Unlock locker when error occurs during webrtc_create()

[Version] 0.3.309
[Issue Type] ASAN defect

Change-Id: I62f7593054a8beaf1b411d5d214a51b6f5f93ef9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Fix not to manage pt values in case of webrtc_media_source_set_payload_type()

If this function is used to set a payload type value, the value of
the parameter can be used even if it is already assigned by other one.

It is cherry-picked from 6c85f708f6c112dd80ecbb3cffc90b246b0b7b32
commit of tizen 9.0.

[Version] 0.3.308
[Issue Type] Improvement

Change-Id: I143beac2ef5737bec77c68c9d789f3ee905ca5a7
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add webrtc_set_sw_video_decoder()

It'd be useful when an user wants to use sw video decoder to a particular
track id even if the system supports a hw decoder by configuration.

[Version] 0.3.307
[Issue Type] Internal API

Change-Id: I1bb5c12422c5e7a901abb9ed54d00b84c57c00f1
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Improve documentation

Missing privilege information has been added to
webrtc_screen_source_set[unset]_crop().

[Version] 0.3.306
[Issue Type] Doxygen

Change-Id: I763d92e19dc0f904b32115f0c43bbc23fcb8631b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_data_channel: Fix use-after-free issue

(cherry-picked from tizen branch)

[Version] 0.3.305
[Issue Type] ASAN defect

Change-Id: Idae9729c00520ca6aa55655fb795c0c9175ed6c0
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

test: Add test case for webrtc_set_bundle_policy_max_compat()

[Version] 0.3.304
[Issue Type] Test application

Change-Id: I7c87d359b8882558c706cf4fe9d01056892e7bdd
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add webrtc_set_bundle_policy_max_compat()

[Version] 0.3.303
[Issue Type] Internal API

Change-Id: I22f27c83598b34eb51d46675493ba807dbf39799
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_display: Destroy tbm bo list when bo size has been changed

It is related to dynamic resolution change in case of EVAS rendering.
It has fixed the crash due to the invalid bo size when facing this case.

[Version] 0.3.302
[Issue Type] Bug fix

Change-Id: Idd0ef5467acf39ca667c5a6b41505858e12f06d2
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_transceiver: Add more information about H264

ST signaling server requires this for NEST doorbell. So it is added.

TIZEN_DA build definition has been introduced for DA compiling option.

[Version] 0.3.301
[Issue Type] Condition added (for product)

Change-Id: I9a40779f5fb70f534b62a7dd37f2261ea9a76b17
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

test: Add missing lock

[Version] 0.3.300
[Issue Type] Coverity defect (MISSING_LOCK)

Change-Id: Id969572e1896b660b620216ecb376a43d4a891fd
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Set payloader type only if it is not occupied

Codes for occupying and releasing payload type have been added in
webrtc_media_source_set_payload_type().

Checking the media_type parameter of webrtc_media_source_get_payload_type()
has been added to fix coverity defect.

[Version] 0.3.299
[Issue Type] Improvement

Change-Id: I9eaf492456d24977d300d0682bcbbf60acd0f9ef
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit 65cecdf58ca7f90bbf1774f944d0eb54db93ff0b)

webrtc_transceiver: Fix unwanted payload type number order

This patch fixes the issue where there is another media source added
before the null source, the payload type number gets reversed because
the payload type number of the null source type was obtained when calling
webrtc_media_source_set_transceiver_codec() whereas other source type
get it when calling webrtc_start().

[Version] 0.3.298
[Issue Type] Bug fix

Change-Id: Ib438dc280c843ddf8652d114a4ced23319f73b71
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add support to show multiple video tracks from one remote peer

Test condition:
 1. peer to peer test (not using room menu)
 2. display type is EVAS
 3. it should not exceed more than 3 remote video tracks.

[Version] 0.3.297
[Issue Type] Testsuite

Change-Id: Idd081077d97a6bab538d49a0f5a3c282079224b9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Fix doxygen

Missing @internal tag is added.
An error type that does not occur is removed from the description.
@Remarks for releasing handle are added.
Mis-typed function name is fixed.

