Commit graph

9418 commits

Author SHA1 Message Date
Seungha Yang
93f19b19d2 isomp4: Add support VP9 codec
Add fourcc for VP9 codec and support it by qtdemux and qtmux
See also https://www.webmproject.org/vp9/mp4/

https://bugzilla.gnome.org/show_bug.cgi?id=790026
2017-11-13 10:08:14 -05:00
Edward Hervey
f89d672eb7 matroskademux: Remove bogus error message
It's just informational
2017-11-13 13:51:20 +01:00
Edward Hervey
50c3733a89 rtpmpvpay: Don't create empty buffer list
If there's nothing to send, just return
2017-11-10 15:51:05 +01:00
Seungha Yang
5dd39d8946 matroskademux: Add parsing Colour element
... and forward colorimetry to downstream. The Colour element describes
various color information (similar to 'colr' box in isobmff).
Note that, due to the comparatively limited syntax for color information
in vpx codecs, the color information in mkv/wemb container level
should be used for sophisticated color handling (e.g., HDR video).

https://bugzilla.gnome.org/show_bug.cgi?id=790023
2017-11-08 11:01:55 +02:00
Youness Alaoui
593615de46 rtpg722pay: Add encoding-params to the src caps template
The G722 payload only accepts G722 audio with channels=1, so it must
specify the encoding-params=1 in its src caps, otherwise it causes issues
with farstream which thinks it supports 2 channels G722 and when
confronted with a remote that has G722/8000/2, it will negotiate it
and error out with a not-negotiated when the caps don't intersect
at runtime.

https://bugzilla.gnome.org/show_bug.cgi?id=789878
2017-11-03 17:20:31 -04:00
Sebastian Dröge
af3ce874a0 rtspsrc: Add "accept-certificate" signal for manually checking a TLS certificate for validity
https://bugzilla.gnome.org/show_bug.cgi?id=785024
2017-11-01 13:42:45 +02:00
Sangkyu Park
257f7c9063 rtspsrc: Print RTSP/SDP messages to gstreamer log instead of stdout
- 'debug' property is deprecated
- All RTSP messages are printed to gstreamer log with 'log' level.

https://bugzilla.gnome.org/show_bug.cgi?id=788917
2017-11-01 11:35:44 +02:00
Justin Kim
2a5aafe425 rtpsesson: downgrade message level to debug when detected XR
When XR packet is detected, warning message leads to misunderstandings.
Until RFC3611 is implemented in gst-plugins-base, the level needs to
be downgraded to avoid confusion.

https://bugzilla.gnome.org/show_bug.cgi?id=789746
2017-11-01 10:57:00 +02:00
Ashish Kumar
1aed3948a8 gst-plugins-good: atoms_recovery: Handled buffer mapping failure
https://bugzilla.gnome.org/show_bug.cgi?id=789413
2017-10-24 22:04:27 -07:00
Thiago Santos
923b83a48c atomsrecovery: read from mdat only what is on headers
It is possible that the mdat has more data than what was stored in the
headers file. If we put that to the output the file will have bogus data
at the end and some players will complain.

https://bugzilla.gnome.org/show_bug.cgi?id=784258
2017-10-24 21:33:25 -07:00
Thiago Santos
69605b6c61 isomp4: atomsrecovery: handle common and large atom headers
Do not assume all files are large files. Check and use the short or
extended atom size field only if needed.

https://bugzilla.gnome.org/show_bug.cgi?id=784258
2017-10-24 21:33:25 -07:00
Thibault Saunier
6d72ae13fe rtsp: Add missing Since marker 2017-10-16 14:21:56 +02:00
Sebastian Dröge
728a1629cf qtdemux: Fix compiler warning
qtdemux.c: In function ‘gst_qtdemux_configure_stream’:
qtdemux.c:7764:34: error: suggest parentheses around ‘&&’ within ‘||’ [-Werror=parentheses]
     if ((stream->n_samples == 1) && (stream->first_duration == 0)
         ~~~~~~~~~~~~~~~~~~~~~~~~~^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
2017-10-10 18:04:50 +03:00
Nael Ouedraogo
5874bafa4f qtdemux: fix assert when moof containing one sample
Avoid computing frame rate when a stream contain moof with only one
sample, to avoid an assert. The moof is considered as still picture.

