webrtc_internal: Add functions for audio track mute

[Version] 0.2.184
[Issue Type] Internal API

Change-Id: Iebd221fe10f32f8a6ab5ccb7c20d70d475d1dee2
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Update doxygen and log regarding ICE candidate message

[Version] 0.2.183
[Issue Type] Doxygen / Log

Change-Id: If8b59c0d09500f3b6e981c8a0c92c3540347f8db
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: 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 NULL source type with OPUS codec.

[Version] 0.2.182
[Issue Type] Improvement / Compatibility

Change-Id: I9df43b5f0091b78f4b9356d127b02ec1cc81eba0
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Check prefix of STUN/TURN server URL

Doxygen is also improved.

[Version] 0.2.181
[Issue Type] Improvement

Change-Id: Ia3d2b0991cdef41709cd2d5c19d8e0c82bf09a40
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Improve _check_and_encode_turn_url()

Becuase a password could be encoded by base64,
it also needs to apply uri encoding.

[Version] 0.2.180
[Issue Type] Improvement

Change-Id: Ic57be7d44791e120d60abca9a3f1f83407e8fbcf
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Apply URL encoding when username of turn server URL has ':'

The form of URL should be turn(s)://username:password@host:port.

If the username has ':', for example '1221435:someidstring',
this could not be applied properly inside of webrtcbin.

In this case, this patch fixes it with using URL encoding
to avoid this situation.

[Version] 0.2.179
[Issue Type] Bug fix

Change-Id: Icd30fdbea39469526abde8016745fc291bf2d4a5
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

CMakefile: Revise file exclusion pattern to include webrtc_internal.h

Devel package must have the internal header.

[Version] 0.2.178
[Issue Type] Bug fix / packaging

Change-Id: I4deb602313cbcd0f0adf01cea19a316958c11fba
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add test cases for bundle policy

Menu items below are added.
 sbp. Set bundle policy
 gbp. Get bundle policy

[Version] 0.2.177
[Issue Type] Add

Change-Id: I1d25c19a2bb3ee8f427b8737aeaf5bd84827033d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add internal API to set/get bundle policy

Enums are added as below.
 - WEBRTC_BUNDLE_POLICY_NONE
 - WEBRTC_BUNDLE_POLICY_MAX_BUNDLE

Functions are added as below.
 - webrtc_set_bundle_policy()
 - webrtc_get_bundle_policy()

[Version] 0.2.176
[Issue Type] API

Change-Id: Ie3a66548f4f0300023ab24a23b84312cd6c888f8
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add test cases for WEBRTC_MEDIA_SOURCE_TYPE_NULL and transceiver codec

[Version] 0.2.175
[Issue Type] Add

Change-Id: I687be28990fbef0b5317c7b3517f2d0d28ef8149
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add internal API regarding NULL source type

New internal enum is added as below.
 : WEBRTC_MEDIA_SOURCE_TYPE_NULL

New internal functions are added as below.
 : webrtc_media_source_set_transceiver_codec()
 : webrtc_media_source_get_transceiver_codec()

Not like, tizen 7.0 branch, webrtc_media_source_set_transceiver_codec()
only supports WEBRTC_MEDIA_SOURCE_TYPE_NULL.

[Version] 0.2.174
[Issue Type] New feature

Change-Id: I95a21b6044b6b3c4557879e39dd42ff428dabc05
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Fix mute error for camera source which doesn't use tizen memory

[Version] 0.2.173
[Issue Type] Bug fix

Change-Id: I9bffc47f70d5192f0b4d67e136709ab03a61532c

webrtc_ini: Remove global varible for verbose log

It is replaced with new function.

[Version] 0.2.172
[Issue Type] Refactoring

Change-Id: I4591a4e588f080c625523d8c0d0c0542cace2afc
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Use PA_PROP_XXX defines instead of hard-coded string

[Version] 0.2.171
[Issue Type] Improvement

Change-Id: I76d4f7e1af27e01bd8beb2fd2a1e228a77ddbc58
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Change execution label for webrtc_test

[Version] 0.2.170
[Issue Type] Smack label

Change-Id: I8a7195ce447332a1de2d96de6e67f6e443e328e4
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Clear event source not fired before overwriting it

It was an issue with a short test case that results a crash in sometimes.

