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] 0.3.100
[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] 0.3.99
[Issue Type] Bug fix

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

webrtc_test: Use hashmap to interpret command

[Version] 0.3.98
[Issue Type] Refactoring

Change-Id: I7b1f4455a32134238c542f82a25daa711ebcc570
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Refactor audio/video mute functions

__validate_audio_source_for_mute() and __validate_video_source_for_mute()
are added to reduce duplicate codes.

Null parameter checks are added.

[Version] 0.3.97
[Issue Type] Refactoring

Change-Id: I7bf5438d7da93f6b5b2727b537822b3189950879
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Check return value of g_io_channel_read_chars()

[Version] 0.3.96
[Issue Type] Coverity defects

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

webrtc_source: Rename functions and replace codes with the function

[Version] 0.3.95
[Issue Type] Refactoring

Change-Id: I517b4ade896132e5a25f5a91f0ad422ae7ca9abd
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Add __complete_rest_of_mediapacketsrc() to reduce duplicate codes

Unnecessary element_list2 is also removed.

[Version] 0.3.94
[Issue Type] Refactoring

Change-Id: I7d9845c730f5236490e9ff66d1968954c9ee97db
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Reduce duplicate codes in __build_camerasrc/videosrc/custom_videosrc()

New sub function is introduced.

[Version] 0.3.93
[Issue Type] Refactoring

Change-Id: Ia689db8ffb5c4801492b3ef18df807041d6d3e4e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Print handle pointer in __bus_watch_cb()

A case using multiple handles in one process is quite common,
it is expected that this additional log will help in debugging.

[Version] 0.3.92
[Issue Type] Log

Change-Id: I1bca4152e8bce955d7a0e548b674e627771b840f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Reduce duplicate codes in __build_audiosrc()/__build_custom_audiosrc()

New sub function is introduced.

[Version] 0.3.91
[Issue Type] Refactoring

Change-Id: I0e0a6f678c710912c864b9652e10dc57689efaf3
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_stats: Fix invalid bitwise value of stats type

Type selection is added to the test case for
webrtc_foreach_stats().

[Version] 0.3.90
[Issue Type] Bug fix

Change-Id: I2be8cac40571f71d79c595909a40d91165f43984
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Use new sub functions in interpret()

Invalid bitwise value of TEST_MENU_APP_SIGNALING is fixed.

[Version] 0.3.89
[Issue Type] Refactoring

Change-Id: Ieb9a43eef5d5cae1ce1a41879c998bad8d99172e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Use new sub functions in displaymenu()

[Version] 0.3.88
[Issue Type] Refactoring

Change-Id: Id671ca12add86cea8b32b38686ecccc6a1fb4a6a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Rearrange menu items

Some menu items to set/unset each callback are removed.
 - "sac" or "uac" can be used instead of these.
Each menu status enum value has type bits.
Function and enum names regarding file source are changed.

[Version] 0.3.87
[Issue Type] Cleanup/Refactoring

Change-Id: Ibfe569120d531bc2e1078f9db2c836fe73ed6b75
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add missing static keyword to functions

Some function names are changed.
Unused function is removed.

[Version] 0.3.86
[Issue Type] Refactoring

Change-Id: I415dc7239837a13a80c537c78d0c097910da4f63
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Make sub functions to change menu state

[Version] 0.3.85
[Issue Type] Refactoring

Change-Id: I91462e96aad3011aca9a9545cca5a08725e2c9f5
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_stats: Update description as per the GStreamer's update

[Version] 0.3.84
[Issue Type] Documentation

Change-Id: I4ad0c0acd8bb3004a8ef860c9e448403b0af3b2d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_data_channel: Release data channel after receiving close callback

When destroying a data channel created by local peer, close callback could
be invoked in the middle of the process. Due to the early disconnection
signals, it'd never happen properly.

Add 'from_remote' variable to check if it is created by
_webrtcbin_on_data_channel_cb(). This kind of data channel can not be
destroyed by webrtc_destroy_data_channel().

A FIXME comment is added in _webrtcbin_on_data_channel_cb().

[Version] 0.3.83
[Issue Type] Improvement

Change-Id: Ic0bf5b3efc0760fe3221888cde038d5b1b4000fd
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_data_channel: Fix memory leak and double free

g_object_unref() is added for data channel object.