The same is already done for one sample given in the moov.

https://bugzilla.gnome.org/show_bug.cgi?id=782217
2017-10-10 18:03:42 +03:00
Thibault Saunier
fb952e3470 rtspsrc: Avoid potentially dereferencing NULL pointer
CID 1418986
2017-10-09 14:39:14 +02:00
Tim-Philipp Müller
6cb51bd8cf rtpjitterbuffer: fix debug message on pt mismatch 2017-10-08 00:07:43 +01:00
Tim-Philipp Müller
d5f72418c8 rtpbin, rtspsrc: fix compiler warnings about 64-bit integer signednes
"warning: this decimal constant is unsigned only in ISO C90" with
gcc 4.8.4 (Ubuntu/Linaro 4.8.4-2ubuntu1~14.04.3)
2017-10-07 15:55:24 +01:00
Sebastian Dröge
263494f9c7 rtpsbcdepay: Fix potential NULL pointer dereference
CID 1418864
2017-10-07 14:06:38 +03:00
Sebastian Dröge
7d80fa4bbf audioecho: Micro-optimize
Gives 1.28x speedup in surround-delay=false mode
2017-10-07 03:03:46 +03:00
Sebastian Dröge
ad062e1879 audioecho: Don't do linear interpolation between samples
Linear interpolation adds quite some noise, and it's unlikely that
anybody will ever need sub-sample accurate delays. Proper resampling
before that will lead to better results.
2017-10-07 03:03:46 +03:00
Nicolas Dufresne
986f3e15b2 flvdemux: Only set pixel-aspect-ratio if specified
If it's not specified, we should let the decoder figure it out.
Apparently the code was already in place, all was to make the code
conditional.

https://bugzilla.gnome.org/show_bug.cgi?id=787795
2017-10-05 21:20:19 -04:00
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
Mark Nauwelaerts
d1d149ce82 qtmux: correctly track chunk size of subtitle stream
... thereby ensuring correct chunk offset tracking for all streams.
2017-07-01 18:59:07 +02:00
Julien Isorce
afbabaefbe rtpstats: fix assertion 'denom > 0' failed
gst_util_uint64_scale_int takes a gint as denom parameter
whereas ctx->clock_rate is a guint32.

It happens when gst_rtp_packet_rate_ctx_reset set clock_rate
to -1.

So just define clock_rate as gint like it is done in rtpsource.h

https://bugzilla.gnome.org/show_bug.cgi?id=784250
2017-06-29 15:58:44 -04:00
Sebastian Dröge
cd17c71dce rtspsrc: Create send/recv mutexes once, not on every connect()
Also fixes a crash caused by freeing an uninitialized mutex in an error
case.

https://bugzilla.gnome.org//show_bug.cgi?id=784282
2017-06-29 18:59:58 +03:00
Mark Nauwelaerts
83987b5037 qtdemux: specify '_swr' atom as a container atom
... so it is parsed as an mp4 style metadata atom as written by muxer
2017-06-28 20:00:44 +02:00
Mark Nauwelaerts
dbcb99ac1d qtmux: initialize mdhd language code as undefined 2017-06-28 20:00:44 +02:00
Nicolas Dufresne
bbe0053f8a rtpjitterbuffer: Add a faststart-min-packets property
When set this property will allow the jitterbuffer to start delivering
packets as soon as N most recent packets have consecutive seqnum. A
faststart-min-packets of zero disables this feature. This heuristic is
also used in rtpsource which implements the probation mechanism and a
similar heuristic is used to handle long gaps.

