Commit graph

9297 commits

Author SHA1 Message Date
Nicolas Dufresne
307018da89 flvdemux: Don't pull passed the EOS
When a truncated FLV is provided and processed in pull mode, we
may endup trying to pull passed EOS, causing a rather confusing
warning as the pull offset is an integer overflow.

https://bugzilla.gnome.org/show_bug.cgi?id=787795
2017-10-05 21:20:19 -04:00
Nicolas Dufresne
cfc1638ca5 flvdemux: Ignore invalid H.264 codec data
This code basically skip over codec_data with empty payload. In
this case, the codec_data variable is the size of the header for
the CODEC part of Video Tag. The remaining is supposed to be the
H.264 codec data, hence should not be empty.

https://bugzilla.gnome.org/show_bug.cgi?id=787795
2017-10-05 21:20:19 -04:00
Nicolas Dufresne
18dbd49fb8 flvdemux: Avoid integer overflow on invalid CTS
If the CTS is negative an would lead to a negtive PTS, clip
the CTS so the PTS will be 0.

https://bugzilla.gnome.org/show_bug.cgi?id=787795
2017-10-05 21:20:19 -04:00
Thibault Saunier
125e835045 rtspsrc: Fix build 2017-10-05 14:37:54 -03:00
Thibault Saunier
ffcd173c09 rtspsrc: Handle TCP as lower transport with RTSP 2.0
Meaning that the interleave fields have to be updated as
if streams setup was working when using pipelined setup
request. Otherwise there is a mismatch between the server
channel count and our own.

This also makes RTSP 2.0 over HTTP working.

https://bugzilla.gnome.org/show_bug.cgi?id=781446
2017-10-05 12:05:40 -03:00
Thibault Saunier
8121752887 rtsp: Start implementing support for RTSP 2.0
- Handle version negotation:

  Added a `default-version` property so that the user can configure
  what to use in case the server does not support version negotation
  (which actually exist)

- Handle pipelined requests, which allow avoiding full round trip to
  setup the RTP streams (request are sent in a raw, and response are
  handled as they arrive).

- Handle the new Media-Properties header

- Handle the new Seek-Style header

- Handle the new Accept-Ranges header

Handling of IPV6 should already be OK.

We are still missing (at least) the following features (which do not
seem really mandatory as they require a "persistent connection between
server and client"):

  - Server to Client TEARDOWN command (Not so usefull fmpov)
  - PLAY_NOTIFY (not needed for our server yet)
  - Support for the new REDIRECT features

and probably some more protocol changes might not be handled yet.

https://bugzilla.gnome.org/show_bug.cgi?id=781446
2017-10-05 12:05:40 -03:00
Thibault Saunier
5faad79049 rtspsrc: Use a macro to debug RTSP messages
Simplifying the code a little.

https://bugzilla.gnome.org/show_bug.cgi?id=781446
2017-10-05 12:05:39 -03:00
Reynaldo H. Verdejo Pinochet
0ddb65dec7 Use proper GtkDoc notation for NULL/FALSE/TRUE 2017-10-03 16:30:10 -07:00
Sebastian Dröge
26cf8c1dc9 rtspsrc: Ignore medias marked as sendonly
We're never going to receive anything from them, so don't create pads
for them. These medias are destinations where *we* could send something.
2017-10-01 16:09:13 +02:00
Sebastian Dröge
58f0eabd61 sbcdepay: Add property to ignore input timestamps
This then just counts samples and calculates the output timestamps based
on that and the very first observed timestamp. The timestamps on the
buffers are continued to be used to detect discontinuities that are too
big and reset the counter at that point.

When receiving data via Bluetooth, many devices put completely wrong
values into the RTP timestamp field. For example iOS seems to put a
timestamp in milliseconds in there, instead of something based on the
current sample offset (RTP clock-rate == sample rate).

https://bugzilla.gnome.org/show_bug.cgi?id=787297
2017-09-28 14:15:12 +03:00
Ponnam Srinivas
c0622addf6 rtph265depay: Fix Memory leak in error case
https://bugzilla.gnome.org/show_bug.cgi?id=787937
2017-09-26 11:09:53 +03:00
Deepak Srivastava
49ed67c4f2 deinterlace: Fixed memory leak in error code path
https://bugzilla.gnome.org/show_bug.cgi?id=788041
2017-09-22 17:25:50 +03:00
Tim-Philipp Müller
a802f5df42 rtpjitterbuffer: implement basic chain_list function
Doesn't do anything fancy yet, but still avoids lots of
unnecessary locking/unlocking that would happen if the
default chain_list fallback function in GstPad got invoked.
2017-09-17 16:33:15 +01:00
Tim-Philipp Müller
07dd3dba6f multifilesink: use new gst_buffer_list_calculate_size() 2017-09-17 12:50:30 +01:00
Patrick Radizi
3de0244532 rtpbin: add option for sanity checking timestamp offset
Timestamp offsets needs to be checked to detect unrealistic values
caused for example by NTP clocks not in sync. The new parameter
max-ts-offset lets the user decide an upper offset limit. There
are two different cases for checking the offset based on if
ntp-sync is used or not:
1) ntp-sync enabled
   Only negative offsest are allowed since a positive offset would
   mean that the sender and receiver clocks are not in sync.
   Default vaule of max-ts-offset = 0 (disabled)
2) ntp-sync disabled
   Both positive and negative offsets are allowed.
   Default vaule of max-ts-offset = 3000000000
The reason for different default values is to be backwards
compatible.