When webrtc_destroy() is called, data channels appended to the data channel
list are also released. Due to the omitted code to remove one from the list
when calling webrtc_destroy_data_channel(), double free can occur.

The above are fixed now.

[Version] 0.3.82
[Issue Type] Bug fix

Change-Id: I2b2942666c8ab992bb2a7fe21fc9546b5bdd3019
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_data_channel: Add sub-function to prepare data channel

[Version] 0.3.81
[Issue Type] Refactoring

Change-Id: Idae4533ee3b790e43daaa60a26999364bdbe791f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Fix spacing

[Version] 0.3.80
[Issue Type] Coding convention

Change-Id: Idbb43d9e817afe715c0ca1d9c956171c262d61ed
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] 0.3.79
[Issue Type] Doxygen

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

Add missing required libraries for pkg config

[Version] 0.3.78
[Issue Type] pkg-config

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

webrtc_test: Print stats type as string

[Version] 0.3.77
[Issue Type] Log

Change-Id: I7126c2da4ec511a10abe2f7d8dd71afb62d74d45
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_doc: Add callback operation description of the data channel module

[Version] 0.3.76
[Issue Type] Documentation

Change-Id: I8d82b12c99c5830fc214236dc646caa2be790a2a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_doc: Add description for statistics module

[Version] 0.3.75
[Issue Type] Documentation

Change-Id: I4a616a34500b5227ba33016a04b39ef950b9ca87
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add new statistics type for 'remote-outbound-rtp'

New statistics type is added as below.
 - WEBRTC_STATS_TYPE_REMOTE_OUTBOUND_RTP

Property enum is added as below for this type
 - WEBRTC_STATS_PROP_REMOTE_TIMESTAMP

Example codes are also added to the doxygen of
webrtc_foreach_stats().

[Version] 0.3.74
[Issue Type] API

Change-Id: I871069caf3dfd9591feff497f0e013a63995f7a9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add new statistics type for 'remote-inbound-rtp'

New statistics type is added as below.
 - WEBRTC_STATS_TYPE_REMOTE_INBOUND_RTP

Property enums are added as below for this type
 - WEBRTC_STATS_PROP_LOCAL_ID
 - WEBRTC_STATS_PROP_ROUND_TRIP_TIME
 - WEBRTC_STATS_PROP_FRACTION_LOST

[Version] 0.3.73
[Issue Type] API

Change-Id: I9547674d2ca3e5a083bd649ae78880efaf2f1d5a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add new statistics type for 'outbound-rtp'

New statistics type is added as below.
 - WEBRTC_STATS_TYPE_OUTBOUND_RTP

Property enums are added as below for this type
 - WEBRTC_STATS_PROP_BYTES_SENT
 - WEBRTC_STATS_PROP_PACKETS_SENT

[Version] 0.3.72
[Issue Type] API

Change-Id: Ia8574ddec63893eebacbd413ef253bed8b4a5102
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add new statistics type for 'inbound-rtp'

New statistics type is added as below.
 - WEBRTC_STATS_TYPE_INBOUND_RTP

Property enums are added as below for this type
 - WEBRTC_STATS_PROP_SSRC
 - WEBRTC_STATS_PROP_TRANSPORT_ID
 - WEBRTC_STATS_PROP_CODEC_ID
 - WEBRTC_STATS_PROP_PACKETS_RECEIVED
 - WEBRTC_STATS_PROP_PACKETS_LOST
 - WEBRTC_STATS_PROP_PACKETS_DISCARDED
 - WEBRTC_STATS_PROP_JITTER
 - WEBRTC_STATS_PROP_REMOTE_ID
 - WEBRTC_STATS_PROP_BYTES_RECEIVED
 - WEBRTC_STATS_PROP_PACKETS_DUPLICATED
 - WEBRTC_STATS_PROP_FIR_COUNT
 - WEBRTC_STATS_PROP_PLI_COUNT
 - WEBRTC_STATS_PROP_NACK_COUNT

[Version] 0.3.71
[Issue Type] API

Change-Id: I2e1d4f7bd65659dcdb9931c6df7505e3180836e9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_stats: Add support for skip callback of stats type or field not exported

[Version] 0.3.70
[Issue Type] Improvement

Change-Id: I87774427c28dea4197239f03947b792ea9d5268b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add statistics API

For now, only 'codec' statistics type is exported.
Other types will be supported with the following patches.