https://bugzilla.gnome.org/show_bug.cgi?id=769536
2017-06-28 11:51:10 -04:00
Mark Nauwelaerts
949902f1e2 qtmux: correctly calculate overall first_ts to ensure stream sync
... by minding and compensating for the dts_adjustment that may have
been introduced in the PTS timeline.
2017-06-24 17:36:54 +02:00
Mark Nauwelaerts
04fd953713 matroskademux: track highest known cluster position and time
... to use as a fallback initial duration estimate and to provide for
interpolation when scanning for position.
2017-06-24 17:36:54 +02:00
Mark Nauwelaerts
84c6aaf45b matroskademux: improve and simplify searching for cluster and position
... avoiding inefficiency proportional to file size
2017-06-24 17:36:54 +02:00
Mark Nauwelaerts
d7b736fa11 matroskademux: increase chunk size when scanning for cluster 2017-06-24 17:36:54 +02:00
Mark Nauwelaerts
b19ef0df30 matroskademux: maintain variable state when searching for position
... so skipping to next cluster happens efficiently
2017-06-24 17:36:54 +02:00
Sebastian Dröge
c438545dc9 rtspsrc: Actually use the receive lock when receiving, not the send lock 2017-06-22 11:39:16 +03:00
Mathieu Duponchelle
0da5679c6f rtspsrc: do not checksum the stream id
https://bugzilla.gnome.org/show_bug.cgi?id=783307
2017-06-16 17:30:11 +02:00
Tim-Philipp Müller
3a0fe9c2f5 qtmux: add support for muxing PNG
Demuxer already supported it.
2017-06-15 23:31:24 +01:00
Sebastian Dröge
a722f6e832 rtspsrc: Use a mutex for protecting against concurrent send/receives
We currently send data to the RTSP connection from multiple threads:
whenever a command is to be handled and whenever RTCP is generated. This
can cause data corruption or worse if both happen at the same time.

As such, protect gst_rtsp_connection_send() and gst_rtsp_connection_receive()
calls with a mutex. While this means that we hold a mutex during the IO
operation, this is not actually a problem as the IO operation can be
interrupted (gst_rtsp_connection_flush()) at any time and is blocking by
itself anyway.
2017-06-15 15:25:23 +03:00
Sebastian Dröge
deb9c62cd9 qtmux: Un-merge the last two stsc entries after serializing
The last entry will most likely get new samples added to it in "robust"
muxing mode, changing the samples_per_chunk and thus making it wrong to
keep the last two entries merged. It will run into an assertion later
when adding a new sample to the chunk.

Thanks to gdiener@cardinalpeak.com for the analysis of the bug and
proposal for a solution.
2017-06-15 11:50:44 +03:00
Sebastian Dröge
a82e38d607 wavparse: Actually clip to upstream size instead of size of the data chunk
There might be other chunks after the data chunk, so clipping the chunk
size with the data size can lead to a negative number and all following
calculations go wrong and cause crashes or worse.

This was introduced in 3ac119bbe2.

https://bugzilla.gnome.org/show_bug.cgi?id=783760
2017-06-14 00:11:17 +03:00
Vivia Nikolaidou
96cd941788 splitmux: Drop allocation queries
They can cause us to deadlock, while we're waiting for a new frame and
upstream is waiting for the allocation query to be answered before
sending a frame

https://bugzilla.gnome.org/show_bug.cgi?id=783753
2017-06-13 18:29:37 +03:00
Mathieu Duponchelle
f6283b082e rtspsrc: uniquify stream ids
https://bugzilla.gnome.org/show_bug.cgi?id=783307
2017-06-07 23:30:05 +02:00
Jimmy Ohn
55e0b18186 qtdemux: remove not needed code
remove not needed code about res variable.

https://bugzilla.gnome.org/show_bug.cgi?id=783422
2017-06-06 11:02:59 +02:00
Tim-Philipp Müller
18b53c2236 rtph265depay: fix caps leak 2017-06-02 11:30:15 +01:00
Tim-Philipp Müller
a68a7fb65d rtph264depay: simplify buffer accumulation control flow
There is no difference between pushing out a buffer directly
with gst_rtp_base_depayload_push() and returning it from the
process function. The base class will just call _depayload_push()
on the returned buffer as well.

So instead of marshalling buffers through three layers and back,
just push them from one place in handle_nal() and always return
NULL from the process vfunc. This simplifies the code a little.

Also rename _push_fragmentation_unit() to _finish_fragmentation_unit()
for clarity. Push sounds like it means being pushed out, whereas
it might just be pushed into an adapter.