[Version] 0.2.169
[Issue Type] Bug fix

Change-Id: Ic82742df40438d7077d7f44585099d4694d0f707
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Fix crash when handling callback in idle

It was possible to access freed memory in log.
The crash rarely happened during ITc_webrtc_create_offer_async_p().

[Version] 0.2.168
[Issue Type] Bug fix

Change-Id: Ib1da621b4c2a853f63446454b356332fd8aaed83
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Fix negotiation state bugs

Setting the result state is moved inside __idle_cb().
Invalid converting enums are also fixed.
Getting the state in the callback is added to webrtc_test.

[Version] 0.2.167
[Issue Type] Bug fix

Change-Id: If91bae0f87397d7b9d7350bdf24f93c34a4e3e7c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Check return value of g_io_channel_read_chars()

[Version] 0.2.166
[Issue Type] Coverity defects

Change-Id: I0320f4b95c94da0dec4ef2f447c90d6b561aa6c1
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Revise description

webrtc_doc.h
 - Fix invalid information

webrtc.h
 - Add @remarks to callback function prototypes

[Version] 0.2.165
[Issue Type] Doxygen

Change-Id: Iac7524e8fcee20341a147d1c8eaefb58cfec1035
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add missing required libraries for pkg config

[Version] 0.2.164
[Issue Type] pkg-config

Change-Id: I3e1bbe2957379c9a58e2fae7e5e22014f837971e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Add omitted lock/unlock mutex for g_cond_signal()

This ensures to call g_cond_wait_until() before sending the signal.

[Version] 0.2.163
[Issue Type] Bug fix

Change-Id: I78b799067cf3f6a4a45ddf58c9341e679415a079
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Fix data type to prevent integer overflow

[Version] 0.2.162
[Issue Type] SVACE

Change-Id: I5722a5c8f5fc8ef0b4d8200ff2d74113cc06037c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add more mutex guard for callbacks

[Version] 0.2.161
[Issue Type] Improvement

Change-Id: I1f536532c3c4bac4101d68c125197d80b267856f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink/source: Add const keywords

[Version] 0.2.160
[Issue Type] Improvement

Change-Id: Icd1575632033d62f245b29d4a7a8528ef879a02e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_data_channel: Remove unreachable code and revise error log

[Version] 0.2.159
[Issue Type] Improvement

Change-Id: I3fb502f8ea712233a660e382904a65fd2e47fee0
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Revise doxygen

Post command regarding error callback is described in case of
failure on sending data via data channel.

[Version] 0.2.158
[Issue Type] Doxygen

Change-Id: I3fa1e5f84b25a35bda9260292945889dda429a9c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Change gcov object install path

[Version] 0.2.157
[Issue Type] Gcov

Change-Id: I10061f55df49d2c7cc4ae43de352cd46808b1e82
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add omitted error checking in webrtc_create()

[Version] 0.2.156
[Issue Type] Improvement

Change-Id: Ib208fc70c4f477495b8f44a155a85e9ba3c5c123
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Remove event sources not invoked when destroying webrtc handle

A crash can happen with the previous codes.

It is fixed by removing event sources of idle callbacks
which are not invoked yet before destroying webrtc handle.

[Version] 0.2.155
[Issue Type] Bug fix

Change-Id: Icff390fdd63ee2aa7bfeedd547d63dbb3e0f5d5a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_ini: Add new item to set bundle policy and apply it

[general]
; SDP bundle policy (0:none, 1:balanced, 2:max compat, 3:max bundle)
bundle policy = 3

Note that 1 and 2 are not supported yet.

[Version] 0.2.154
[Issue Type] Improvement

Change-Id: I47f72ad12d21399727a398ea74da7e452b5a71ec
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_ini: Revise two default values

1. DEFAULT_USE_ULPFEC_RED is changed to FALSE
 FEC is optional functionality that also affects bitrate and latency,
 so, set it FALSE as a default value.

2. DEFAULT_VPXENC_KEYFRAME_MAX_DIST is changed from 999999 to 10
 The previous value which has been brought from www.webmproject.org
 with no thought of that we are using 'wait-for-keyframe' of VP8
 depayloader. With a high value, received video stream can be shown
 to be freezed when a RTP packet gets lost. Therefore, this value is
 now changed to the reasonable value.

Note that these values affect only if there's empty value in ini file
or there's no ini file in a target.