[Version] 0.3.296
[Issue Type] Doxygen

Change-Id: I045d70214b458d67cab8aee8d926e38cec68ba84
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Remove meaningless parameter checks

The first parameter of callbacks from gstreamer would be always not null.

Once webrtc handle has been created, there's no possibility that
'gst.webrtcbin' or 'gst.source_slots' is null.

A typo is also fixed.

[Version] 0.3.295
[Issue Type] Clean up

Change-Id: I03fc688b1b694c86c77d958fe4729734b799b897
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add webrtc_set_display_qos()

[Version] 0.3.294
[Issue Type] Internal API

Change-Id: I2a644795109b9534fc1ad65499c3ffb62c7ea3ec
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_display: Fix errors when setting EVAS display mode/visible

These are fixed by using checking display type that shows
whether if the mm_display_interface_set_display_mainloop_sync()
is performed or not.

[Version] 0.3.293
[Issue Type] Bug fix

Change-Id: I070318bb73be96abd07da3d0ec419c0703c83109
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add webrtc_set_display_surface_id()

[Version] 0.3.292
[Issue Type] API

Change-Id: I8678951f128dcb448645e1032838c4392f9e2a20
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Save mute value and apply it when audiosink is created

[Version] 0.3.291
[Issue type] Improvement

Change-Id: Icb5181afd2c759756187cd2a68ce4fa315c84c15

Use g_autoptr for JsonParser variable

[Version] 0.3.290
[Issue type] Refactoring

Change-Id: Iaf7fac85e87150abdfae106827fb84b7c41d001e

webrtc_private: Print debug message in GstMessage object

[Version] 0.3.289
[Issue type] Logs

Change-Id: Ied0efd7e522cddbf3b1f291aec24b78deb91ace7

Use GST_TIME_FORMAT for buffer timestamps

[Version] 0.3.288
[Issue type] Logs

Change-Id: Ibc961bf988aa9225b5f4f4d1d7df2a381fbc459c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_stats: Dump stats result into the file

The JSON data of stats result will be written into
/tmp/webrtc-stats-[pid]-[handle address].dump
if 'stats log period' field in the ini file is set to some values.

[Version] 0.3.287
[Issue type] New feature

Change-Id: If66b7936166fdb9412f748f38a5374fd472df677

webrtc_stats: propagate the user cb return using the return of gst_structure_foreach()

[Version] 0.3.286
[Issue type] Refactoring

Change-Id: If7e8dabfc8fb87f4a93259b81a37f47f3b01608b

webrtc_stats: Add null check code before calling gst_structure_foreach()

It is to avoid printing warning message from a console.

[Version] 0.3.285
[Issue type] Improvement

Change-Id: I52f0c9ad3f7c218b5af9d70706625a1e7848fcbb

fixup! webrtc: Fix crash issue

This modification is intended for webrtc_create_offer[answer]_async().

Change-Id: Id38abb95de712288a46b88976b6f9843042386da

webrtc_dns: Check error of getifaddrs()

[Version] 0.3.284
[Issue type] Coverity defect (CHECKED_RETURN)

Change-Id: I8cfa281059b5b8efc5b2f6660bd89bcb475bfef2
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc: Fix crash issue

Some codes associated with the FEC from the patch
- 'webrtc: Add missing codes for updating transceiver'
are partially reverted due to the C# TCT crash.

It is fixed with GST 1.22 on Tizen 7.5.