This change has the side-effect that multiple NALs in a single STAP
(such as SPS/PPS) may no longer be pushed out as a single buffer if
we output NALs in byte-stream format (i.e. not aggregate AUs), but
that shouldn't really make any difference to anyone.
2017-06-01 17:32:17 +01:00
Juan Navarro
72d2afda18 rtpsession: print value of unknown RTCP Payload Type
This adds printing the actual value of any unknown RTCP PT
to the already existing WARNING log message.

https://bugzilla.gnome.org/show_bug.cgi?id=783248
2017-05-31 10:20:27 +03:00
Tim-Philipp Müller
a9f9166004 rtpopusdepay: minor perf improvements
Use the ::process_rtp_packet() vfunc to avoid mapping the
RTP buffer twice.

gst_rtp_buffer_get_payload_buffer() returns a new sub-buffer
which will always be writable, so no need to make it writable.
2017-05-24 16:41:45 +01:00
Tim-Philipp Müller
f9a740b319 rtp: opus: use existing utility funcs for copying/dropping metas
We had our own copies of those while the code was in -bad, but now
we can use the existing utility functions instead of re-implementing
them.
2017-05-24 16:18:01 +01:00
Tim-Philipp Müller
4a28e649c3 rtp: cache meta tag quarks and add more utility functions for metas
Every g_quark_from_static_string() is a hash table lookup serialised
on the global quark lock in GLib. Let's just look up the two quarks
we need once and cache them locally for future use. While we're at it,
add new utility functions for the two most commonly used tags
(audio + video). Make first argument a gpointer so we don't have to
cast and make the code ugly. These are used for logging purposes
only anyway.
2017-05-24 13:32:10 +01:00
vijay
810c0bb084 aacparse : Fix, Caps were not set while reusing aacparse
While reusing aacparse caps were not set.This fix enables aacparse to reuse in same pipeline.

https://bugzilla.gnome.org/show_bug.cgi?id=783027
2017-05-24 13:59:52 +02:00
Tim-Philipp Müller
e67a5a9f18 multifile: remove some cruft 2017-05-21 10:33:43 +01:00
Luis de Bethencourt
fb00a357cd multifilesink: fix property name in example pipeline
Since the move from CVS the property name of the documentation example
has been filename instead of location. Users trying the gst-launch
command as is will get:
no property name "filename" in element

Fixing it.
2017-05-20 11:51:33 +01:00
Edward Hervey
4f713717de isomp4: Safely ignore [skip] atoms
Instead of warning about them
2017-05-18 18:12:15 +02:00
Tim-Philipp Müller
6cb49cf2c3 Fix up package name and origin in some plugins 2017-05-18 10:44:07 +01:00
Nicolas Dufresne
b68d936ae0 Remove plugin specific static build option
Static and dynamic plugins now have the same interface. The standard
--enable-static/--enable-shared toggle are sufficient.
2017-05-16 14:41:19 -04:00
Vivia Nikolaidou
a910329f6d splitmuxsink: Add alignment-threshold argument
If a non-reference stream is behind the reference stream by an amount of
time smaller than the alignment threshold (in nsec), it counts as being
after it.

https://bugzilla.gnome.org/show_bug.cgi?id=782563
2017-05-16 15:30:52 +03:00
Vivia Nikolaidou
5f7fe63fea qtmux: Do not check timecode data for mp4 container
Timecode trak is only supported for mov right now, not for mp4. That
code would otherwise create an invalid trak if the muxed video contained
timecode metadata.

https://bugzilla.gnome.org/show_bug.cgi?id=782684
2017-05-16 15:16:58 +03:00
Sebastian Dröge
347e814074 qtmux: When accepting renegotiation, just return TRUE and change nothing
We only accept new caps if they are basically the same. We don't want to
reset anything as if the caps are new, otherwise various state could get
out of sync with the current run.
2017-05-11 20:01:15 +02:00
Sebastian Dröge
10f4693163 qtmux: In prefill mode, only pad buffers with > 0 sized memories as needed
Adding a 0-byte memory has not much effect.

Also add some debug output.
2017-05-11 19:21:22 +02:00
Sebastian Dröge
63492ee141 qtmux: Lateness is in QT timescale, diff in GstClockTime
Print the right one in debug output to get meaningful numbers.
2017-05-10 15:58:41 +02:00
Sebastian Dröge
163b62fc25 qtmux: Error out if a gap edit list has to be written in prefill mode
We don't have any space reserved for this in the moov and the
pre-finalized moov would have broken A/V synchronization. Error out here
now
2017-05-10 14:31:40 +02:00
Sebastian Dröge
c0f505e80e qtmux: Calculate with reserved moov size instead of last moov size
We have some padding added after the initial moov, so a bigger updated
moov can be handled to some degree and is expected. Previously we just
ignored the padding and errored out in cases when the padding would've
just been enough.
2017-05-10 11:42:09 +02:00
Sebastian Dröge
419f53259f qtmux: Error out directly if sending filler data results in a flow error
CID 1405994
2017-05-10 11:12:23 +02:00
Sebastian Dröge
3fcfe774e6 qtmux: In prefill mode, handle the case when only the first chunk was ever used 2017-05-09 16:02:43 +02:00
Sebastian Dröge
55f949676e qtmux: Add new prefill recording mode
This sets up a moov with the correct sample positions beforehand and
only works with constant framerate, I-frame only streams.

