Print version log when creating handle

[Version] 1.1.42
[Issue Type] Log

Change-Id: I94647b31517bcf0e53f8977c1cc684dc8e5ac4f9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Improve codes regarding setting element properties from ini

Some have been changed to comply with the previous style of
saving key value pair from ini.

[Version] 1.1.41
[Issue Type] Improvement

Change-Id: I3394c556d5e493237e7888c2b4cf625e57091c8e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

TCT fail fix : add null pointer reference exception

Change-Id: I9708e67c00ce839d17300426357bfe197b8d7295

add ini configs for sink, decoder property

Change-Id: Iaa04ae6f149af68361c41a2cc1f7a034ec7f4541

don't release tbm surface if h/w codec case, add STV0

Change-Id: Ief66d638f32f95fd115934bbce32cbc44797d8cf

support h/w codec

Change-Id: I9f84ec0f744500f0a747ebfa57e33341bc193555

webrtc_test: Add 'quick start' test case

qs. Quick start - 2 peers (AV sender - AV receiver)

[Version] 1.1.40
[Issue Type] Test application

Change-Id: Ic83d16b3aa36b102e4bda8cb8701e3bc1d12618f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Add defensive code regarding destroying handle

[Version] 1.1.39
[Issue Type] ASAN defect

Change-Id: Iaf57f54224e91b18c40ba1055c6e5c7cb35b7536
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

fixup! webrtc_private: Check and release remaining transceivers when destroying handle

The logic is moved to _source_slot_destroy_cb() in webrtc_source.c.
NULL source type is excluded to apply these unreffing codes to avoid
'head-use-after-free' issue on ASAN test when webrtc_destroy().

[Version] 1.1.38
[Issue Type] Bug fix

Change-Id: I6d2d8b7bcd80631c13ca952c570c057f0dbe8534
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Check and release remaining transceivers when destroying handle

These unreferencing objects will finally lead to call nice_stream_finalize()
of libnice via gstwebrtcbin.

[Version] 1.1.37
[Issue Type] Resource leak

Change-Id: Iec14856967c0c14670784e48d7a82c5b9b480702
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source_loopback: Check the return value of gst_bin_add()

[Version] 1.1.36
[Issue Type] Coverity defect

Change-Id: I220c49e058ee0ac90da7e499bb701f397005b615
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Change of mutex used for unintended purposes

desc_mutex/cond was made to protect description.
but signaling_state_cb has used desc_mutex/cond to ensure signaling state change
before returning of _webrtcbin_set_session_description()
(from e73ac03c4aa1e8a144d0ced1d12af43685d3b712)

User want to set_local_description() in offer_created_cb because that is good way in the flow.
but, offer_created_cb uses desc_mutex/cond and _webrtcbin_set_session_description() uses too.
that caused deadlock, so we divide mutex/cond.

[Version] 1.1.35
[Issue Type] Bug fix

Change-Id: Ia2a665b70818889e275308dfe94ad0019ec62312
Signed-off-by: Youngwoo Cho <young222.cho@samsung.com>

webrtc_source_simulcast: Add missing gst_object_unref()

It should be done because of the increased ref. count
by calling gst_element_get_parent().

[Version] 1.1.34
[Issue Type] Improvement

Change-Id: Ic75792686402c547818f34a97bd36e767a0b025f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Check the return value of gst_bin_add()

[Version] 1.1.33
[Issue Type] Coverity defects

Change-Id: I53b005b655caad3543b693e7efef5b83baa90962
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Check media_type parameter

[Version] 1.1.32
[Issue Type] Coverity defects

Change-Id: Ia96b2cf3f5f316204341bad1839bf6e3f76a10a7
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Unlock locker when error occurs during webrtc_create()

[Version] 1.1.31
[Issue Type] ASAN defect

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

Check pend state in webrtc_start_media_source()

The source must be constructed even if it is called right after
calling webrtc_start() which only changes the pend state inside
of it.

[Version] 1.1.30
[Issue Type] Bug fix

Change-Id: I612b07594ac6d4eb94dfddbe484e94204f31156e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Improve _set_manual_payload_type()

A condition has been added not to allow 0 value.
Valid range has been added to the documentation.

[Version] 1.1.29
[Issue Type] Bug fix

Change-Id: Ieb33ed50d8175c67770668c1a7e95a798845bddd
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Improve documentation

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

Missing # character for enum value has been added.