[Version] 0.3.283
[Issue type] Avoid crash

Change-Id: Ida6d8cebe909c94bb1acc5c5c07c86649c09f18b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_stats: Fix memory leak

[Version] 0.3.282
[Issue type] Resource leak

Change-Id: Ia4d3a70b80a4eb69b0052dc36f9875b8598dd5d9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_dns: Use version 4 UUID as a unique hostname

It is mentioned in 3.1.1. of the link below.
: https://www.ietf.org/archive/id/draft-ietf-mmusic-mdns-ice-candidates-03.html

[Version] 0.3.281
[Issue type] Improvement

Change-Id: Ia4665c17fd812fb61d0e5f8c33c3ef7ecc97955a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit 5a1a61f89bb969e8fe807123677e7decb0a4e202)

webrtc: Add missing codes for updating transceiver

The same codes from webrtc_create_offer[answer]() are added
to webrtc_create_offer[answer]_async().

[Version] 0.3.280
[Issue type] Bug fix

Change-Id: I582139cff77237ec9237a6473cdd7efd347a3b47
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_dns: Consider more IP address ranges as private IP

Address ranges below are considered as private IP.
 Class A: 10.0.0.0 - 10.255.255.255
 Class B: 172.16.0.0 - 172.31.255.255

[Version] 0.3.279
[Issue type] Improvement

Change-Id: Ib65e55aed1a7524cd77d41c1376a3452cb5946b7
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Replace private IP with hostname in local ICE candidate

[Version] 0.3.278
[Issue type] New feature

Change-Id: I351db37c3a1fafec9b63bb10840b145f8eb0fa53
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_ini: Add new item to enable to conceal private IP address

Default value is false. If 'conceal private ip' in ini file is
set to yes, then hostname for the private IP address will be used
in each local ICE candidate.

e.g)
[general]
conceal private ip = yes

[Version] 0.3.277
[Issue type] New feature

Change-Id: Ida5eb1db00d40661cb32c6501515874a85246328
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_dns: Add support for registering DNS service

This patch uses dns_sd.h of mdnsresponder to register DNS service
and to publish hostname for private IP address.

[Version] 0.3.276
[Issue type] New feature

Change-Id: I9a9d64c367b2a49f2fe6e2ca7d1cd3ceca91770d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit cbb39ec1054d882ea9619b4d91b59493f5e68ccd)

webrtc_sink_snapshot: Create queue before creating convert thread

Sometimes the new thread tries to access queue not created yet which causes
a crash. It is fixed to create queue first and then create thread as well as
check if the result of pop operation is NULL.

[Version] 0.3.275
[Issue Type] Crash

Change-Id: Icefd4d6d9b38141384fbc70b3a7097a6f674fd4d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink_snapshot: Remove switch-case

Instead, bit-wise check is used whether to encode data or not.
Some functions to convert type are added.

[Version] 0.3.274
[Issue Type] Refactoring

Change-Id: Ib05495d97ef390429440a3c0e049e5018868758c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add menu to take snapshot

[Version] 0.3.273
[Issue Type] Add

Change-Id: I2d869ff6e7db5dd3e367b4a24fd31321a2dc51d5
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add webrtc_take_snapshot() API

[Version] 0.3.272
[Issue Type] Internal API

Change-Id: I9750f05f7a64d414b9280644ebb6c84afb071417
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink_snapshot: Support PNG format

[Version] 0.3.271
[Issue Type] Add

Change-Id: I663ea2cf682492929745e3b9022ec86c9e5c4051
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink_snapshot: Support JPEG format

Location of locking mutex is moved to guard variables properly.

[Version] 0.3.270
[Issue Type] Add

Change-Id: Iad94e4309c72e6c603d7249ad3460a18feeae9cc
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink_snapshot: Adopt a thread for converting video format

[Version] 0.3.269
[Issue Type] Improvement

Change-Id: If60e49be6aa87f0011fd3ab4ae10bb10a03a150c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit 91c98f1fdac5f6bc3658dc6274d830f3ece1f624)

webrtc_sink_snapshot: Add support for capture video frame

A video frame of receiving video stream can be captured by this new
functionality.

As of now, it only supports
1. to capture the decoded video frame and convert it only to RGB24 format.
2. to save dump file only with a macro definition.

These could be expanded to get improved and have more functionalities
next time.
1. support a user callback.
2. support various output formats.
3. adopt a dedicated thread for converting format.