https://bugzilla.gnome.org/show_bug.cgi?id=785733
2017-09-15 13:33:14 +03:00
Patrick Radizi
23f7739ba4 rtpbin: add option for increasing ts_offset gradually
Instant large changes to ts_offset may cause timestamps to move
backwards and also cause visible effects in media playback. The new
option max-ts-offset-adjustment lets the application control the rate to
apply changes to ts_offset.

https://bugzilla.gnome.org/show_bug.cgi?id=784002
2017-09-14 13:15:56 +03:00
Mark Nauwelaerts
cf3f195012 matroskademux: search_cluster should find preceding cluster before target
... since failing this constraint takes search_pos by surprise which might
then end up in an infinite loop.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=787538
2017-09-11 20:39:22 +02:00
George Kiagiadakis
286e1e62be rtprtx{send,receive}: improve the debug messages
* use INFO/DEBUG/LOG/TRACE equaly and meaningfully;
  previously rtprtxsend:LOG and rtprtxreceive:LOG would generate
  a totally different amount of log traffic and sometimes it was
  impossible to see the information you wanted without useless
  spam being printed around
* improve the wording, give a reasonable and self-explanatory
  amount of information
* print SSRCs in hex
* avoid G_FOO_FORMAT for readability (we are just printing integers)
2017-09-07 14:43:32 +03:00
Jan Schmidt
d191584531 qtmux: Allow MPEG layer 1/2, AC3 and Opus in qtmux
qtmux is supposed to be the muxer that allows all formats,
with others (mp4mux and friends) being profile-restricted.
2017-09-06 23:55:38 +10:00
Tim-Philipp Müller
6f9cb1716a rtph265depay: fix keyunit detection
https://bugzilla.gnome.org/show_bug.cgi?id=787254
2017-09-05 13:56:18 +01:00
Edward Hervey
5bc5e07531 dcaparse: Really fix "usage before unmap"
Previous patch would try to unref a buffer that was pushed downstream.

Instead only unref when/if needed and keep usage of the cleanup: goto
block
2017-09-04 17:06:39 +02:00
Arun Raghavan
5cf368b36c dcaparse: Don't unmap buffer before accessing data from it
The previous patch added a check for a substream header after
gst_buffer_unmap(), which is incorrect.
2017-09-03 15:23:10 +05:30
Matej Knopp
f7854411fa dcaparse: preserve DTS HD substream 2017-09-03 15:05:08 +05:30
Arun Raghavan
301e8d558e rtpsbcpay: Fix some tabs that crept in somehow 2017-08-29 22:12:35 +05:30
Sebastian Dröge
71104f452e rtpbin: Also log local and SR RTP running times when doing ntp-sync=true 2017-08-29 19:14:25 +03:00
Matthew Waters
f602b8e5b0 rtpbin: also create session when creating the send_rtcp_src_%u pad
If one requests the send_rtcp_src_%u pad before a recv_rtcp_sink_%u pad,
the session/pad would never be created and NULL was returned.
Switching the request order would work.

https://bugzilla.gnome.org/show_bug.cgi?id=786718
2017-08-29 12:47:30 +10:00
Sky Juan
bc290059a4 ac3parse: fix not-linked handling causing glitches when selecting stream
Fix chain function not handling not-linked from baseparse.

When an input data is separated into 2 buffers, the second buffer
would not be pushed into the adapter if baseparse returns not-linked
for first buffer.

This caused glitches when switching streams and selecting
a stream that was previously unselected.

https://bugzilla.gnome.org/show_bug.cgi?id=786268
2017-08-16 14:09:31 +01:00
Jan Alexander Steffens (heftig)
1de2df03be goom2k1: Convert source files to UTF-8
Causes problems with the new gtk-doc 1.26 otherwise,
but is a good idea in any case.

https://bugzilla.gnome.org/show_bug.cgi?id=786364
2017-08-16 13:07:45 +01:00
Eduard Sinelnikov
6d61471da6 wavparse: Add support for growing WAV files
With some fixes by me.
2017-08-14 16:02:35 +03:00
Arun Raghavan
e6b6583a5e rtpsbcpay: Fix compile error 2017-08-14 17:39:15 +05:30
Jochen Henneberg
f641ac60e3 rtpsbcpay: fix if buffer size exceeds MTU
The plugin queued buffer data if not all buffer data fit
into a single RTP packet. Now RTP packets are pushed as long
as enough data is available.
2017-08-14 16:56:17 +05:30
Mathieu Duponchelle
5e48e85fb7 rtpstats: fix unsigned integer comparisons.
Callers of the API (rtpsource, rtpjitterbuffer) pass clock_rate
as a signed integer, and the comparison "<= 0" is used against
it, leading me to think the intention was to have the field
be typed as gint32, not guint32.

