequentially next sample we need to read. This can jump and
* create a DISCONT. */
sample = gst_base_audio_src_get_offset (src);
}
GST_DEBUG_OBJECT (src, "reading from sample %" G_GUINT64_FORMAT, sample);
/* get the number of samples to read */
total_samples = samples = length / bps;
/* FIXME, using a bufferpool would be nice here */
buf = gst_buffer_new_and_alloc (length);
data = GST_BUFFER_DATA (buf);
do {
read = gst_ring_buffer_read (ringbuffer, sample, data, samples);
GST_DEBUG_OBJECT (src, "read %u of %u", read, samples);
/* if we read all, we're done */
if (read == samples)
break;
/* else something interrupted us and we wait for playing again. */
GST_DEBUG_OBJECT (src, "wait playing");
if (gst_base_src_wait_playing (bsrc) != GST_FLOW_OK)
goto stopped;
GST_DEBUG_OBJECT (src, "continue playing");
/* read next samples */
sample += read;
samples -= read;
data += read * bps;
} while (TRUE);
/* mark discontinuity if needed */
if (G_UNLIKELY (sample != src->next_sample) && src->next_sample != -1) {
GST_WARNING_OBJECT (src,
"create DISCONT of %" G_GUINT64_FORMAT " samples at sample %"
G_GUINT64_FORMAT, sample - src->next_sample, sample);
GST_ELEMENT_WARNING (src, CORE, CLOCK,
(_("Can't record audio fast enough")),
("Dropped %" G_GUINT64_FORMAT " samples. This is most likely because "
"downstream can't keep up and is consuming samples too slowly.",
sample - src->next_sample));
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
}
src->next_sample = sample + samples;
/* get the normal timestamp to get the duration. */
timestamp = gst_util_uint64_scale_int (sample, GST_SECOND, spec->rate);
duration = gst_util_uint64_scale_int (src->next_sample, GST_SECOND,
spec->rate) - timestamp;
GST_OBJECT_LOCK (src);
if (!(clock = GST_ELEMENT_CLOCK (src)))
goto no_sync;
if (clock != src->clock) {
/* we are slaved, check how to handle this */
switch (src->priv->slave_method) {
case GST_BASE_AUDIO_SRC_SLAVE_RESAMPLE:
/* not implemented, use skew algorithm. This algorithm should
* work on the readout pointer and produces more or less samples based
* on the clock drift */
case GST_BASE_AUDIO_SRC_SLAVE_SKEW:
{
GstClockTime running_time;
GstClockTime base_time;
GstClockTime current_time;
guint64 running_time_sample;
gint running_time_segment;
gint last_read_segment;
gint segment_skew;
gint sps;
gint segments_written;
gint last_written_segment;
/* get the amount of segments written from the device by now */
segments_written = g_atomic_int_get (&ringbuffer->segdone);
/* subtract the base to segments_written to get the number of the
last written segment in the ringbuffer (one segment written = segment 0) */
last_written_segment = segments_written - ringbuffer->segbase - 1;
/* samples per segment */
sps = ringbuffer->samples_per_seg;
/* get the current time */
current_time = gst_clock_get_time (clock);
/* get the basetime */
base_time = GST_ELEMENT_CAST (src)->base_time;
/* get the running_time */
running_time = current_time - base_time;
/* the running_time converted to a sample (relative to the ringbuffer) */
running_time_sample =
gst_util_uint64_scale_int (running_time, spec->rate, GST_SECOND);
/* the segmentnr corrensponding to running_time, round down */
running_time_segment = running_time_sample / sps;
/* the segment currently read from the ringbuffer */
last_read_segment = sample / sps;
/* the skew we have between running_time and the ringbuffertime (last written to) */
segment_skew = running_time_segment - last_written_segment;
GST_DEBUG_OBJECT (bsrc,
"\n running_time = %"
GST_TIME_FORMAT
"\n timestamp = %"
GST_TIME_FORMAT
"\n running_time_segment = %d"
"\n last_written_segment = %d"
"\n segment_skew (running time segment - last_written_segment) = %d"
"\n last_read_segment = %d",
GST_TIME_ARGS (running_time), GST_TIME_ARGS (timestamp),
running_time_segment, last_written_segment, segment_skew,
last_read_segment);
/* Resync the ringbuffer if:
*
* 1. We are more than the length of the ringbuffer behind.