[Version] 0.3.268
[Issue Type] New feature

Change-Id: I662f41ef639182b78f2ffc95d7e036625cbe5c53
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit 4bb87effbb964b1d6e6ece2604a6aac053d7fe99)

webrtc_test: Add menu for new internal APIs tests

1. set/get audio track mute
2. get video track resolution

[Version] 0.3.267
[Issue Type] Add

Change-Id: I953ee0f0b3b1d982c393ada0bd4058228d981307
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add webrtc_get_video_resolution() API

Missing preconditions are added to doxygen of some functions.

[Version] 0.3.266
[Issue Type] Internal API

Change-Id: I3ccfbf39e122a753023dafe16e5a13f36513a039

webrtc_sink: Update received video frame resolution

[Version] 0.3.265
[Issue Type] New feature

Change-Id: I3ec24e1a964e884d51263cb9e3966db7374eee0a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

test: CMakeLists: Apply variable for ui dependencies

[Version] 0.3.264
[Issue Type] Refactoring

Change-Id: Ia6e36badf26b128be9c67580eb6eba99aba75f8c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit 12237bd6539a0c37b32ebec5009783ec5d21a3da)

Check if handle is destroying before posting idle-callbacks

Sometimes, deadlock happens in C# TCT because of a sequence below.
 - webrtc handle seems to be destroyed before negotiation
  callbacks have not been processed in the main thread posted
  by g_idle_add_full().

It is checked that state and error callbacks have no chance
to get this situation.

[Version] 0.3.263
[Issue Type] Improvement

Change-Id: Ia7359bb45aaa1095dd77ee673c5b3b72056da62b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add functions for audio track mute

It's cherry-picked from tizen_6.5 branch.

[Version] 0.3.262
[Issue Type] Internal API

Change-Id: Ia942f612e126c7a5a2bc6b44dcd52bd41be7816a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Generate dot file when an error callback is invoked

[Version] 0.3.261
[Issue Type] Debug

Change-Id: I279ac95a82ee5bdea1e2c83ae31f6e59e0c9a8ae
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Fix SVACE defect (SEC_INVALID_ASSIGNMENT_WITH_UNSIGNED_VALUES)

[Version] 0.3.260
[Issue Type] Svace

Change-Id: Idfcfeeda1487ac0eb2cf3b71db0c7316052842d4

webrtc_sink: Change location of invoking _apply_display()

It is moved to outside of webrtc_set_display() which previously
locks a mutex.

[Version] 0.3.259
[Issue Type] Improvement

Change-Id: I252300784648941c136cd2b517d2fa81b51d9d8d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_ini: Add new item for video h/w encoder caps extension

e.g.)
[source camera]
video hw encoder element = v4l2h264enc
; comma separated list of 'field-name=value' that will be set to the capsfilter
video hw encoder caps ext = profile=constrained-baseline,level=3

Some video hardware encoder could require specific fields for
its caps. For example, level, profile, and so on.

A capsfilter for encoder is added.

[Version] 0.3.258
[Issue Type] Improvement

Change-Id: I2433db5f11b33712e2b720956a57f7bf340fc0b1
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_doc: Add '#' to stats enum

[Version] 0.3.257
[Issue Type] Documentation

Change-Id: I2f13e939588f937d0308617e8c2756d0a4b5268f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Move _gst_set_element_properties() to _source_private.c

[Version] 0.3.256
[Issue Type] Refactoring

Change-Id: I2ce2ade6789480d6b9485815d89d56734b6bd165
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Update doxygen and log regarding ICE candidate message

[Version] 0.3.255
[Issue Type] Doxygen / Log

Change-Id: If8b59c0d09500f3b6e981c8a0c92c3540347f8db
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit b78813bab3559eb73312f0f3c683b218c3733b54)

remove raw_format from source structure

Change to get raw_format from ini only when it is needed.

