Post global tags only after we've added our source pads, so that
tag events get sent downstream in addition to tag messages posted
on the bus. This makes sure tags can be picked up automatically
when transcoding, but also by tagreadbin/playbin2. Fixes#519721.
While we're at it, also add a container-format tag.
When we receive a DISCONT as input, don't clear our complete state but simply
mark a discont that will be put on the next buffer. The code will be able to
handle and throw away incomplete data.
Add some more debug info.
Remove an unused variable.
Let's not put every single mp3 frame in our index, a few frames per
second should be more than enough. For now use an index interval
of 100ms-500ms depending on the upstream size, to keep the index at
a reasonable size. Factor out the code that adds the index entry
into a separate function for better code readability.
While technically upstream may be seekable even if it doesn't know
the exact size, I can't think of a use case where this distincation
is relevant in practice, so for now just assume we're not seekable
if upstream doesn't provide us with a size. Makes sure we don't
build a seek index when streaming internet radio with sources that
pretend to be seekable until you try to actually seek.
Don't overwrite the origin flow return by whatever flow we get
when trying to push the remaining internally queued payloads.
We want to do our eos logic, ie. send an EOS event or segment-done
message in any case. Makes things EOS properly when an EOS event
is forced upon the pipeline so that the source returns
FLOW_UNEXPECTED to a pulling asfdemux. Should fix#582056.
Add a multichannel map to the output caps, and send at least a CODEC and
BITRATE tag. I'm not too sure about the 5.1 and 7.1 channel maps. I have
no samples and can't find info about the channel ordering, but this is
better than nothing.
Some Xing headers apparently start the TOC at byte 1 instead of 0. Don't
reject them because of it, just subtract the initial offset when reading
the table.
Be more lenient about what we accept as changing bits in a header - basically,
only require that the mp3 sync marker is present, for the mpeg version,
layer and samplerate.
Fixes: #581464
Some mp3 streams have an offset in timestamps, requiring us to push the
frame *AFTER* segment.stop in order for the decoder to be able to push
all data up to the segment.stop position.
This also makes timestamps (more) consistent before and after a possible
seek, and moreover makes for reasonable position reporting in live stream
(whose payload timestamps should not be taken for granted).
* Improve newsegment handling, e.g. upstream might live in TIME.
* Only send newsegment if we have needed info.
* Avoid reading past end of data section.
The problem that happens is the following:
* A packet with multiple payloads comes in
* Those payloads get handled one by one
* The first payload contains the first audio payload with timestamp A
* The second payload contains the first video (key)frame with timestamp V (where V < A)
With the previous code, the following would happen:
* the first payload gets processed, then passed to queue_for_stream
* queue_for_stream detects it's the first valid timestamp received and stores
first_ts = A
* the second payload gets processed, then pass to queue_for_stream
* queue_for_stream detects the timestamp is lower than first_ts... and
discards it... resulting in losing the first keyframe of the video stream
We've been having this issue for *ages*... it's just that nobody noticed it
that much with playbin. But with playbin2's aggresive multiqueue handling, this
will result in multiqueue not being able to preroll (because the video decoder will
be dropping a ton of buffers before (maybe) receiving the next keyframe).
Tested with over 200 asf files, and they all play the first frame correctly now,
even the most braindead ones.
This might be caused by entering the if() line 1214 and then not having
any activated_streams.. resulting in reaching line 1267 without having
any valid flow value.
Fixes playback of Windows Media RTSP streams and other non-Real RTSP
streams where the server errors out because it can't handle the
Real-specific 'Required: com.real.retain-entity-for-setup' header
we've been adding unconditionally in the recent past.
For reference:
rtsp://66.111.34.191:601/broadcast/alnour.rm
rtsp://195.134.224.231/snowboard_100.wmv
On win32, we're required to link to all the libraries used - including
ones only indirectly used by other libs. So, add gstaudio, gsttag, and
(for windows only) winsock.
Parse the ETag from the describe method and pass the sessionid as the value for
the If-Match header is subsequent setup calls.
Fixes support for more RealMedia RTSP streams.
Don't introduce glitches in the output by a) relaxing the threshold for
taking upstream timestamps in preference to our calculated timestamps and
b) only set the discont flag on outgoing buffers in response to an incoming
discont buffer.
Fixes: #575046
Don't allow a change in sample rate/channels/layer/version unless we can
see another frame at the correct offset. Prevents accidently flipping
due to simple single-bit corruption.
Since SEEK event handling might perform some conversion
from TIME to BYTES, do not let upstream fool application
into (TIME) seeking not being possible.
Integer underflow made accurate seeks to near zero fail and seek to
completely the wrong place. Fix by clamping to zero, since we can't seek
to negative times anyway.
Add a new utri handler for pnm:// that for now just redirects to the same uri
with the rtsp:// protocol, which usually works nowadays.
Separate the registration of the various plugins into a separate source file.
Drop packets with an invalid replicated data length
instead of continuing with an invalid timestamp
and uninitialized payload metadata.