* The length of the ringbuffer then gets to dictate
* the threshold for what is concidered "too late"
*
* 2. If this is our first buffer.
* We know that we should catch up to running_time
* the first time we are ran.
*/
if ((segment_skew >= ringbuffer->spec.segtotal) ||
(last_read_segment == 0)) {
gint new_read_segment;
gint segment_diff;
guint64 new_sample;
/* the difference between running_time and the last written segment */
segment_diff = running_time_segment - last_written_segment;
/* advance the ringbuffer */
gst_ring_buffer_advance (ringbuffer, segment_diff);
/* we move the new read segment to the last known written segment */
new_read_segment =
g_atomic_int_get (&ringbuffer->segdone) - ringbuffer->segbase;
/* we calculate the new sample value */
new_sample = ((guint64) new_read_segment) * sps;
/* and get the relative time to this -> our new timestamp */
timestamp =
gst_util_uint64_scale_int (new_sample, GST_SECOND, spec->rate);
/* we update the next sample accordingly */
src->next_sample = new_sample + samples;
GST_DEBUG_OBJECT (bsrc,
"Timeshifted the ringbuffer with %d segments: "
"Updating the timestamp to %" GST_TIME_FORMAT ", "
"and src->next_sample to %" G_GUINT64_FORMAT, segment_diff,
GST_TIME_ARGS (timestamp), src->next_sample);
}
break;
}
case GST_BASE_AUDIO_SRC_SLAVE_RETIMESTAMP:
{
GstClockTime base_time, latency;
/* We are slaved to another clock, take running time of the pipeline clock and
* timestamp against it. Somebody else in the pipeline should figure out the
* clock drift. We keep the duration we calculated above. */
timestamp = gst_clock_get_time (clock);
base_time = GST_ELEMENT_CAST (src)->base_time;
if (GST_CLOCK_DIFF (timestamp, base_time) < 0)
timestamp -= base_time;
else
timestamp = 0;
/* subtract latency */
latency =
gst_util_uint64_scale_int (total_samples, GST_SECOND, spec->rate);
if (timestamp > latency)
timestamp -= latency;
else
timestamp = 0;
}
case GST_BASE_AUDIO_SRC_SLAVE_NONE:
break;
}
} else {
GstClockTime base_time;
/* to get the timestamp against the clock we also need to add our offset */
timestamp = gst_audio_clock_adjust (clock, timestamp);
/* we are not slaved, subtract base_time */
base_time = GST_ELEMENT_CAST (src)->base_time;
if (GST_CLOCK_DIFF (timestamp, base_time) < 0) {
timestamp -= base_time;
GST_LOG_OBJECT (src,
"buffer timestamp %" GST_TIME_FORMAT " (base_time %" GST_TIME_FORMAT
")", GST_TIME_ARGS (timestamp), GST_TIME_ARGS (base_time));
} else {
GST_LOG_OBJECT (src,
"buffer timestamp 0, ts %" GST_TIME_FORMAT " <= base_time %"
GST_TIME_FORMAT, GST_TIME_ARGS (timestamp),
GST_TIME_ARGS (base_time));
timestamp = 0;
}
}
no_sync:
GST_OBJECT_UNLOCK (src);
GST_BUFFER_TIMESTAMP (buf) = timestamp;
GST_BUFFER_DURATION (buf) = duration;
GST_BUFFER_OFFSET (buf) = sample;
GST_BUFFER_OFFSET_END (buf) = sample + samples;
*outbuf = buf;
return GST_FLOW_OK;
/* ERRORS */
wrong_state:
{
GST_DEBUG_OBJECT (src, "ringbuffer in wrong state");
return GST_FLOW_WRONG_STATE;
}
wrong_offset:
{
GST_ELEMENT_ERROR (src, RESOURCE, SEEK,
(NULL), ("resource can only be operated on sequentially but offset %"
G_GUINT64_FORMAT " was given", offset));
return GST_FLOW_ERROR;
}
stopped:
{
gst_buffer_unref (buf);
GST_DEBUG_OBJECT (src, "ringbuffer stopped");
return GST_FLOW_WRONG_STATE;
}
}
/* this internal thread does nothing else but write samples to the audio device.