Currently only support for ProRes and raw audio is implemented but
adding new codecs is just a matter of defining appropriate maximum frame
sizes.

https://bugzilla.gnome.org/show_bug.cgi?id=781447
2017-05-09 14:17:04 +02:00
Sebastian Dröge
f0163a016c qtmux: Error out on discontinuities/gaps when muxing raw audio
When muxing raw audio, we have no way of storing timestamps but are just
storing a continuous stream of audio samples. If the difference between
the expected and the real timestamp becomes to big, we should error out
instead of silently creating files with wrong A/V sync.

https://bugzilla.gnome.org/show_bug.cgi?id=780679
2017-05-09 14:16:51 +02:00
Tim-Philipp Müller
d826c5ad96 matroskademux: improve index memory usage
Re-arrange order of index entry struct members to avoid padding
bytes in the middle of the struct, thus potentially reducing the
overall size of the struct and reducing memory used by the index.
On Linux x86_64 the size goes down from 32 bytes to 24 bytes for
each index entry.
2017-05-06 22:39:31 +01:00
Seungha Yang
66448be77f qtdemux: Fix crash on mss stream caused by invalid stsd entry access
Since mss has no moov, default stsd entry should be created with media-caps.

https://bugzilla.gnome.org/show_bug.cgi?id=782042
2017-05-02 14:20:02 +03:00
Sebastian Dröge
875fc630d5 qtdemux: Don't crash in debug output if stream==NULL
That case is correctly handled below but not in the debug output.

https://bugzilla.gnome.org/show_bug.cgi?id=781270
2017-04-27 12:56:27 +03:00
Sebastian Dröge
a593e4966e qtdemux: Don't perform seeks with inconsistent seek values
If gst_segment_do_seek() fails, we shouldn't try seeking on that
resulting segment but just error out. Crashes further down the line
otherwise.
2017-04-25 17:12:30 +03:00
Tim-Philipp Müller
e7e7f26200 rtph264depay: don't insert SPS/PPS inline for AVC output
SPS/PPS are in the caps in this case and shouldn't be in
the stream data.
2017-04-24 17:29:37 +01:00
Sebastian Dröge
c99f7579f3 rtspsrc: Chain up to the parent class' provide_clock() implementation
If no clock was provided directly by rtspsrc. This behaviour was removed
by f8013487c9 and results in rtspsrc not
providing the system clock via the rtpjitterbuffer.

As a result, if another element like an audio sink, provides a clock,
the pipeline would select that (when going to PAUSED/PLAYING again later).
Audio clocks usually don't progress in PAUSED, and thus our live source
won't be able to use the clock to produce data, making the sink never
preroll and everything is stuck.
2017-04-21 19:14:09 +01:00
Jürgen Sachs
7c658c3c04 qtdemux: reset sample_description_id to default
Fixes stream where sample_description_id is specified in the tfhd

https://bugzilla.gnome.org/show_bug.cgi?id=778337
2017-04-21 06:45:10 +02:00
Sebastian Dröge
fe77e2efe8 splitmuxsink: Don't use an explicit name for requesting audio pads
... unless the muxer uses the same audio pad template name as
splitmuxsink. We can't request a pad called "audio_0" on a muxer that
wants pads to be "sink_%d".
2017-04-20 13:17:33 +01:00
ChangBok Chae
3d19d6d8bc flvdemux: remove duplicated segment initialization
It's also done in gst_flv_demux_cleanup().

https://bugzilla.gnome.org/show_bug.cgi?id=779106
2017-04-20 12:42:58 +01:00
Xavier Claessens
6b42569ded splitmuxsink: Correctly catch FLUSH events in probes
https://bugzilla.gnome.org/show_bug.cgi?id=767498
2017-04-20 20:17:35 +10:00
Tim-Philipp Müller
50a4b5bc0d Revert "rtpbin: pipeline gets an EOS when any rtpsources byes"
This reverts commit eeea2a7fe8.