This led to situations where we could call scale_int with
a MAX_UINT32 (-1) guint32 as the denom, thus raising an
assertion.

https://bugzilla.gnome.org/show_bug.cgi?id=785991
2017-08-11 13:29:24 +02:00
Michael Olbrich
61429a78a8 qtdemux: allow larger files
For really long files such as contiguous recordings of a whole day, the
50MB limit is not sufficient.

https://bugzilla.gnome.org/show_bug.cgi?id=781458
2017-08-10 16:17:56 +03:00
Sebastian Dröge
7d89e73336 qtdemux: Fix offsets for reading lpcm specific fields
We were reading at the completely wrong positions, 16 bytes later in the
data.

Also add support for high-aligned samples.
2017-08-10 16:13:05 +03:00
Deepak Srivastava
059420b678 wavparse: Fix memory leak in wavparse element
Fixing of leaking the text field of the GstWavParseNote and
GstWavParseLabl structure.

https://bugzilla.gnome.org/show_bug.cgi?id=785429
2017-08-10 13:23:47 +03:00
Sebastian Dröge
4b8d0a294a mulawdec: Unmap input buffer if failing to map the output buffer 2017-07-27 14:23:14 +03:00
Satya Prakash Gupta
e3f6b9f367 alawdec: Fix Memory leak in error case
https://bugzilla.gnome.org/show_bug.cgi?id=785435
2017-07-27 14:23:14 +03:00
Sebastian Dröge
317d3380bb matroskamux: For audio tracks, take the default duration from the first buffer
... if we don't have any better idea from the caps. This allows writing
SimpleBlocks for a majority of audio streams where the duration of
frames is usually fixed. And as a side effect, allows VLC to play
streams with Opus as it only works with SimpleBlocks currently:
  https://trac.videolan.org/vlc/ticket/18545

https://bugzilla.gnome.org/show_bug.cgi?id=784969
2017-07-25 11:28:46 +03:00
Mark Nauwelaerts
7bd439efb5 qtdemux: preferably send open-ended segment rather than repeated segment events 2017-07-19 18:35:58 +02:00
Mark Nauwelaerts
854f85acb4 qtdemux: fix seeking in fragmented file without mfra random access info
... which no longer worked due to unconditionally clearing sample info and
ending up in inconsistent state.  Let's tread a bit more carefully and also
allow for the old seek handling that resorts to scanning if no mfra info
is available.
2017-07-19 18:35:49 +02:00
Nicola Murino
1bbdfa8738 matroskamux: add properties to control cluster duration
https://bugzilla.gnome.org/show_bug.cgi?id=784971
2017-07-18 10:08:33 +03:00
Sebastian Dröge
c0318f3d60 qtdemux: Fix parsing of RLE depth
Regression introduced by 86b427dc70

https://bugzilla.gnome.org/show_bug.cgi?id=784812
2017-07-13 12:47:27 +03:00
Yasushi SHOJI
c7f42cc3bc rtpgsmpay: fix accidental garbage data before actual payload
Do not allocate payload size outbuf if appending payload buffer.

The commit 137672ff18 attached payload
to the output buffer but forgot to remove payload allocation.  That
effectively doubled payload size and add zero'ed or random bytes.

Makes the following pipeline work again:

gst-launch-1.0 -v audiotestsrc wave=2 ! gsmenc ! rtpgsmpay ! rtpgsmdepay ! gsmdec ! autoaudiosink

https://bugzilla.gnome.org/show_bug.cgi?id=784616
2017-07-09 13:21:23 +01:00
Mark Nauwelaerts
286df32e03 matroskademux: segment seek position is expressed in buffer time
... so it need not be corrected again for stream start
2017-07-09 10:55:00 +02:00
Mark Nauwelaerts
dad2b41e4d avidemux: provide average bitrate tag 2017-07-09 10:55:00 +02:00
Olivier Crête
96e71b0286 rtpsession: Send EOS if all internal sources sent bye
The ones which are not internal should not matter, and we should
wait for all sources to have sent their BYEs.

And add unit test

https://bugzilla.gnome.org/show_bug.cgi?id=773218
2017-07-04 21:14:10 -04:00
Olivier Crête
7e7e52caa0 rtpsession: Only send EOS if all sources have been marked bye
Now that multiple sender RTPSource can share the same RTPSession, we
must not send an EOS unless they're all marked bye.
2017-07-04 13:36:44 -04:00
Nicolas Dufresne
bf5cbce3b4 rtprtxreceive: Add memory and boudary checks
This element was not checking if mapping the RTP buffer and the payload
worked, and was not checking if the RTX payload was large enough.

https://bugzilla.gnome.org/show_bug.cgi?id=784484
2017-07-04 09:58:15 -04:00
Tim-Philipp Müller
d712a2f4b7 imagefreeze: fix use-after-free on seek event
Get seqnum before unreffing the seek event.

https://bugzilla.gnome.org/show_bug.cgi?id=784486
2017-07-03 20:43:12 +01:00
Mark Nauwelaerts
f9f1a03653 qtmux: robustify time tracking for sparse subtitle stream 2017-07-01 18:59:14 +02:00