Function is added to get all the properties per statistics type.
 - webrtc_foreach_stats()

Callback function prototype is added as below.
 - typedef bool (*webrtc_stats_cb)(webrtc_stats_type_e type,
                                   webrtc_stats_prop_info_s *prop_info,
                                   void *user_data);

Enum is added as below for statistics type.
 - WEBRTC_STATS_TYPE_CODEC

Struct is added to be used as a parameter of the callback function.
 - webrtc_stats_prop_info_s

Enums are added as below for statisics property
 - WEBRTC_STATS_PROP_TIMESTAMP
 - WEBRTC_STATS_PROP_ID
 - WEBRTC_STATS_PROP_PAYLOAD_TYPE
 - WEBRTC_STATS_PROP_CLOCK_RATE
 - WEBRTC_STATS_PROP_CHANNELS
 - WEBRTC_STATS_PROP_MIME_TYPE
 - WEBRTC_STATS_PROP_CODEC_TYPE
 - WEBRTC_STATS_PROP_SDP_FMTP_LINE

Enums are added as below for statistics property data type
 - WEBRTC_STATS_PROP_TYPE_BOOL
 - WEBRTC_STATS_PROP_TYPE_INT
 - WEBRTC_STATS_PROP_TYPE_UINT
 - WEBRTC_STATS_PROP_TYPE_INT64
 - WEBRTC_STATS_PROP_TYPE_UINT64
 - WEBRTC_STATS_PROP_TYPE_FLOAT
 - WEBRTC_STATS_PROP_TYPE_DOUBLE
 - WEBRTC_STATS_PROP_TYPE_STRING

[Version] 0.3.69
[Issue Type] API

Change-Id: I52bbe25d6c03c4db1b0e0ffbf7c8f293da0b62a0
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_data_channel: Change the application of macro for coverage mesurement exclusion

It is updated as per the following ITC update.
: https://review.tizen.org/gerrit/#/c/test/tct/native/api/+/269285/

[Version] 0.3.68
[Issue Type] Line coverage

Change-Id: I623da2cf1986c42170533b67d6cc1bc1e5a9eef9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_stats: Add user callback parameters to _webrtcbin_get_stats()

Some improvements are also applied
 : Use gst_promise_new_with_change_func()'s notify parameter to free userdata
 : Rename __gststructure_foreach_cb() to __stats_field_foreach_cb()
 : Separate user data structure for __stats_field_foreach_cb()

[Version] 0.3.67
[Issue Type] Improvement

Change-Id: Iad04f0b544b0c2c311a83c0497996da6f47a6d72
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Use GStrv instead of gchar** on explict NULL-terminated vector string

use g_auto for GStrv whenever possible

[Version] 0.3.66
[Issue Type] Refactoring

Change-Id: I58458c31bf4ff6e384358b9eb3bf6be53d71c531

webrtc_stats: Update codec, remote-inbound-rtp and remote-outbound-rtp stats

[codec]
 channels, mime-type, codec-type and sdp-fmtp-line fields are added.

[remote-inbound-rtp]
 fraction-lost field is added.

[remote-outbound-rtp]
 packets-sent and bytes-sent fields are added.

These are newly added due to the GStreamer 1.20 update.

[Version] 0.3.65
[Issue Type] Update

Change-Id: I8857968b3f286d84bf2a54ec8391197d8acadb57
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] 0.3.64
[Issue Type] Bug fix

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

webrtc_stats: Revise to allow pre-defined fields per stats type

It is also possible to check easily fields incoming from gstreamer
which are not defined in this library yet.

[Version] 0.3.63
[Issue Type] Refactoring

Change-Id: Ied57fcf1ad4b350588d1456ddca56f6fe4003774
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_stats: Print values in __gststructure_foreach_cb()

[Version] 0.3.62
[Issue Type] Log

Change-Id: I9c17a00657e64459b4106741618bfd4fd84cd5db
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_stats: Fix to get valid user data in __webrtcbin_stats_cb()

[Version] 0.3.61
[Issue Type] Bug fix

Change-Id: I2393b2298118c52beb86eb8444d90978bc6c0e4c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

fixup! Add more mutex guard for callbacks

New mutex variable is defined for idle cb event source.