It breaks EOS in some sender pipelines, see
https://bugzilla.gnome.org/show_bug.cgi?id=773218#c20
2017-04-19 12:28:12 +01:00
Edward Hervey
58e3033747 qtdemux: Reset adapter in more discontinuity cases
In push mode we process as much as possible in the adapter. When we receive
a DISCONT buffer which we can't match to an actual sample (based on the existing
sample table) and there is still data remaining in the incoming adapter,there is
one of two cases happening:
1) We are doing reverse playback, in which case we should flush out all pending
  data
2) We have leftover data from the previous incoming buffer... which we can't do
  anything about.

For the second case, make sure we flush out the remaining data so that we can start
parsing again from scratch.

https://bugzilla.gnome.org/show_bug.cgi?id=781319
2017-04-17 13:10:06 +02:00
Edward Hervey
474819cc3d rtspsrc: Use GST_ELEMENT_ERROR_WITH_DETAILS
Allows the application to know the exact status code that was returned
by the server in a programmatic fashion.

https://bugzilla.gnome.org/show_bug.cgi?id=781304
2017-04-17 13:10:06 +02:00
Seungha Yang
3b9ea4b85f qtdemux: Fix leak on QtDemuxStreamStsdEntry
Fix unit test failure

https://bugzilla.gnome.org/show_bug.cgi?id=781362
2017-04-16 23:37:55 +01:00
Sebastian Dröge
e51c08b0a2 qtmux: Fix timescale of timecode tracks
They should have ideally the same timescale of the video track, which we
can't guarantee here as in theory timecode configuration and video
framerate could be different. However we should set a correct timescale
based on the framerate given in the timecode configuration, and not just
use the framerate numerator.
2017-04-14 13:38:53 +03:00
Edward Hervey
c6e9c67f63 qtdemux: Properly reset demuxer when all streams are EOS
Make sure offset and neededbytes are properly resetted when all
streams are EOS in push-mode.

Avoids cases when some data might still be pushed by upstream (because
it didn't yet see the resulting GST_FLOW_EOS yet) and qtdemux gets
completely lost.

https://bugzilla.gnome.org/show_bug.cgi?id=781266
2017-04-13 13:26:36 +02:00
Edward Hervey
7077425c3d qtmux: Don't forget to update pad->last_buf
buf is the current pad->last_buf value. If ever it gets copied/unreffed,
we need to make sure to write back the new  pointer to the last_buf
variable.

Fixes using wrong pointer values in the case of decrasing DTS value
2017-04-12 13:26:59 +02:00
Jürgen Sachs
9684c88c60 qtdemux: fix: sample description index override in tfhd not evaluated
https://bugzilla.gnome.org/show_bug.cgi?id=778337
2017-04-12 11:04:46 +02:00
Edward Hervey
26f9869668 qtdemux: Add out-of-bound check
Make sure we don't read invalid memory
2017-04-12 11:03:24 +02:00
Thiago Santos
9ac3861a9b qtdemux: move parsing of tkhd out of stsd entry loop
It needs only to be read once.
2017-04-11 23:42:56 -07:00
Thiago Santos
cf6733a55e qtdemux: check for a different stsd entry before pushing a sample
Before pushing a sample, check if there was a change in the current
stsd entry. This patch also assumes that the first stsd entry is
used as default for the first sample. It might cause an uneeded
caps renegotiation when this isn't the case.
2017-04-11 23:42:56 -07:00
Thiago Santos
86b427dc70 qtdemux: parse all stsd entries
stsd can have multiple format entries, parse them all.

This is required to play DVB DASH profile that uses multiple entries
to identify the different available bitrates/options on dash streams

The stream format-specific data is not stored into QtDemuxStreamStsdEntry
2017-04-11 23:42:55 -07:00
Thiago Santos
54e252e095 qtdemux: rework stsd sample entries access
Instead of using the stsd as a base pointer, use the actual stsd
entry as the stsd can have multiple entries. This is rarely used
for file playback but is a possible profile with in DVB DASH specs.