[Version] 0.2.153
[Issue Type] Improvement

Change-Id: I19dd64b5bdb3fd21f09ab213cf2d8bae9b52a190
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Add excluded element list to CREATE_ELEMENT_FROM_REGISTRY()

The excluded element list from ini file is now referenced by two locations.
1. sink side (previous one) - _decodebin_autoplug_select_cb()
2. source side - CREATE_ELEMENT_FROM_REGISTRY() in __create_rest_of_elements()

[Version] 0.2.152
[Issue Type] Improvement

Change-Id: I277dfd574d8abbb9cd25e961a46efbaff7855ffc
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Show server ip/port/status when using private signaling server

It is also fixed to use designated initializers for some string arrays.

[Version] 0.2.151
[Issue Type] Improvement

Change-Id: I28f50fee799466de6ac0fee6de1772374a196246
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

fixup! Added vpx encoder system configure setting for real-time CBR encoding and streaming

Wrong comparisons are fixed. The previous patch is now affected.

Change-Id: If16b7a7be15ad1062b3736352bcdccd7071e8f15
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Fix typos

DEFAULT_NAME_XXX should be ELEMENT_NAME_XXX.

[Version] 0.2.150
[Issue Type] Typo fix

Change-Id: Id365803d8e92272aec050e9bccc1a4b2075bfa28
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Set channel and samplerate if available when making a media format

This code blocks is activated when user calls webrtc_set_encoded_audio_frame_cb().

[Version] 0.2.149
[Issue Type] Improvement

Change-Id: I663a3e3416beb2cf974f346c43bd0b750ae79737
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Use list to carry elements to remove these from file source

Variable and function are also renamed to use 'payloader' not 'payload'.

[Version] 0.2.148
[Issue Type] Refactoring

Change-Id: Iaf165625dc135d2e0248e3c103ffc0aa9775bcd4
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Merge branch 'tizen' into tizen_6.5

Change-Id: I2f807e574904dfe7ddbe10f8c671f07fe0e55ee7

webrtc_internal: Revise webrtc_screen_source_set/unset_crop()

Parameter check codes are revised.
g_mutex_locker_new() applies to this function.

[Version] 0.2.147
[Issue Type] Refactoring

Change-Id: I1ce481aa9e65a34a5b813258885019a0f7e6afa9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Use list to carry elements for making encoded media packet source

Some codes exiting without releasing resources are fixed.
Level of logs in __link_elements() is changed.
Redundant logs are removed.

[Version] 0.2.146
[Issue Type] Improvement

Change-Id: I3d315984ab5fd4d2046a546d823549a29ab4c7e2
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Improve list handling in __complete_mediapacketsrc_from_raw_format()

When an error occurs, a node memory of list for appsrc is not freed.
It is now fixed.

[Version] 0.2.145
[Issue Type] Bug fix

Change-Id: I092cfb4dc86570fca0f77e4c68171c5b9a228908
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Improve error handling when failure on adding element to bin

If an error occurs when calling gst_bin_add() for element list,
error handling codes for unreferencing the elements should be divided
with two phases, one is for elements already added to the bin, the other
one is for the rest of elements in the list.

[Version] 0.2.144
[Issue Type] Improvement

Change-Id: Ie5a8c9eaa0f8dd462ec0079a59e85ebc1b8f070a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Fix to add ice candidate to the valid handle

[Version] 0.2.143
[Issue Type] Bug fix

Change-Id: Ic3c8a754382b222d74143fd8062146449d20d842
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Append webrtc handle pointer address to webrtcbin name

This can help user analyze logs more easily.

[Version] 0.2.142
[Issue Type] Debug

Change-Id: I6afaa2d622f76cf57a63823dff4c80e6fd709b8f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Support data channel in case of room join test

It is now possible send/receive text message via data channel
in case of room join test.
'zs', 'zb' menu can be used to send message to peers in the room.

[Version] 0.2.141
[Issue Type] New feature

Change-Id: Ia4847e8a91e55023c789bd113cbc6e4c6d8e1813
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Fix bug of room join test

In case of room scenario, when a remote peer is joining the room
where the first handle has already joined, new webrtc handle uses
same media sources that the first handle is using.

[Version] 0.2.140
[Issue Type] Bug fix

Change-Id: Ifa9ecea9b01a35370d871a8277668cba4b64083c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Abandon connection change menu

The connection change menu intended to use multiple websocket
connections is not that useful considering the conflict of display
object. This application is now modified to use only one websocket
connection with signaling server.