[Version] 1.1.28
[Issue Type] Doxygen

Change-Id: I2c01e891fdb245ba9706d6e36a032cadd03e70ca
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add examples for webrtc_get_local[remote]_description()

[Version] 1.1.27
[Issue Type] Doxygen

Change-Id: Ib7e84f979fe847333a349a48ce307088f299fb97
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

fixup! Enable resource manager commonization

Even if it does not need to register resource handle according to
the webrtc configuration, it must not emit the error in the state
of release.

Change-Id: I82798b7f84850f13d5d8b824fab517ed100e324a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_data_channel: Fix use-after-free issue

[Version] 1.1.26
[Issue Type] ASAN defect

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

Fix doxygen

To generate html link properly for @ref section_name + . + \n,
At this case (ends with .\n), it does not work with doxygen 1.11.0
Needs to add a space before .

[Version] 1.1.25
[Issue Type] Doxygen

Change-Id: I3d4cdc657d03529aeb13ba989cc45d2d947737fe
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add webrtc_media_source_set_payload_type() and webrtc_media_source_get_payload_type()

[Version] 1.1.24
[Issue Type] API

Change-Id: I9d28bff26198e2239783e4f28467e8b98a124876
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add webrtc_start_sync()

[Version] 1.1.23
[Issue Type] API

Change-Id: I1093273002c4774835cdf067d89ef25ae138a3c3
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add webrtc_get_local_description() and webrtc_get_remote_description()

[Version] 1.1.22
[Issue Type] API

Change-Id: Ifee7174a06158b57d8dd0b0883c68bf9bf050300
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add WEBRTC_BUNDLE_POLICY_MAX_COMPAT enum

It has been moved from internal API.
: webrtc_set_bundle_policy_max_compat() has been removed
 from internal header.

[Version] 1.1.21
[Issue Type] API

Change-Id: I0ffbf07a0ac62c20ab39cf1e493ae03b45d97250
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Enable resource manager commonization

[Version] 1.1.20
[Issue Type] Update

Change-Id: I659b3cc0c87c12c75c39ffaf0a6852e680b56f47

webrtc_transceiver: Check return value

[Version] 1.1.19
[Issue Type] Coverity defect (UNUSED_VALUE)

Change-Id: I33376dd5999ce087abf58997fbff04ea8d39377e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Add condition to return payload type when destroying source

It is to avoid printing error log in some cases.

[Version] 1.1.18
[Issue Type] Improvement

Change-Id: Idcfcbe670afb1530bdb3d8cabca3f2cfce6d8e9d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add 'ssrc' outparam to webrtc_media_source_add_transceiver_encoding()

[Version] 1.1.17
[Issue Type] API

Change-Id: I5b2be20c9e2c635edc9612fe64671c22f68eb73d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source_mediapacket: Revise the timing of invoking the nego. needed cb

Previously, in case of h264 format, pushing a h264 buffer is required to
invoke webrtcbin's negotiation needed callback. With setting the particular
property of rtph264pay, it'll be invoked before pushing the first buffer.

[Version] 1.1.16
[Issue Type] Improvement

Change-Id: I9fb47ded651886a9a3dfb23b68b57b787830ca2f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Revise webrtc_media_source_set[get]_payload_type()

Logic has been improved for multiple codecs of null source.
Now, payload type can be set to each codec of the null source.

[Version] 1.1.15
[Issue Type] Improvement

Change-Id: I5864d7f9bfd5e937e407f7b24d0950958bc6a826
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

test: Add VP8 format for webrtc_media_packet_source_set_format() test

[Version] 1.1.14
[Issue Type] Test application

Change-Id: I46e45afdc389c5e13d2a3a4c4c12f8cc971ad89d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source_mediapacket: Set picture-id-mode in case of VP8 payloader

It is applied similarly with the commit below.
: 17e92a5dcf40ad3a82cde8fa3c1cb41652f2ccb3

_update_payloader_vp8_picture_id_mode() is extracted from the
previous codes.