[Version] 0.3.60
[Issue Type] Improvement

Change-Id: Iee25695dbee13d25ff027baf39be79986a7bd9a0

webrtc_test: Fix data type to prevent integer overflow

[Version] 0.3.59
[Issue Type] SVACE

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

webrtc_data_channel: Reference data channel object in _on_data_channel_cb()

It is added due to the GStreamer update to 1.20 that includes patches below.
: https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/2186
Otherwise, some warning messages occur when releasing data channels.

[Version] 0.3.58
[Issue Type] Update

Change-Id: Ic1437a4e6e46610ed9eecd406208781d1b0231db
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add more mutex guard for callbacks

[Version] 0.3.57
[Issue Type] Improvement

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

webrtc_stats: Add support for masking stats type

Entering logs are added for each callback.

[Version] 0.3.56
[Issue Type] Improvement

Change-Id: I5e38d9640e5e0c4fa5c9520eacdcaf5edac0c58e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Fix codes along with GStreamer 1.19.3 update

[Version] 0.3.55
[Issue Type] Update

Change-Id: Icc8b596b7261e7e1a632edb9457e2be599bd01c9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Use gst_element_request_pad_simple()

gst_element_get_request_pad() is deprecated since GStreamer 1.19.1.

[Version] 0.3.54
[Issue Type] Update

Change-Id: I9a81e2d80a036d070d4ca3c00da511beac1e1dd4
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink/source: Add const keywords

[Version] 0.3.53
[Issue Type] Improvement

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

webrtc_source: Use transceiver pointer instead of mline variable

Since gst 1.19.x version, at this point in 'on-new-transceiver'
signal callback of webrtcbin, the transceiver mline index could be
set to -1. Therefore, it is changed to refer the transceiver object
itself.

[Version] 0.3.52
[Issue Type] Refactoring

Change-Id: I3b1b1eab5309362374df40dd96d89b1fff32257a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_data_channel: Remove unreachable code and revise error log

[Version] 0.3.51
[Issue Type] Improvement

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

Revise doxygen

FPS is mentioned to functions regarding video framerate.
Post command regarding error callback is described in case of
failure on sending data via data channel.

[Version] 0.3.50
[Issue Type] Doxygen

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

webrtc_test: Add menu to get mute

Unused definition is removed.
A space is added before asterisk in case of casting with pointer type.

[Version] 0.3.49
[Issue Type] Add

Change-Id: Id0a253eb77d55c4148138ed7019db9bfaacbe589
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add more macro to exclude lines from coverage measurement

[Version] 0.3.48
[Issue Type] Line coverage

Change-Id: Icfa24afaa6ca76d881a58a6a4bb27090c5936f99
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add space before asterisk in case of casting with pointer type

[Version] 0.3.47
[Issue Type] Coding convention

Change-Id: I574d1f547c66851d428c25c2f9d9125d939559aa
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add test cases for bundle policy and video frame rate

Menu items below are added.
 f. Set video framerate
 m. Get video framerate
 sbp. Set bundle policy
 gbp. Get bundle policy

[Version] 0.3.46
[Issue Type] Add

Change-Id: I9a147947a86e340726734e58b565f20d8c12cc69
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add 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] 0.3.45
[Issue Type] API

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

Add API to set/get video frame rate

Functions are added as below.
 - webrtc_media_source_set_video_framerate()
 - webrtc_media_source_get_video_framerate()

[Version] 0.3.44
[Issue Type] API

Change-Id: I3f48537153c245a17a1833404f4441513a2cf6c2
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Change gcov object install path

[Version] 0.3.43
[Issue Type] Gcov

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

Add missing parameter check code

[Version] 0.3.42
[Issue Type] Improvement

Change-Id: I89cb4efc0a9b533c000f76b65c05ee3cba7c90bf

Add omitted error checking in webrtc_create()

[Version] 0.3.41
[Issue Type] Improvement

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

Add new data channel buffered amount APIs

Functions are added as below.
 - typedef void (*webrtc_data_channel_buffered_amount_low_cb)()
 - webrtc_data_channel_get_buffered_amount()
 - webrtc_data_channel_set_buffered_amount_low_cb()
 - webrtc_data_channel_get_buffered_amount_low_threshold()
 - webrtc_data_channel_unset_buffered_amount_low_cb()