The room joining scenario still can have multiple peers(webrtc handles)
with only one websocket connection.

[Version] 0.2.139
[Issue Type] Clean-up

Change-Id: I712665a9f1040e192706716e6fa6b4f75fb7599a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Show text message from data channel to the display

[Version] 0.2.138
[Issue Type] Improvement

Change-Id: I217484ed9bf9d74a1519dee8726afaf8f9b5c4d4
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Use list to carry elements for loopback pipeline

[Version] 0.2.137
[Issue Type] Refactoring

Change-Id: I35530c0bb7fdb89bdcdde5897d068a8acfaa599c

Apply macros to exclude lines from coverage measurement #2

[Version] 0.2.136
[Issue Type] Line coverage

Change-Id: I472ebce3a0748d0056ec929a920388262f1e1288
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Apply GENERATE_DOT() macro to loopback and filesrc

The filesrc pipeline name and loopback render pipeline name are
changed to be identified easily that which source belongs to it
by its name.

Dot file names are also changed.

[Version] 0.2.135
[Issue Type] Debug feature

Change-Id: I843eb7b4e8f42df20c5cd04a89c42a773e79af8c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Revise macro definitions

Some are modified to use do/while(0).

[Version] 0.2.134
[Issue Type] Improvement

Change-Id: Iae3b8aa519077fd03593651b3478deb7f2397e57
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Add parameter to __create_rest_of_elements() to check the exact media type.

file source can have audio and video together in the "media type".
So, actual media type to make proper elements should not be determined only by the "media type".

[Version] 0.2.133
[Issue Type] Improvement

Change-Id: Idc9ffa36d5ca01bdf59410be78dad8c57158e0d5

webrtc_private: Add explicit pipeline parameter to GENERATE_DOT macro

This will be used to print dot of other piplines.
Log level is changed to 'warning' in _generate_dot().

[Version] 0.2.132
[Issue Type] Debug feature

Change-Id: Iaa91b2d8ab77614f5b3d6f2dc0e729c79eb1b34a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Skip creating resource manager handle if any resources required in ini file

[Version] 0.2.131
[Issue Type] Improvement

Change-Id: I1648153301ef56f27c5d07f83d2b8a781baf81d2
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Skip making rendering sink during stopping or destroying handle

[Version] 0.2.130
[Issue Type] Improvement

Change-Id: I2a563aefd9734333cd9e54f9c25a3ef36e1eb2b0
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

define SAFE_G_LIST_FREE_FULL()

[Version] 0.2.129
[Issue Type] Refactoring

Change-Id: I05c1c0953d74427faa01b5be108439723f68d8af

webrtc_source: rearrange codes to reduce code complexity

[Version] 0.2.128
[Issue Type] Refactoring

Change-Id: Ie67fa8314d79461b51d8424ab7b3bca695ccfa24

webrtc_source: Enable file path change for the same source

[Version] 0.2.127
[Issue Type] Improvement

Change-Id: I3af2d27448915826bbd2becd1fade6e3c99dd14c

Apply macros to exclude lines from coverage measurement

[Version] 0.2.126
[Issue Type] Line coverage

Change-Id: I7ec0f6c1aaf05d23d393fb0d4fec098ad2b58599
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add gcov package for line coverage automation

[Version] 0.2.125
[Issue Type] Line coverage

Change-Id: I65bcf327b8735057aa2162f4f3f3e1c90d9e6b0b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Maintain received ICE candidates before destroying handle

Based on gstreamer webrtcbin, it does not gather ICE candidate again
when after gathering completed until unref the bin.
These ICE candidates can be used after webrtc_stop().

Missing * are added in menu for internal API.

[Version] 0.2.124
[Issue Type] Improvement

Change-Id: Ie43a3157315f5586bcbbec377cdab8b234c825c5
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_data_channel: Close data channel before destroying the handle

It'll trigger the close callback on the data channel.

[Version] 0.2.123
[Issue Type] Improvement

Change-Id: I578a70a3677652addd5aa9896bdf7323ee67988a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private/sink: Print handle pointer address

Some logs for webrtc handle and decodebin are added.