[Version] 1.1.13
[Issue Type] Improvement

Change-Id: I6e390f72f0664ee709d14f9db2843d41b33b6542
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Remove unnecessary braces

[Version] 1.1.12
[Issue Type] Convention

Change-Id: Ia2a925d7101da5ddf5349ca707bd99a1a2d4fe91
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

test: Add test case for webrtc_null_source_set_media_type()

[Version] 1.1.11
[Issue Type] Test application

Change-Id: I2cb459eadb42073d83d8cbc74a30a0f8f2207d79
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add webrtc_null_source_set_media_type()

If this function is called to the particular null source(recvonly),
then all of available codecs which defined in ini file are set to
the transceiver. Therefore offer or answer description will represents
various codec information via rtpmap attributes.

[Version] 1.1.10
[Issue Type] Internal API

Change-Id: Ifba9314b2eb769ae67a5dbdd8d9aab402ce28257
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

fixup! webrtc_internal: Revise parameter of webrtc_media_source_set[get]_payload_type()

A compile error on 64-bit has been fixed.

Change-Id: I23615ded2b4d3d1319f18b63a20ec6586a84e517

webrtc_internal: Revise parameter of webrtc_media_source_set[get]_payload_type()

webrtc_test has been fixed to comply with the changes.

[Version] 1.1.9
[Issue Type] Internal API

Change-Id: I530e8210aa838e6dd4651cf0dc0b5a5631117ff1
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Apply more macro to exclude lines from coverage measurement

[Version] 1.1.8
[Issue Type] Coverage

Change-Id: I938ec05d398ae463edfe668413c82ef87b2f5ac7
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.
Plus, ortc of mediasoup requires these fields as well.

[Version] 1.1.7
[Issue Type] Improvement

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

test: Add test case for webrtc_set_bundle_policy_max_compat()

[Version] 1.1.6
[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] 1.1.5
[Issue Type] Internal API

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

webrtc_source_private: Set picture-id-mode in case of VP8 payloader

7-bit picture id has been set to vp8 payloader.
It has also been verified with Chrome browser.

[Version] 1.1.4
[Issue Type] Compatibility

Change-Id: Ib05694631fbb1424ba0c21194f103e94b9484f24
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Make sure to get ssrc before returning webrtc_start_media_source()

It is modified to return the function after invoking
__webrtcbin_on_negotiation_needed_cb() to get ssrc properly when
the following funcion is called - webrtc_create_offer().

[Version] 1.1.3
[Issue Type] Bug fix

Change-Id: Ic1b7abc76a2fb932978f17ea9af1392628937361
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] 1.1.2
[Issue Type] Bug fix

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

test: Add menu for webrtc_start_media_source()

[Version] 1.1.1
[Issue Type] Test application

Change-Id: Ia6b07b0affa309e732d9b86fb8a989465ffcbb7d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add webrtc_start_media_source() function

This function will be used after adding a media source during state
of #WEBRTC_STATE_NEGOTIATING or #WEBRTC_STATE_PLAYING.
It has been added for re-negotiation scenario.

[Version] 1.1.0
[Issue Type] API

Change-Id: I0985473ce0c37767c8734072104f9b34132475a5
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.

[Version] 1.0.5
[Issue Type] Improvement

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

Change state constraint on webrtc_foreach_stats()

It has been changed to allow WEBRTC_STATE_NEGOTIATING state
to get some stats information in this state later.

[Version] 1.0.4
[Issue Type] Release state constraint

Change-Id: Iafc094f695badd4d5c823d70b59cbf400c33cc1e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Remove state constraint on functions for configuration

Functions below are revised.
 : webrtc_set_stun_server()
 : webrtc_add_turn_server()
 : webrtc_set_bundle_policy()
 : webrtc_set_ice_transport_policy()

[Version] 1.0.3
[Issue Type] Release state constraint

Change-Id: I8235391d57ec42d2886aeba2fcbdc511f7a7628d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Remove state constraint on webrtc_create_data_channel()

[Version] 1.0.2
[Issue Type] Release state constraint

Change-Id: I438efbfa0225d16736cd6801d16d51f50efffe87
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Remove state constraint on functions for media source

It allows for easing the state constraints for below functions.
 : webrtc_add_media_source()
 : webrtc_remove_media_source()
 : webrtc_media_source_set_transceiver_direction()
 : webrtc_media_source_set_transceiver_codec()
 : webrtc_mic_source_set_sound_stream_info()
 : webrtc_camera_source_set_device_id()
 : webrtc_media_packet_source_set_format()
 : webrtc_file_source_set_path()