[Version] 0.3.254
[Issue Type] API

Change-Id: I9aba23be4a85964a9a6ffc6dc840e3003823f7b4

webrtc.h: Update webrtc_screen_source_set_crop() API description

[Version] 0.3.253
[Issue Type] API

Change-Id: I5d1cc4c6727eabd7facacaf3c4c3838056d9be44

Make a function to set origin video source resolution

[Version] 0.3.252
[Issue Type] Refactoring

Change-Id: Ifc3a95a78c3265d34f6cb9b825ee6bc76a8ad0e5

webrtc_source_screen: move resolution setting code for screen source to webrtc_source_screen.c

setting the default video resolution to screen resolution is only required for screen source.

[Version] 0.3.251
[Issue Type] Refactoring

Change-Id: I25dbf22a6d8a8469705e6fee7c69bbd1369b2a32

webrtc_source_screen: move screen source related code to webrtc_source_screen.c

[Version] 0.3.250
[Issue Type] Refactoring

Change-Id: Ib1f9d05448de4e0f26a705195323125fd4bd9cd6

webrtc_display: Return NULL in case of mm_display_interface_init() failure

[Version] 0.3.249
[Issue Type] Bug fix

Change-Id: I389a9a8e99e5ebfd2544a641a1b30f61c8bd155f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source_private: Reduce code complexity

[Version] 0.3.248
[Issue Type] Refactoring

Change-Id: Iecdd0ed61d19c2899085614e96a9a71204ad54dd

Make internal APIs for crop screen source public

Below functions are moved and have platform privilege.
 - webrtc_screen_source_set_crop()
 - webrtc_screen_source_unset_crop()

[Version] 0.3.247
[Issue Type] API

Change-Id: Iad8d3b9842db4c033052e62596c5a79610de22b1

add videorate to support dynamic framerate change

It doesn't support increasing framerate.

[Version] 0.3.246
[Issue Type] Improvement

Change-Id: Iea5ce0881ae55e1d881d847b88ff41dcede75a6f

Apply restriction for use of WEBRTC_MEDIA_SOURCE_TYPE_SCREEN

Locations of macro to exclude lines from coverage measurement
are also modified.

[Version] 0.3.245
[Issue Type] API

Change-Id: Ibbe0bf8a32f2c61b3734690cbe9068809d88349d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source_screen: sensor API dependency change

[Version] 0.3.244
[Issue Type] package change

Change-Id: I8df9aad81f4d28b86a2503f3a7261f5b44389395

webrtc_source_camera: Add CAMERA_DEVICE_TYPE_DUMMY for videotestsrc element

A tizen emulator uses 'videotestsrc' element as a camera source.
Now, we use 'pattern' property for device id in this case.

This fixes UTC failures - set/get camera device id.

[Version] 0.3.243
[Issue Type] Bug fix / UTC failure

Change-Id: I36ca271ebb5543fdb88cc197b7cd3fc05936f59a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_transceiver: Add encoding-params(2) to caps in case of OPUS codec

It represents channels.
In case of gst to gst, it worked fine without it. But in case of gst
to web, it does not work(web API failure).
It is also mandatory according to RFC7587, so it is added.

These codes are related to OPUS media of 'recvonly' in offer description.

[Version] 0.3.242
[Issue Type] Improvement / Compatibility

Change-Id: I57596fbb5b9b7c99efe856e5127de36d89ef6d88
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_transceiver: Fix to add missing payload type of an offer description

When transceiver direction was set to 'recvonly', a particular transceiver
was added with a preference caps without payload type. When trying to
create an offer description, the payload type must be added to the caps
regardless its source type.

[Version] 0.3.241
[Issue Type] Bug fix

Change-Id: I924b297a3a23522ca7466c45eb0d65de1972d82c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add more macro to exclude lines from coverage measurement

[Version] 0.3.240
[Issue Type] Line coverage

Change-Id: I0a9b3a680c19674e8cf33a7e03447cae4879186c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Move some structures to the proper source file

[Version] 0.3.239
[Issue Type] Refactoring

Change-Id: Id3bfa94408aeaad06a19159dfdb33f2b06f04d53

webrtc_test_validate: Add 'channel-mapping-family' value to appsrc caps for OPUS codec

It fixes not-negotiated issue on appsrc.