[Version] 0.2.122
[Issue Type] Log

Change-Id: Ib07f9216c482d3e73a38cf51069e8cabc0669c94
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc/webrtc_source: Print webrtc handle pointer address

[Version] 0.2.121
[Issue Type] Log

Change-Id: I37e8f900275a79a48d171f1104d4f64367ede6f7
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Disable clock synchronization of loopback pipeline audiosink

Webrtc handle can have a source that consists of audio, video or both
media types. Each type can have a loopback pipeline. So it is set to FALSE
to render the incoming data from pad probe callback as soon as possible.

[Version] 0.2.120
[Issue Type] Improvement

Change-Id: I3f46e96123031598a5d86fa8fc85c1ec96772e4a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: add queue after the decodebin in the filesrc pipeline

[Version] 0.2.119
[Issue Type] Improvement

Previously, a probe for loopback support was attached to decodebin's pad,
but decodebin deletes all pads when state goes to NULL.
The loopback setting must be maintained until the user unset it.
So, a queue was added after the decodebin and support loopback.

Change-Id: Ia1aaba9fa3b5b995161216294dcb49b6930c8624

webrtc_source: refactor audio/video branches using static mapping table

[Version] 0.2.118
[Issue Type] Refactoring

Change-Id: I5046a03cf070e54a8e7624b273219ac4099e0d3b

webrtc_source: rearrange codes to reduce code complexity

[Version] 0.2.117
[Issue Type] Refactoring

Change-Id: I31b43afb40ae1bd5a829dc1ccb99685bd4ead1a4

webrtc_source: Return error when loopback pipeline has already been set

[Version] 0.2.116
[Issue Type] Bug fix

Change-Id: Ie6b408cfb75cc7a827a94b068e8d0042a064cb3a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: rearrange codes of __create_rest_of_elements_for_filesrc_pipeline() to reduce code complexity

[Version] 0.2.115
[Issue Type] Refactoring

Change-Id: Ibed6a8d5b613e91af1a8a6c6f944e59dcdeae569

webrtc_source: Fix invalid use of capsfilter

[Version] 0.2.114
[Issue Type] Improvement

Change-Id: I98dc18b2139c767f6d917962d1e5d613a679d37b

Add API to set/get file source looping

Functions are added as below.
- webrtc_file_source_set_looping()
- webrtc_file_source_get_looping()

[Version] 0.2.113
[Issue type] API

Change-Id: Ie088db29ac4aeaf19fe2d5f85138787c4da5c9f7

Change the structure of file src

A separate pipeline for filesrc is added, and the existing src bin receives input with appsrc.
This makes functions such as file looping convenient by separately managing pipelines.

[Version] 0.2.112
[Issue Type] Improvement

Change-Id: I69e1edea62515eb57987624e12bf863fa653b3fc

webrtc_test: Apply -Wcast-function-type and fix the error

It is added to comply with VD COSMOS build configuration.

[Version] 0.2.111
[Issue Type] Improvement

Change-Id: I104e55c2520708641b6bf56daf7a9765a4f41c2e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_websocket: Fix missing field initializers

Apply -Wmissing-field-initializers and fix the errors.
It is added to comply with VD build configuration.

[Version] 0.2.110
[Issue Type] Improvement

Change-Id: Iab651b06a2d89e51c4c8f40c9fb5831d4038c8c6
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink/data_channel: Fix coverity issues (CHECKED_RETURN)

[Version] 0.2.109
[Issue Type] Coverity (CHECKED_RETURN)

Change-Id: I6911282dc9f226be4eafa59f485d2a5cc109a244
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Fix coverity issue of USE_AFTER_FREE

[Version] 0.2.108
[Issue Type] Coverity (USE_AFTER_FREE)

Change-Id: Ic67674d991bfe8de19e03058b77646248d634221
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Prepare for test case with esplusplayer to render data when using encoded frame callback

It'll be the default case to test encoded frame callback.
For now, it is excluded when TV profile build.

[Version] 0.2.107
[Issue Type] Improvement

Change-Id: I85bae50e99bd937daf2e53aa901a1ecd90a6de98
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc: Add missing precondition for negotiation callbacks

[Version] 0.2.106
[Issue Type] Doxygen

Change-Id: If95522da7cdd138ca5a4f1eb734bd9b4f0a7353d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Apply -Wsign-compare and fix the errors

It is added to comply with VD build configuration.

[Version] 0.2.105
[Issue Type] Improvement

Change-Id: Ia863063842cd95f23c6db3b320923ae182ef6945
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Apply -Wshadow and fix the errors

It is added to comply with VD build configuration.