Below are internal functions.
 : webrtc_add_media_source_internal()
 : webrtc_media_source_set_payload_type()

[Version] 1.0.1
[Issue Type] Release state constraints

Change-Id: Idbad43285d35ccf3842e91ce31683ff748864a62
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Change state constraint on functions for negotiation

It allows for easing the state constraints for below functions.
 : webrtc_create_offer[_async]()
 : webrtc_create_answer[_async]()
 : webrtc_set_local_description()
 : webrtc_set_remote_description()

[Version] 1.0.0
[Issue Type] Release state constraints

Change-Id: Ia7a7db4a7ac4648e4fd7cb1bf46ff079710b727b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

test: Add menu for setting transceiver receiving data drop

Two items have been added as below.

srd. Set transceiver recv drop
grd. Get transceiver recv drop

[Version] 0.4.55
[Issue Type] Test application

Change-Id: I1f42282761c8a786ec26a3fa7b62062584a2911b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit e895b88fff02c8355943d13ba4c17eae6f11e18d)

webrtc_internal: Add functions to set/get drop receiving data

Funtions are added as below.
 : webrtc_media_source_set_transceiver_recv_drop()
 : webrtc_media_source_get_transceiver_recv_drop()

Receiving packets from a remote peer are related to a specific transceiver
which was created by media source in this API.
To drop receiving packets without decoding, these new functions are
added.

[Version] 0.4.54
[Issue Type] Internal API

Change-Id: I414bea2fda91a08170928452ecc457f5d29b3962
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit b7686cd3ed6228aa6d7783e7d63f58b81cd3c438)

webrtc_private: Add probe callback to the newly added source pad of webrtcbin

_add_probe_to_pad_for_render() and _remove_probe_to_pad_for_render() are moved
from webrtc_source_loopback.c.

It has been fixed to use gst_object_ref() before assigning it to slot's variable.

This probe callback will be used to decide to render the RTP data or not.

[Version] 0.4.53
[Issue Type] Improvement

Change-Id: I9b539c842e754e4d6d93cc9f95f28f1b79184f00
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit ba84ea6e3b92131c67206f7545f328c8915d9066)

webrtc_private: Ensure signaling state change before returning of _webrtcbin_set_session_description()

Though posting to invoke the signaling state changed callback in idle
still exists, now it is possible to call webrtc_create_answer() without
checking the signaling state changed callback right after calling
webrtc_set_remote_description().

[Version] 0.4.52
[Issue Type] Improvement

Change-Id: If03504c56ec44654f729ec5ea4c2cd96f4c3547f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit d50e506fdff1700698991d00373b57eb30b56f88)

webrtc_transceiver: Revise the postfix number of mid

Considering the postfix value of gstreamer element name accumulates
even if the handle is destroyed and re-created in the same process,
it has been revised.

[Version] 0.4.51
[Issue Type] Bug fix

Change-Id: Ia513a40637230fcf158f7f8a8e584af104c9bd6e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit fdca0f93759f51cd204f3ac73aaf49b1d18a7776)

test: Add menu for getting transceiver mid

[Version] 0.4.50
[Issue Type] Test application

Change-Id: I66e0c8824a0f7522cfeaaa5c6418988a584356e5
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit 9ca9c8cb3b25b708f6876ce7e5d9239d18913827)

webrtc_internal: Add function to create session description from sdp string

webrtc_util_create_description() has been added.

[Version] 0.4.49
[Issue Type] Internal API

Change-Id: I3718e504f37c165a9bc5d39c83ec976f2a111bde
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit 63a0c7ac8e865bdcbf2affc68dd9bb50e0788685)

Revise to check the return value of json_parser_new()

Error log for this situation is also added.

[Version] 0.4.48
[Issue Type] Improvement

Change-Id: I6035f5dd3ce04029b39f601a298c942ec97b62f4
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit d4070245a3b7d0563dee0349f8d6e7da4c30e5fb)

webrtc_internal: Add function to strip session description

webrtc_util_strip_description() has been added to only get the
string member of sdp object in the original description.