[Version] 0.3.40
[Issue Type] API

Change-Id: I3279925a1508955eded709927639ca2249c20137

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] 0.3.39
[Issue Type] Bug fix

Change-Id: Icff390fdd63ee2aa7bfeedd547d63dbb3e0f5d5a

webrtc_test: Add vp8 decoding pipeline when __DEBUG_VALIDATE_ENCODED_FRAME_CB__ is enabled

[Version] 0.3.38
[Issue Type] Debug

Change-Id: I1a99f64c86f10769b539e6a15a0fda64b1f19a4d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add opus decoding pipeline when __DEBUG_VALIDATE_ENCODED_FRAME_CB__ is enabled

[Version] 0.3.37
[Issue Type] Debug

Change-Id: I47828c656e7ac86ee9111434b53478f9f50d4d4d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_ini: Add new item to set in-band FEC and packet loss percentage

e.g)
[media source]
use inbandfec = no
packet loss percentage = 0

[source audiotest]
; values below will override the default one of [media source] above
use inbandfec = yes
packet loss percentage = 10

[Version] 0.3.36
[Issue Type] Improvement

Change-Id: If4fb6b658d02d7890ddb9924ebe3aceb5cdc4f08
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Rename payload id to payload type(pt)

New one is the term the most commonly used.

[Version] 0.3.35
[Issue Type] Refactoring

Change-Id: Ic0b070cab1fd445ae0bd327f382a4f3d349356ff
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Enable in-band FEC of OPUS encoder

Revise caller of g_object_set()/get() to use multiple lines.

[Version] 0.3.34
[Issue Type] New feature

Change-Id: I8f514758e0e768c1ffad1f2288c87207f963c05a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Enable in-band FEC of OPUS decoder

[Version] 0.3.33
[Issue Type] New feature

Change-Id: I48d271615a1f742e15c2ba2d10e4de023d4430cc
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Change parameter and use existing macro to print some log

[Version] 0.3.32
[Issue Type] Refactoring

Change-Id: I46e430d17016bffb4c5b64a88c9597a147564aa2
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] 0.3.31
[Issue Type] Improvement

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

Fix resource leak when get caps from pad

[Version] 0.3.30
[Issue Type] Resource leak

Change-Id: I1123877111921b88c898299cc2a9ef1f7ae3dec1

webrtc_test: Add menu to set/get RTP packet drop probability

sdp. Set RTP packet drop probability
gdp. Get RTP packet drop probability

[Version] 0.3.29
[Issue Type] New feature

Change-Id: I40899c4948614e0b94fd2f8485335b38e21533ca
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Remove unused internal API

Use webrtc_set[get]_rtp_packet_drop_probability() instead.

[Version] 0.3.28
[Issue Type] Clean up

Change-Id: I1e32d62c2a727aba58dd0cf8cf43dec096fd00f7
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_internal: Add APIs to set/get RTP packet drop probability

Functions below are added.
 - webrtc_set_rtp_packet_drop_probability()
 - webrtc_get_rtp_packet_drop_probability()

RTP packets can be dropped before sending or after being received
with this new function.

[Version] 0.3.27
[Issue Type] New feature

Change-Id: I8d2ca412bb272d750e40d39332d999e3b0cc0085
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Parse fmtp attribute and save useinbandfec value to the handle

It is parsed from a remote description. This information will be used
to set related property to a decoder.

[Version] 0.3.26
[Issue Type] New feature

Change-Id: Ieef7a244405c617d64b28c069f5f55cbc65fde69
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

fixup! webrtc_private: Add structure for data recovery types and update it

One more condition is added for an offerer.
 1. set local offer description (changed to HAVE_LOCAL_OFFER)
 2. set remote answer description (changed to STABLE)

Function name is also changed.

Change-Id: I2a2bdc5058c41750a2a0e46c50b7112cda6b0a72

Fix caps double unref when making element

[Version] 0.3.25
[Issue Type] Bug fix

Change-Id: I8ac9b81d6301eb0c519d6976b6fe620cad6bade5

Separate the code to get encoder

[Version] 0.3.24
[Issue Type] Refactoring

Change-Id: Iee321f4b3126c0d1545421d95594076b395ccd4a

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] 0.3.23
[Issue Type] Improvement

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

webrtc_private: Add some parsing functions for rtpmap attribute

A verbose log for attribute key/value is added.