* It will write each segment in the ringbuffer and will update the play
* pointer.
* The start/stop methods control the thread.
*/
static void
audioringbuffer_thread_func (GstAudioRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioSinkRingBuffer *abuf = GST_AUDIO_SINK_RING_BUFFER_CAST (buf);
WriteFunc writefunc;
GstMessage *message;
GValue val = { 0 };
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
GST_DEBUG_OBJECT (sink, "enter thread");
GST_OBJECT_LOCK (abuf);
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (abuf);
writefunc = csink->write;
if (writefunc == NULL)
goto no_function;
g_value_init (&val, G_TYPE_POINTER);
g_value_set_pointer (&val, sink->thread);
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (sink));
gst_message_set_stream_status_object (message, &val);
GST_DEBUG_OBJECT (sink, "posting ENTER stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
while (TRUE) {
gint left, len;
guint8 *readptr;
gint readseg;
/* buffer must be started */
if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
gint written;
left = len;
do {
written = writefunc (sink, readptr, left);
GST_LOG_OBJECT (sink, "transfered %d bytes of %d from segment %d",
written, left, readseg);
if (written < 0 || written > left) {
/* might not be critical, it e.g. happens when aborting playback */
GST_WARNING_OBJECT (sink,
"error writing data in %s (reason: %s), skipping segment (left: %d, written: %d)",
GST_DEBUG_FUNCPTR_NAME (writefunc),
(errno > 1 ? g_strerror (errno) : "unknown"), left, written);
break;
}
left -= written;
readptr += written;
} while (left > 0);
/* clear written samples */
gst_audio_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_audio_ring_buffer_advance (buf, 1);
} else {
GST_OBJECT_LOCK (abuf);
if (!abuf->running)
goto stop_running;
if (G_UNLIKELY (g_atomic_int_get (&buf->state) ==
GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
GST_OBJECT_UNLOCK (abuf);
continue;
}
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
GST_DEBUG_OBJECT (sink, "wait for action");
GST_AUDIO_SINK_RING_BUFFER_WAIT (buf);
GST_DEBUG_OBJECT (sink, "got signal");
if (!abuf->running)
goto stop_running;
GST_DEBUG_OBJECT (sink, "continue running");
GST_OBJECT_UNLOCK (abuf);
}
}
/* Will never be reached */
g_assert_not_reached ();
return;
/* ERROR */
no_function:
{
GST_DEBUG_OBJECT (sink, "no write function, exit thread");
return;
}
stop_running:
{
GST_OBJECT_UNLOCK (abuf);
GST_DEBUG_OBJECT (sink, "stop running, exit thread");
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT_CAST (sink));
gst_message_set_stream_status_object (message, &val);
GST_DEBUG_OBJECT (sink, "posting LEAVE stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
return;
}
}
static gboolean
gst_mms_start (GstBaseSrc * bsrc)
{
GstMMS *mms = GST_MMS (bsrc);
guint bandwidth_avail;
if (!mms->uri_name || *mms->uri_name == '\0')
goto no_uri;
if (mms->connection_speed)
bandwidth_avail = mms->connection_speed;
else
bandwidth_avail = G_MAXINT;
/* If we already have a connection, and the uri isn't changed, reuse it,
as connecting is expensive. */
if (mms->connection) {
if (!strcmp (mms->uri_name, mms->current_connection_uri_name)) {
GST_DEBUG_OBJECT (mms, "Reusing existing connection for %s",
mms->uri_name);