This still doesn't support stsd with multiple entries but makes it
easier to do so.
2017-04-11 14:13:56 -07:00
Thiago Santos
bd32bcc36c qtdemux: get stsd child by index instead of type
There might be multiple children with the same type
2017-04-11 14:13:56 -07:00
George Kiagiadakis
7f6c783930 rtprtxqueue: implement handling of the max-size-time property
https://bugzilla.gnome.org/show_bug.cgi?id=780867
2017-04-11 09:44:33 +03:00
Sebastian Dröge
d6b145df5f qtmux: Fix last_dts tracking for raw audio and similar formats
Accumulate the durations directly and don't scale yet another time by
the number of samples.
2017-04-10 15:55:54 +03:00
Lyon Wang
f26835d8bb scaletempo: Scale GAP event timestamp and duration like for buffers
https://bugzilla.gnome.org/show_bug.cgi?id=781008
2017-04-09 10:45:21 +03:00
Edward Hervey
9a066e548d aacparse: streamline and improve AudioSpecificConfig parsing
AudioSpecifigConfig is used in a variety of AAC streams but was
being parsed differently. Instead, make everyone use the same parsing.

* Remove unused 'bits' field (it was always set to 0 if present)
* Add proper GAConfig parsing (to know the  number of samples per frame
  if present).

Fixes wrong rate/channels configuration in streams coming from qtdemux

https://bugzilla.gnome.org/show_bug.cgi?id=780966
2017-04-06 12:04:46 +02:00
Edward Hervey
345c566552 aacparse: Handle Parametric Stereo with HE-AAC(v2)
According to ISO/IEC:14496-2:2009 , in the case of HE-AACv2 (audioObjecType
29) parametric stereo is used (a single mono track is used and then
transformations are applied to it to provide a stereo output).

We therefore report two channels in the case where there is one reported
in the audioChannelConfiguration.

Fixes the various issues where a demuxer would report two channels, but
then the parser would say there's only one channel, and then the decoder
would output two channels.
2017-04-04 18:16:39 +02:00
Sebastian Dröge
6f3737f0c5 qtmux: Simplify buffer refcounting in add_buffer() and remove unneeded NULL checks 2017-04-04 15:24:45 +03:00
Sebastian Dröge
6b889bea69 qtmux: Select the best pad based on the cached last_buf if any
last_buf is the one we're going to write next, not buf. As such we
should check timestamps against that one if there is one to select the
earliest pad.

Also remember the currently selected pad in the very beginning when
storing the first last_buf.

This both solves some edge cases where not the correct next pad was
selected corresponding to the target interleave.
2017-04-04 15:24:45 +03:00
Sebastian Dröge
eb307baff4 qtmux: Error out immediately if a timecode is to be written but downstream return not-OK 2017-04-04 15:24:45 +03:00
Edward Hervey
1e4cdc6d6f qtdemux: Update variables before early exit
This is an update of d78d589627

We still exit as early as possible in case of non-ok/non-unlinked combined
flow, but we first make sure that we update the internal position variables.

This ensures that if upstreams "ignores" the flow return (and carries on pushing),
we don't end up processing data with completely bogus variables/positions.
2017-04-03 11:37:52 +02:00
Douglas Bagnall
a9f26c2a14 interleave: avoid using uninitialised ordering_map
If self->channel_positions == NULL (which seems unlikely),
self->default_channels_ordering_map will be used unintialised.

We avoid that by keeping track of the channel_mask, which is set when
the ordering map is initialised.

https://bugzilla.gnome.org/show_bug.cgi?id=780331
2017-03-31 14:10:34 +03:00
Douglas Bagnall
c08d719453 interleave: don't overflow channel map with >64 channels
When there are more than 64 channels, we don't want to exceed the
bounds of the ordering_map buffer, and in these cases we don't want to
rempa at all. Here we avoid doing that.

https://bugzilla.gnome.org/show_bug.cgi?id=780331
2017-03-31 14:10:34 +03:00
Sebastian Dröge
2b50a338c3 qtdemux: In gap mode, consider the mdat offset when calculating the remaining mdat size
The mdat generally does not start at offset 0, we have to include the
size of the moof and whatever else was in front of the mdat.
2017-03-27 21:52:51 +03:00