[Version] 0.4.47
[Issue Type] Internal API

Change-Id: Ib8c81c26a4924680c8d042fd73027badb7c7c201
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit 77b8fceb46bfdf0d9062f5d29869297adabf8ee7)

webrtc_internal: Add webrtc_start_sync() function

The main difference with webrtc_start() is that this returns
after changing state to WEBRTC_STATE_NEGOTIATING without
invoking state changed callback.

[Version] 0.4.46
[Issue Type] Internal API

Change-Id: Ia4d295de5ec861a7b7d20deed35c6c764a0e01d7
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit ef0c8ad1e94441a702d67dc1671191b6f7e9fc8a)

webrtc_internal: Add function to get transceiver mid

webrtc_media_source_get_transceiver_mid() has been added.

[Version] 0.4.45
[Issue Type] Internal API

Change-Id: Ia69250ac27b46c68e01bbae125ebbbd103d874da
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit de4b925b47d7bea80afa945778e463965a95e32f)

webrtc_internal: Add function to get media type by source id

webrtc_media_source_get_type() has been added.

[Version] 0.4.44
[Issue Type] Internal API

Change-Id: I8659fcb85957ae353cf3e774d34477f8ebea7388
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add functions to get local/remote description

Functions have been added as below.
 - webrtc_get_local_description()
 - webrtc_get_remote_description()

[Version] 0.4.43
[Issue Type] Internal API

Change-Id: Ifb481b4b005a40145e959d31e3d2ccc3bb6c23f1
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

test: Add menu for activating transceiver encoding

[Version] 0.4.42
[Issue Type] Test application

Change-Id: If83d2eac958ba87d8a24d53862297c224e7ee734
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add API to active transceiver encoding option

webrtc_media_source_active_transceiver_encoding() has been added.

This function is to activate or deactivate each rid based stream.

[Version] 0.4.41
[Issue Type] Internal API

Change-Id: I65b4edcb7b171ad7c17b97153047888039a54e9f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_transceiver: Apply simulcast info to null source type

In case of answerer, null source type(recvonly) is used to select
codec information for receiving suggested streams from offer description.
If the offer description suggests RID based simulcast, this patch enables
to choose some of RIDs in the answer description to receive them.

[Version] 0.4.40
[Issue Type] New feature

Change-Id: Ifb036274f2e7c94c5be94d3f782e0edc16b1ab5a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

test: Add menu for adding/removing transceiver encoding

[Version] 0.4.39
[Issue Type] Test application

Change-Id: I3a89ca3ae6828f063ab564bd8a3931879e572fdb
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add APIs to add/remove transceiver encoding option

Two functions are added.
 : webrtc_media_source_add_transceiver_encoding()
 : webrtc_media_source_remove_transceiver_encoding()

Multiple source bin for rid-based simulcast could be added to
a particular media source. Each source bin has its own encoding
option and a rtp payloader with a specific ssrc.
RTP header extensions are also added according to definitions
from gstreamer.

[Version] 0.4.38
[Issue Type] Internal API

Change-Id: Iebc1fd223d81b04a7eb47a7a5d7181277737f7ad
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink_dump: Check return value of g_strdup_printf()

[Version] 0.4.37
[Issue Type] Svace defect (DEREF_OF_NULL.RET.ALLOC)

Change-Id: I30166ee0e3661a9e3aab51cf9f845471ff1d15a4
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Apply ASSERT() to the result of g_hash_table_insert()

[Version] 0.4.36
[Issue Type] Refactoring

Change-Id: Ia298d0697c6cc06c2417d686ebccc1b497b0d5fa
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_tbm/stats: Introduce ASSERT() macro function

Note that this assertion would be helpful not only during early
of the development phase to check unexpected wrong codes but
also during the runtime with being attributed to the compiler
optimizations.

Enough awareness of parameters whether if they are inevitable
or not would be prerequisite before applying it.

[Version] 0.4.35
[Issue Type] Debug

Change-Id: I29834774100c613a0f2154b589b8142342be40d8
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

test: Add missing lock

[Version] 0.4.34
[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] 0.4.33
[Issue Type] Improvement

Change-Id: I9eaf492456d24977d300d0682bcbbf60acd0f9ef
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

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] 0.4.32
[Issue Type] Bug fix

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

webrtc_transceiver: Get mid from transceiver name

This mid value will be set later to rtp header extension
for simulcast preparation.

