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] 0.4.59
[Issue Type] Condition added (for product)

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

test: Add test case for webrtc_set_bundle_policy_max_compat()

[Version] 0.4.58
[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] 0.4.57
[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] 0.4.56
[Issue Type] Bug fix

Change-Id: Idd0ef5467acf39ca667c5a6b41505858e12f06d2
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>

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>

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>

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>

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>

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>

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>

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>

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>

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>

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>

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>
(cherry picked from commit 23e7af22279728cc493ffe2b52aee841d84a2c3d)

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>
(cherry picked from commit 62899691a458f092fa1beedb5ac4b5b444e65abd)

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>
(cherry picked from commit 5a3887876bb22b17e1a6937ce727854bb768baea)

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>
(cherry picked from commit a6848ad00afe28fd38eee1388b24e415c74656c5)

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>
(cherry picked from commit ee22a7e6b61f25ea0509c3a8a129123c99ccfa9b)

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>
(cherry picked from commit 060828225281c1c9258ac205ea895f0724f1aa89)

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>
(cherry picked from commit 941cdb71d03ca95508806f629ee6938488f1cc6c)

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>
(cherry picked from commit ae3b41cc027386fa40c579bfd3767fa228b10549)

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>
(cherry picked from commit 410af450e62993aa1ad54bd56b51f3449b9292b0)

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>
(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] 0.4.32
[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>
(cherry picked from commit 9bd556786740b77796dd51008a642641a394827b)

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>
(cherry picked from commit e09fe0750e51a0147878b83fed3e1432c48eb9cf)

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>

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>

webrtc_private: Ensure to invoke error callback in main thread

[Version] 0.4.12
[Issue Type] Improvement

Change-Id: I3410d3baf9d113fd4936be6d5a93ce33839dc3e2
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source_camera: Support tizencamerasrc element

[Version] 0.4.11
[Issue Type] Improvement

Change-Id: Id01e931221c709a48af2e4ab02c1fbc826f24d6d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Apply new TIZEN_FEATURE_SIGNALING definition

[Version] 0.4.10
[Issue Type] Feature

Change-Id: Ic5d5e7be966883af356734cd627be3fabccd23b3
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Change the application of macro for coverage measurement exclusion

[Version] 0.4.9
[Issue Type] Line coverage

Change-Id: Iadbf2acef199da87181ee0a51a76cfd0194e2e82
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Apply new TIZEN_FEATURE_SNAPSHOT definition

[Version] 0.4.8
[Issue Type] Feature

Change-Id: Ic5c67f3066c5157f76303bd17e8c3fe1fb46ba3f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Apply new TIZEN_FEATURE_DNS definition

[Version] 0.4.7
[Issue Type] Feature

Change-Id: I63e85cddacd1cd56dc47cb6c73fa162a623dc07e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add webrtc_set_display_surface_id()

[Version] 0.4.6
[Issue Type] API

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

webrtc_test_espp: Remove unnecessary function call

Codes to call esplusplayer_activate() are removed.
Some variable names are changed.

[Version] 0.4.5
[Issue Type] Improvement

Change-Id: Id3d7f1c841b4e155a2a0cfa0e18033fc3c8d2bec
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Set channels or rate to media format even if only one has a valid value

More logs are added.

[Version] 0.4.4
[Issue Type] Improvement

Change-Id: I34d985ee78051dc82ff517920739b037c97714e8
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

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

[Version] 0.4.3
[Issue Type] Improvement

Change-Id: Icb5181afd2c759756187cd2a68ce4fa315c84c15

Add new error type - WEBRTC_ERROR_NETWORK_RESOURCE_FAILED

[Version] 0.4.2
[Issue type] API

Change-Id: I640059cf94bac87fba28ef4e5ff37cadc6f86680
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add new stats types

new enums are added as below.

for webrtc_stats_type_e
 : WEBRTC_STATS_TYPE_CANDIDATE_PAIR
 : WEBRTC_STATS_TYPE_LOCAL_CANDIDATE
 : WEBRTC_STATS_TYPE_REMOTE_CANDIDATE

for webrtc_stats_prop_e
 : WEBRTC_STATS_PROP_KIND
 : WEBRTC_STATS_PROP_ADDRESS
 : WEBRTC_STATS_PROP_PORT
 : WEBRTC_STATS_PROP_CANDIDATE_TYPE
 : WEBRTC_STATS_PROP_PRIORITY
 : WEBRTC_STATS_PROP_PROTOCOL
 : WEBRTC_STATS_PROP_RELAY_PROTOCOL
 : WEBRTC_STATS_PROP_URL
 : WEBRTC_STATS_PROP_LOCAL_CANDIDATE_ID
 : WEBRTC_STATS_PROP_REMOTE_CANDIDATE_ID

Some enum values are changed.
Some definitions are deprecated.

[Version] 0.4.1
[Issue type] New feature

Change-Id: I021af299a1b3ddb048f29a3888b5dc86d8191ab8

Use g_autoptr for JsonParser variable

[Version] 0.3.289
[Issue type] Refactoring

Change-Id: Iaf7fac85e87150abdfae106827fb84b7c41d001e

webrtc_private: Print debug message in GstMessage object

[Version] 0.3.288
[Issue type] Logs

Change-Id: Ied0efd7e522cddbf3b1f291aec24b78deb91ace7

Use GST_TIME_FORMAT for buffer timestamps

[Version] 0.3.287
[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] 0.3.286
[Issue type] New feature

Change-Id: If66b7936166fdb9412f748f38a5374fd472df677

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

[Version] 0.3.285
[Issue type] Refactoring

Change-Id: If7e8dabfc8fb87f4a93259b81a37f47f3b01608b

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_stats: Add null check code before calling gst_structure_foreach()

It is to avoid printing warning message from a console.

[Version] 0.3.283
[Issue type] Improvement

Change-Id: I52f0c9ad3f7c218b5af9d70706625a1e7848fcbb

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>

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>

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>
(cherry picked from commit dd6a0fcb0483d6c380c16aab597d5b88d6d8a9f0)

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>

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>

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>

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>

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