[Version] 0.2.104
[Issue Type] Improvement

Change-Id: Id1481b5077723fd0ca74107fe1d618a0d5974c20
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_display: Revise logic for display mode/visible

Default mode and visible values are set when allocating display.
These values can be updated by setter APIs regardless of sink_element set.
Check properties for mode and visible before g_object_set().

[Version] 0.2.103
[Issue Type] Improvement

Change-Id: I9f98c764e73e23e7f6a87a167ba1211460b45360
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add menu to get data channel label

dl. Get data channel label

[Version] 0.2.102
[Issue Type] Add

Change-Id: I37a592bb42d3a6a854b8dca03c223b873b9e71ee
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Print * to represent the internal API

[Version] 0.2.101
[Issue Type] Revise

Change-Id: Ibdbe0c53d503dadf50f22180a714d51ae44b4419
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Use fakesink to drop receiving audio data if stream_info is not set

[Version] 0.2.100
[Issue Type] Improvement

Change-Id: I056d6b1b4fce1dfcb0a20a969631326ec7d7be7d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Use fakesink to drop receiving video data if display is not set

[Version] 0.2.99
[Issue Type] Improvement

Change-Id: I69296f2abe7792e5b6f2c468d5b1bbba58b4f860
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add new internal APIs for setting or unsetting crop screen source

- webrtc_screen_source_set_crop()
- webrtc_screen_source_unset_crop()
- Add function test to webrtc_test

[Version] 0.2.98
[Issue Type] New feature

Change-Id: Ib1f36d6b84ce3ff429ff0ae20879b50b6f5af011
Signed-off-by: Hyunil <hyunil46.park@samsung.com>

webrtc_doc: Update description

[Version] 0.2.97
[Issue Type] Document

Change-Id: I546caef069a8ba2f7319dabe72e021e4e7260dac
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Lock mutex of display in __build_videosink()

It is improved to guard display structure while accessing it.

[Version] 0.2.96
[Issue Type] Improvement

Change-Id: I61f86eaa3970e659b614d3bdb9cb9d589254b23b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_ini: Add support for printing stats log periodically

[general]
stats log period = 0

It is added to print statistics log periodically to check current
situation of data transmission without any user input.
In case of 0 sec, it does not print any stats logs.

[Version] 0.2.95
[Issue Type] Log

Change-Id: Ibc0b418d7c6544f995b5d78822d8df241c064f7c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

move webrtc_file_source_set_path() to internal

[Version] 0.2.94
[Issue Type] API

Change-Id: I8913184b8ab51d85f70fdcfda0aec0dc585645d8

fixup! webrtc_ini: Add new item to set libnice verbose log

Change-Id: Ife051178de6bc972a4925124c0270fae74c86c24
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_ini: Add new item to set libnice verbose log

[Version] 0.2.93
[Issue Type] Improvement

Change-Id: Ib173a0b87c0cf9aed322e158c302127b35682117

webrtc_test: Add menu for creating offer/answer asynchronously

[Version] 0.2.92
[Issue Type] New feature

Change-Id: I2b30064c4731d82c38786913ff0ddc0f866144f4
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add new asynchronous API to create offer/answer

Functions are added as below.
 - webrtc_create_offer_async()
 - webrtc_create_answer_async()

[Version] 0.2.91
[Issue Type] API

Change-Id: I5641f98fcd272ddd52f5173c048a9db3a94a9222
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add missing initializing variables after free()

It caused a double-free crash when negotiating again with new handle
even if the 'd'(destroy) menu was executed for the previous handle
without program exit.

[Version] 0.2.90
[Issue Type] Bug fix

Change-Id: I48df929d6744d434f23f7d550d692e92b0b61609
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Set omitted display->sink_element in case of OVERLAY display type

It will be used when setting a display mode/visible to the track id of
video loopback pipeline.

[Version] 0.2.89
[Issue Type] Bug fix

Change-Id: I03cc2b3807495d3851643fc804f03570ad2ebab8
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Use fixed payload id for particular codecs

This patch enables normal operation with that codecs between
web API and gstreamer webrtc at last.

[Version] 0.2.88
[Issue Type] Improvement

Change-Id: Ib4094ac5814d59632032f649a69e0b45bc5b4b1d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>