[Version] 0.3.238
[Issue Type] Bug fix

Change-Id: I2e256e7d7abff69e1e1bec6c1c2da025b5923af0
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Revise webrtc_media_source_set_payload_type() for media packet source

This function should affect media packet source. It is fixed.

[Version] 0.3.237
[Issue Type] Bug fix

Change-Id: I12b49da60526d2ba93cef07e1ceea9b4e887df33
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source_screen: Use mm_display_interface_get_screen_size() instead of elm_win_screen_size_get()

This will remove the dependency of elementary lib.

[Version] 0.3.236
[Issue Type] Dependency

Change-Id: I0ee14f4c100edf1c4b5aec02732768eb82acbf54
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source_file: move file source related code to webrtc_source_file.c

[Version] 0.3.235
[Issue Type] Refactoring

Change-Id: I2df68784dcb962533faa95c3661a99f12aec9f0a

webrtc_source_private: Add function to release request pad

_release_request_pad()

[Version] 0.3.234
[Issue Type] Refactoring

Change-Id: Id6837eb82fae01c28a02c14f71c1e3fa47427488

webrtc_source_loopback: Add some functions to update loopback

Below functions added.
 -_set_need_decoding_for_loopback()
 -_destroy_looopback_render_pipeline()

[Version] 0.3.233
[Issue Type] Refactoring

Change-Id: I94b7ac49fb9be0d830e701d6000bbbbe1eeafbb1

webrtc_source_screen: Check return values and release resource of sensor API

[Version] 0.3.232
[Issue Type] Coverity defect(CHECKED_RETURN) & bug fix

Change-Id: Ic5f90f54c1bd070101da6e1433d186d5444dd305
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add webrtc_source_loopback and move related codes to it

[Version] 0.3.231
[Issue Type] Refactoring

Change-Id: I7c402b2db477f87436dd6e6f0d463a06419e3cd0

CMakefile: Exclude capi-system-sensor dependency when using TIZEN_TV

sensor-internal.h could not be found with tv build.
(it is found by QB.)

[Version] 0.3.230
[Issue Type] Build dependency

Change-Id: I56c0eb28b251d3de9a701f31d326a42dd4d8d6ca
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Move some APIs to proper source file and make them static

[Version] 0.3.229
[Issue Type] Refactoring

Change-Id: I377c6c12b46d0d4ec80aa77be773efb19309c155

Add webrtc_transceiver.c and related codes are moved into it

[Version] 0.3.228
[Issue Type] Refactoring

Change-Id: Ic010a677849bc9253e3fcb0459af2a8dcf1eb9d5
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test_espp: Revise logic to submit audio/video packets properly

Following steps below are required for espp.
1. activate both stream types
2. invoke prepare async
3. after receiving ready to prepare callback for each stream type
4. submit packets for each stream type
5. finally, start espp handle when prepare async done callback is invoked

[Version] 0.3.227
[Issue Type] Bug fix

Change-Id: I3a72b9ab9088e5a0241b4da49b6dfe3ecb1319c1
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_ini: Add functions to get value from ini

[Version] 0.3.226
[Issue Type] Refactoring

Change-Id: I78e4fb36d6562ed0395a7c9b6bf26cdd4ad35e2f

remove redundant code

Plus, apply tizen coding rule.

[Version] 0.3.225
[Issue Type] Refactoring

Change-Id: I31ac8625327a316cafdf3db7aeeeffc150d6c512

fixup! webrtc_sorce_screen: Destroy rotate sensor listener handle

Rotate sensor listener handle must be destroyed before destroying webrtc.

Change-Id: I0537f37ba387aa3a6fec08eb39afd94069d23c26

webrtc_source_screen: move pad probe position for loopback on crop

screen source can have crop element, so the video stream for loopback should be collected on crop element.
and proper resolution after crop should be applied to video loopback.