[Version] 0.4.31
[Issue Type] Improvement

Change-Id: Ie85ddca278eb6a179e1753e5618f010a140ef8e2
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Adjust order of returning error values

The order of errors for particular functions were not in compliance with
the following sequence.
 - NOT_SUPPORTED > PERMISSION_DENIED > INVALID_PARAMETER

[Version] 0.4.30
[Issue Type] Improvement

Change-Id: I816ce028bc31abb601871538ea57db96f1ee4110
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add locking mutex for state

[Version] 0.4.29
[Issue Type] Coverity defect (MISSING_LOCK)

Change-Id: If29613750c55c1179a3393de65f52866be4997b8
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add new param of postfix number to CREATE_ELEMENT_FROM_REGISTRY macro function

-1 value will affect nothing.
Otherwise, the element name will be followed by the postfix number.

[Version] 0.4.28
[Issue Type] Improvement

Change-Id: I1529255b78b153deb61cd0f755f32cf7871fe9f9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Expand parameter to set ssrc later when making a rtp caps

If set it 0, it'll be set to a random number inside of gst plugin as it is.

[Version] 0.4.27
[Issue Type] Improvement

Change-Id: I2259ac28270709480c967fe1e40e3ef9551517dc
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] 0.4.26
[Issue Type] Testsuite

Change-Id: Idd081077d97a6bab538d49a0f5a3c282079224b9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>
(cherry picked from commit e3dd9012c7ec6781b59b989fc3616819df17c3e0)

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] 0.4.25
[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] 0.4.24
[Issue Type] Clean up

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

webrtc_internal: Add webrtc_set_display_qos()

[Version] 0.4.23
[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] 0.4.22
[Issue Type] Bug fix

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

Fix SVACE defect (DEREF_OF_NULL.RET.STAT)

[Version] 0.4.21
[Issue Type] SVACE (VD)

Change-Id: Ie0d31d03e2d63100b22f7a37bedaead826d7fede
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

test: Use g_file_set_contents() to dump file

[Version] 0.4.20
[Issue Type] Refactoring

Change-Id: I00411336b80159693394c23089f6bfeab74a8fba
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Revise @since_tizen 7.5 to 8.0

[Version] 0.4.19
[Issue Type] Doxygen

Change-Id: Iba1131b33364afd3277c7297a38336d8777d09e5
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Fix SVACE defect (DEREF_OF_NULL.EX)

[Version] 0.4.18
[Issue Type] SVACE (VD)

Change-Id: I3af6a032af8ce2a0d7ed91c6529d74511c8c562b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Fix SVACE defects (SIGN_EXTENSION)

[Version] 0.4.17
[Issue Type] SVACE

Change-Id: Ie8e5b4dadf738d4a17323e29ac0712fb6fd6ce4f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Change to use effect_method API

AEC APIs in sound-manager have been changed.

[Version] 0.4.16
[Issue Type] Feature

Change-Id: Idd9a9988b80aa47e72639e14724d40ce0a4406b7
Signed-off-by: Jaechul Lee <jcsing.lee@samsung.com>

Apply display feature

Some functions are not supported on the binary for the headed device.
Without this feature, these will return NOT SUPPORTED error.

[Version] 0.4.15
[Issue Type] Feature

Change-Id: I8344c964244f79f78e8864c1f18ae815d903a9f6
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Export webrtc_set[get]_audio_mute() API to public header

[Issue Type] API

Change-Id: I64ac88a28c153ddc15661ca4f71ea596a3b48f0c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Fixed the build error using gcc 13

Change-Id: I18c2d92099b8b5762cec3cf03af717e5c9d7ef31

webrtc_private: Fix build break

PA_PROP_MEDIA_ECHO_CANCEL_METHOD was removed in pulseaudio proplist.h.
So, new property named 'PA_PROP_MEDIA_PREPROCESSOR_METHOD' should be
used.

[Version] 0.4.14
[Issue Type] Build break

Change-Id: Ia6a5d0d221dd360aa73c5fc6ed4f4d36b8315c41
Signed-off-by: Jaechul Lee <jcsing.lee@samsung.com>

Change tizen version 7.5 to 8.0

It is related to ACR-1750.

[Version] 0.4.13
[Issue Type] Doxygen

Change-Id: I47c1cbd0e2ca9c0bc8db710db8f44e542fe57746
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>