return TRUE;
} else {
mmsx_close (mms->connection);
g_free (mms->current_connection_uri_name);
mms->current_connection_uri_name = NULL;
}
}
/* FIXME: pass some sane arguments here */
GST_DEBUG_OBJECT (mms,
"Trying mms_connect (%s) with bandwidth constraint of %d bps",
mms->uri_name, bandwidth_avail);
mms->connection = mmsx_connect (NULL, NULL, mms->uri_name, bandwidth_avail);
if (mms->connection) {
/* Save the uri name so that it can be checked for connection reusing,
see above. */
mms->current_connection_uri_name = g_strdup (mms->uri_name);
GST_DEBUG_OBJECT (mms, "Connect successful");
return TRUE;
} else {
gchar *url, *location;
GST_ERROR_OBJECT (mms,
"Could not connect to this stream, redirecting to rtsp");
location = strstr (mms->uri_name, "://");
if (location == NULL || *location == '\0' || *(location + 3) == '\0')
goto no_uri;
url = g_strdup_printf ("rtsp://%s", location + 3);
gst_element_post_message (GST_ELEMENT_CAST (mms),
gst_message_new_element (GST_OBJECT_CAST (mms),
gst_structure_new ("redirect", "new-location", G_TYPE_STRING, url,
NULL)));
/* post an error message as well, so that applications that don't handle
* redirect messages get to see a proper error message */
GST_ELEMENT_ERROR (mms, RESOURCE, OPEN_READ,
("Could not connect to streaming server."),
("A redirect message was posted on the bus and should have been "
"handled by the application."));
return FALSE;
}
no_uri:
{
GST_ELEMENT_ERROR (mms, RESOURCE, OPEN_READ,
("No URI to open specified"), (NULL));
return FALSE;
}
}
static GstFlowReturn
gst_dvbsrc_read_device (GstDvbSrc * object, int size, GstBuffer ** buffer)
{
gint count = 0;
gint ret_val = 0;
GstBuffer *buf = gst_buffer_new_and_alloc (size);
GstClockTime timeout = object->timeout * GST_USECOND;
GstMapInfo map;
g_return_val_if_fail (GST_IS_BUFFER (buf), GST_FLOW_ERROR);
if (object->fd_dvr < 0)
return GST_FLOW_ERROR;
gst_buffer_map (buf, &map, GST_MAP_WRITE);
while (count < size) {
ret_val = gst_poll_wait (object->poll, timeout);
GST_LOG_OBJECT (object, "select returned %d", ret_val);
if (G_UNLIKELY (ret_val < 0)) {
if (errno == EBUSY)
goto stopped;
else
goto select_error;
} else if (G_UNLIKELY (ret_val == 0)) {
/* timeout, post element message */
gst_element_post_message (GST_ELEMENT_CAST (object),
gst_message_new_element (GST_OBJECT (object),
gst_structure_new_empty ("dvb-read-failure")));
} else {
int nread = read (object->fd_dvr, map.data + count, size - count);
if (G_UNLIKELY (nread < 0)) {
GST_WARNING_OBJECT
(object,
"Unable to read from device: /dev/dvb/adapter%d/dvr%d (%d)",
object->adapter_number, object->frontend_number, errno);
gst_element_post_message (GST_ELEMENT_CAST (object),
gst_message_new_element (GST_OBJECT (object),
gst_structure_new_empty ("dvb-read-failure")));
} else
count = count + nread;
}
}
gst_buffer_unmap (buf, &map);
gst_buffer_resize (buf, 0, count);
*buffer = buf;
return GST_FLOW_OK;
stopped:
{
GST_DEBUG_OBJECT (object, "stop called");
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
return GST_FLOW_FLUSHING;
}
select_error:
{
GST_ELEMENT_ERROR (object, RESOURCE, READ, (NULL),
("select error %d: %s (%d)", ret_val, g_strerror (errno), errno));
gst_buffer_unmap (buf, &map);
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