[Version] 0.3.22
[Issue Type] Improvement

Change-Id: I6412f5842f196b2310c42a48b39ae129d3c54e5c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Improve codes for setting transceiver FEC

It is revised to update the values before creating offer/answer.
Logic is branched by situation if it is an offer or answer.

[Version] 0.3.21
[Issue Type] Improvement

Change-Id: I3e71683ed0adc4413cab30f2ef27b941db12e9de
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Revise FEC setting condition in case of answerer without media source

In this case, it is fixed to depend on related values of remote offer description
instead of the ini value.

'fec-percentage' is not required for a 'recvonly' transceiver.

[Version] 0.3.20
[Issue Type] Improvement

Change-Id: I5de375283865ae604e590c7a74c099a649b3354c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Add structure for data recovery types and update it

It is updated from remote offer description. And it'll be used to
determine to enable the recovery mechanism for answerer.

[Version] 0.3.19
[Issue Type] Improvement

Change-Id: I35716c66381cfae8931f17b07817d0c93f52395b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Fix resource leak when get caps from pad

[Version] 0.3.18
[Issue Type] Resource leak

Change-Id: Ia25db3063b47c6f00c59f2cfabf0b8cb0a22c38b

Improve the code to check media type

[Version] 0.3.17
[Issue Type] Refactoring

Change-Id: Icc49b8e30ab2744d6d08a64f59dcc2b7c4b93c30

Move internal file source functions to the public header

Functions below are moved to the public header.
 - webrtc_file_source_set_path()
 - webrtc_file_source_set_looping()
 - webrtc_file_source_get_looping()

[Version] 0.3.16
[Issue Type] API

Change-Id: I62ae2ca6a8a52dde7a9471e894476a61b4d59c45

webrtc_private: Use const keyword to return type of _ini_get_source_by_type()

The value of returned structure pointer should not be modified.
All the callers are also modified to use the keyword.

[Version] 0.3.15
[Issue Type] Improvement

Change-Id: I14b6c2184835fe94dcf4f0cf4403242068e82cbb
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Set source element properties values from ini file

[Version] 0.3.14
[Issue Type] Improvement

Change-Id: I3367d1d8ae0f5b20be644330a3131c9906edcefd
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Use _av_tbl to get element name

[Version] 0.3.13
[Issue Type] Refactoring

Change-Id: Ic6c2aacd44f4a01bf65f0b8ec5c91c011a93b843

Blocking filesrc streaming until the connection with peer is completed

Common streaming sending is blocked while connecting with peer.
but until now, filesrc has not been blocked because filesrc constructs it's own pipeline.

[Version] 0.3.12
[Issue Type] Improvement

Change-Id: I18c45e075b5b1db6e78822d7af2ee6a0148a8d8b

webrtc_ini: Add new item to set source element properties

ini file example)
[source xxx]
source element properties = prop_name1=value1, prop_name2=value2

_gst_set_element_properties() is added to set property list
to an element.

[Version] 0.3.11
[Issue Type] New feature

Change-Id: Ic9bd7e7c019c8eb6c086b48d33dbe572e6ed23f5
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] 0.3.10
[Issue Type] Improvement

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

Add new audio/video loopback unset APIs

Functions are added as below.
 - webrtc_media_source_unset_audio_loopback()
 - webrtc_media_source_unset_video_loopback()

[Version] 0.3.9
[Issue Type] API

Change-Id: I78612a39367a56891bed8c7d6c6fa6aeae007098

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_test: Add sub-menu to use mic only in room case

Parameter of _webrtc_add_media_source() is revised.

[Version] 0.3.8
[Issue Type] Add

Change-Id: I8312d9f19701b2859bc736745fc54d1c79b8ab57
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] 0.3.7
[Issue Type] Improvement

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

webrtc_source: Fix typos

DEFAULT_NAME_XXX should be ELEMENT_NAME_XXX.

[Version] 0.3.6
[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] 0.3.5
[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] 0.3.4
[Issue Type] Refactoring

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

webrtc_test: Add menu to set/get RTP packet drop probability

[Version] 0.3.3
[Issue Type] New feature

Change-Id: Icee1dcba86b477a44a97708481a006b2f36c28fb
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>