All other code assumes that the timestamps are valid.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c: (channel_mode_class),
(GST_TYPE_MP3_CHANNEL_MODE), (mp3_type_frame_length_from_header),
(gst_mp3parse_emit_frame), (mp3parse_get_query_types):
* gst/mpegaudioparse/gstmpegaudioparse.h:
Do an initial class_ref on an internal enum type from within the
class_init function so that there aren't any issues when multiple
mp3parse elements are started in separate threads at the same
time. (Why we use an enum type here if the tag is registered as
a string type, I don't know). Also remove custom UNUSED macro
and use GLib's instead.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_chain):
Remove duplicate and broken code for the streaming case and simply reuse
the much better working pull based code. Fixes#560348.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_add_video_stream):
Only copy sane aspect ratio values on the caps. Fixes#559682.
Original commit message from CVS:
Patch by: Tal Shalif <tshalif at nargila dot org>
* gst/mpegstream/gstdvddemux.c:
(gst_dvd_demux_get_subpicture_stream):
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_get_video_stream),
(gst_mpeg_demux_get_audio_stream):
Fix memmory corruption due to not storing the new updated pointer
after a g_renew(). Fixes#558896.
Original commit message from CVS:
* gst/realmedia/rmdemux.c: (gst_rmdemux_add_stream),
(gst_rmdemux_descramble_mp4a_audio),
(gst_rmdemux_handle_scrambled_packet):
Add suport for mpeg4 and aac audio. See #556714.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
Calculate samples per frame correctly for "MPEG 2.5" layer 3.
Fixes skipping on these files.
Original commit message from CVS:
* gst/mpegaudioparse/gstmpegaudioparse.c:
(gst_mp3parse_sink_event):
Post a GST_ELEMENT_ERROR if we get EOS before seeing any valid
frames. Partially fixes bug #552237.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_loop):
Fix aggregated GST_FLOW_RETURN check for when to send an error message
on the bus.
Re-fixes #546859
Original commit message from CVS:
* gst/realmedia/rdtdepay.c: (gst_rdt_depay_init),
(gst_rdt_depay_setcaps), (gst_rdt_depay_sink_event),
(create_segment_event), (gst_rdt_depay_push),
(gst_rdt_depay_handle_data), (gst_rdt_depay_change_state):
* gst/realmedia/rdtdepay.h:
Parse other values from the incomming caps.
Add event handler to handle flushing and segments.
Create segment events.
* gst/realmedia/rdtjitterbuffer.c: (rdt_jitter_buffer_insert):
Do skew correction based on RDT timestamps.
* gst/realmedia/rdtmanager.c: (activate_session),
(gst_rdt_manager_parse_caps), (gst_rdt_manager_setcaps),
(create_recv_rtp):
Parse caps to get the clockrate needed for the jitterbuffer.
* gst/realmedia/rmdemux.c: (gst_rmdemux_parse_video_packet):
Apply timestamp fixup after correcting for initial timestamp and
internal base timestamp corrections.
Original commit message from CVS:
* gst/realmedia/rdtdepay.c: (gst_rdt_depay_handle_data),
(gst_rdt_depay_change_state):
* gst/realmedia/rdtdepay.h:
Check seqnum gaps and drop duplicate packets or mark outgoing buffers
with a DISCONT flag when needed.
* gst/realmedia/rdtmanager.c: (gst_rdt_manager_query_src):
Report the configure latency instead of a hardcoded value.
Original commit message from CVS:
* gst/realmedia/rdtmanager.c: (create_session), (activate_session),
(free_session), (gst_rdt_manager_query_src),
(gst_rdt_manager_src_activate_push),
(gst_rdt_manager_handle_data_packet), (gst_rdt_manager_chain_rdt),
(gst_rdt_manager_loop), (create_recv_rtp):
Include the new rdt jitterbuffer in the session manager.
Original commit message from CVS:
* gst/realmedia/rdtdepay.c: (gst_rdt_depay_class_init),
(gst_rdt_depay_finalize), (gst_rdt_depay_setcaps),
(gst_rdt_depay_push), (gst_rdt_depay_handle_data),
(gst_rdt_depay_chain), (gst_rdt_depay_change_state):
* gst/realmedia/rdtdepay.h:
Use new RDT parsing helper functions.
Copy discont flags correctly.
Push the header from the chain function instead of the setcaps function.
Copy incomming timestamp to the output buffers instead of doing magic
with the RDT timestamps.
Original commit message from CVS:
* gst/realmedia/rmdemux.c: (gst_rmdemux_init),
(find_seek_offset_time), (gst_rmdemux_reset), (gst_rmdemux_chain),
(gst_rmdemux_parse_mdpr), (gst_rmdemux_descramble_cook_audio),
(gst_rmdemux_descramble_dnet_audio),
(gst_rmdemux_parse_video_packet), (gst_rmdemux_parse_audio_packet):
* gst/realmedia/rmdemux.h:
Keep track of the first timestamp of the stream and add this to the
outgoing buffer timestamps so that we can handle live streams.
Set discont flag on the first buffers and after a seek.
Original commit message from CVS:
* gst/asfdemux/gstasfdemux.c:
Properly aggregate flow returns for both push and pull mode, so we shut
down if all pads are unlinked.
Fixes#546859.