}
}
static gboolean
gst_dvbsrc_open_frontend (GstDvbSrc * object, gboolean writable)
{
struct dvb_frontend_info fe_info;
const char *adapter_desc = NULL;
gchar *frontend_dev;
GstStructure *adapter_structure;
char *adapter_name = NULL;
frontend_dev = g_strdup_printf ("/dev/dvb/adapter%d/frontend%d",
object->adapter_number, object->frontend_number);
GST_INFO_OBJECT (object, "Using frontend device: %s", frontend_dev);
/* open frontend */
if ((object->fd_frontend =
open (frontend_dev, writable ? O_RDWR : O_RDONLY)) < 0) {
switch (errno) {
case ENOENT:
GST_ELEMENT_ERROR (object, RESOURCE, NOT_FOUND,
(_("Device \"%s\" does not exist."), frontend_dev), (NULL));
break;
default:
GST_ELEMENT_ERROR (object, RESOURCE, OPEN_READ_WRITE,
(_("Could not open frontend device \"%s\"."), frontend_dev),
GST_ERROR_SYSTEM);
break;
}
close (object->fd_frontend);
g_free (frontend_dev);
return FALSE;
}
GST_DEBUG_OBJECT (object, "Device opened, querying information");
if (ioctl (object->fd_frontend, FE_GET_INFO, &fe_info) < 0) {
GST_ELEMENT_ERROR (object, RESOURCE, SETTINGS,
(_("Could not get settings from frontend device \"%s\"."),
frontend_dev), GST_ERROR_SYSTEM);
close (object->fd_frontend);
g_free (frontend_dev);
return FALSE;
}
GST_DEBUG_OBJECT (object, "Got information about adapter : %s", fe_info.name);
adapter_name = g_strdup (fe_info.name);
object->adapter_type = fe_info.type;
switch (object->adapter_type) {
case FE_QPSK:
adapter_desc = "DVB-S";
adapter_structure = gst_structure_new ("dvb-adapter",
"type", G_TYPE_STRING, adapter_desc,
"name", G_TYPE_STRING, adapter_name,
"auto-fec", G_TYPE_BOOLEAN, fe_info.caps & FE_CAN_FEC_AUTO, NULL);
break;
case FE_QAM:
adapter_desc = "DVB-C";
adapter_structure = gst_structure_new ("dvb-adapter",
"type", G_TYPE_STRING, adapter_desc,
"name", G_TYPE_STRING, adapter_name,
"auto-inversion", G_TYPE_BOOLEAN,
fe_info.caps & FE_CAN_INVERSION_AUTO, "auto-qam", G_TYPE_BOOLEAN,
fe_info.caps & FE_CAN_QAM_AUTO, "auto-fec", G_TYPE_BOOLEAN,
fe_info.caps & FE_CAN_FEC_AUTO, NULL);
break;
case FE_OFDM:
adapter_desc = "DVB-T";
adapter_structure = gst_structure_new ("dvb-adapter",
"type", G_TYPE_STRING, adapter_desc,
"name", G_TYPE_STRING, adapter_name,
"auto-inversion", G_TYPE_BOOLEAN,
fe_info.caps & FE_CAN_INVERSION_AUTO, "auto-qam", G_TYPE_BOOLEAN,
fe_info.caps & FE_CAN_QAM_AUTO, "auto-transmission-mode",
G_TYPE_BOOLEAN, fe_info.caps & FE_CAN_TRANSMISSION_MODE_AUTO,
"auto-guard-interval", G_TYPE_BOOLEAN,
fe_info.caps & FE_CAN_GUARD_INTERVAL_AUTO, "auto-hierarchy",
G_TYPE_BOOLEAN, fe_info.caps % FE_CAN_HIERARCHY_AUTO, "auto-fec",
G_TYPE_BOOLEAN, fe_info.caps & FE_CAN_FEC_AUTO, NULL);
break;
case FE_ATSC:
adapter_desc = "ATSC";
adapter_structure = gst_structure_new ("dvb-adapter",
"type", G_TYPE_STRING, adapter_desc,
"name", G_TYPE_STRING, adapter_name, NULL);
break;
default:
g_error ("Unknown frontend type: %d", object->adapter_type);
adapter_structure = gst_structure_new ("dvb-adapter",
"type", G_TYPE_STRING, "unknown", NULL);
}
GST_INFO_OBJECT (object, "DVB card: %s ", adapter_name);
gst_element_post_message (GST_ELEMENT_CAST (object), gst_message_new_element
(GST_OBJECT (object), adapter_structure));
g_free (frontend_dev);
g_free (adapter_name);
return TRUE;
}