[Version] 0.3.224
[Issue Type] Improvement

Change-Id: Ie7585d3115cad392fe272a254ff460ee0e92b353

Fix doxygen error

@ is doxygen command.
To avoid this error it needs to add an @ before @.
(It is reported and guided by API team)

[Version] 0.3.223
[Issue Type] Doxygen

Change-Id: Ie79b8018e19b3bb44d2cf8c1d59d84fe3487c211
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

CMakefile: Add capi-system-sensor dependency only when using TIZEN_FEATURE_UI

This library is only used in case of using screen source.

[Version] 0.3.222
[Issue Type] Build dependency

Change-Id: I374cd02c2b7820b5ab39d992e7417e6d7aa31668
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink/source: Fix resource leaks

[Version] 0.3.221
[Issue Type] Coverity defect

Change-Id: I318bcb7d6e962df59c36988d02d89d785dc23a7b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add more functions with @see doxygen commnad for webrtc_add_media_source()

[Version] 0.3.220
[Issue Type] Doxygen

Change-Id: Ia787c48506065e5f3e43ed13d79d492e58173698
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test_espp: Fix issue about calling esplusplayer_prepare_async()

This patch fixed issues below
 1. _prepare_async() function is called too early
 2. Occasionally, redundant _prepare_async() function call happens

[Version] 0.3.219
[Issue Type] Bug fix

Change-Id: Iaca99089df929d385cf535a6accd4a0b80fc6a4c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add more error conditions to webrtc_media_source_set_payload_type()

Conditions are added to return WEBRTC_ERROR_INVALID_PARAMETER.
 1. check if the media type is valid.
 2. check if the codec only allows fixed payload type or not
  and check the input parameter value.

[Version] 0.3.218
[Issue Type] Improvement

Change-Id: I711f0d7ffc9d88e634d93656f91e3789e13c928f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

spec: Exclude headless test package when tizen_espp_render option is set

Before this patch, when using 'tizen_espp_render' option, a build error
regarding webrtc_test_headless occurs. It is fixed now.

[Version] 0.3.217
[Issue Type] Bug fix / packaging

Change-Id: I5c242fcdddbd60e042bcbd00358c5204dd2774e3
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: crop and resolution change cannot be done at the same time

The results can vary depending on the order of resolution change and crop.
Therefore, those cannot be set at the same time.

[Version] 0.3.216
[Issue Type] Improvement

Change-Id: I305e1f9e72ce43629d733e31ec5d18cb005b1a9e

webrtc_source_screen: remove screen mode parameter from webrtc_screen_source_set_crop()

Check if the screen is rotated directly through the sensor.

[Version] 0.3.215
[Issue Type] Improvement

Change-Id: I3067c8ffe651a4b2ac7230ee8adeb91d2f39dc67

webrtc_test: Fix deadlock issue

Without stopping push packets to a media packet source and then
when trying to remove the source or to destroy webrtc handle,
it was blocked. This issue is fixed.

[Version] 0.3.214
[Issue Type] Bug fix

Change-Id: I6c8a5710b1772cf1756f99db8de1b7f07411f442
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Remove redundant codes in switch-case

Some codes are moved to the top or the bottom of the function
to reduce lines.

[Version] 0.3.213
[Issue Type] Refactoring

Change-Id: I22889884caf3f562703b557824f5b0dddb320cff
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Get mlineindex from source index

It fixes an issue that FEC is not applied when webrtc handle with
source(s) is operated as an answerer.

Due to the implementation of webrtcbin, only a transceiver created
by webrtcbin itself has a valid prop value of mlineindex. Others have
-1 value for this. Hence, _update_transceivers_fec() is now using an
index of sources as a mlineindex.

[Version] 0.3.212
[Issue Type] Bug fix

Change-Id: I4e3ead3413effe202e3034d88ec6bf7f2aa0f221
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>