webrtc_ini: Add new item to check for requirement of video overlay resource

If the video sink element for video overlay requires a resource
acquisition, 'video overlay resource required' item should be set
to 'yes' in the configuration file.

e.g.)
[rendering sink]
video overlay resource required = yes

Some logs in __dump_ini() is revised.
Coding style is a little changed in _acquire_resource_if_needed().
It is changed that creating video sink element before acquiring a resource.

[Version] 0.2.10
[Issue Type] Improvement

Change-Id: I5ae7948a155c1c1b02f129610ee00a491cab0c96
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Remove unnecessary build definition

These are the resource manager dependency codes for public tizen platform.
It's not necessary to make a particular definition as the previous one.

For tizen tv binary building, since the resource management is not used
rather, '#ifndef TIZEN_TV' is now used.

[Version] 0.2.9
[Issue Type] Build definition

Change-Id: Ibc67d3f346c7016fddc5344b0626298e02222612
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Use GList to carry elements via parameter

[Version] 0.2.8
[Issue Type] Refactoring

Change-Id: Id19b2b6052799491b4c7558ea18170e203d94746
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Ensure the NEGOTIATING state to get ready for negotiation operation

Invoking state callback for NEGOTIATING state is moved to the inside of
negotiation needed callback. This ensures to get the SSRC parameter
successfully while creating offer SDP.

[Version] 0.2.7
[Issue Type] Improvement

Change-Id: I4cfe0f5214b12925170ba2cd6fbfa4c52a66daa5
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Merge __build_audiosrc() with __build_audiotestsrc()

[Version] 0.2.6
[Issue Type] Refactoring

Change-Id: Ice41abc6e0b92de41bacc95fac7dc2dd620df018
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add support for the dynamic resolution change

webrtc_set_video_source_resolution() API is revised to support
for this feature during the streaming.
 - state limitation inside of this function is released.
 - doxygen is revised according to this change.

Note that if the source element does not support the DRC, this API
will return WEBRTC_ERROR_INVALID_OPERATION.

If the DRC is supported for a particular source, 'video drc support'
field should be set to 'yes' in ini configuration file.

e.g.)
[source videotest]
video drc support = yes

[Version] 0.2.5
[Issue Type] New feature

Change-Id: I145fe0cb139e06d75611eadb10d74b9ba0858198
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add new group - CAPI_MEDIA_WEBRTC_DATA_CHANNEL_MODULE

Some are fixed for coding convention.

[Version] 0.2.4
[Issue Type] Doxygen

Change-Id: I3cab0a43aa6c6afae726a592ee5992f186494004
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Change path of configuration file

It is changed
 - from /etc/multimedia/mmfw_webrtc.ini
 - to /hal/etc/multimedia/mmfw_webrtc.ini

[Version] 0.2.3
[Issue Type] Update

Change-Id: I1b8e1356a88be40377d4234e44007acb9624fcf6
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Set latency to webrtcbin directly

The webrtcbin based on 1.18 version has latency property that
set the latency property of rtpbin internally.

[Version] 0.2.2
[Issue Type] Update

Change-Id: I18716dacc202ad195b844365e9e8c21300d1d121
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Set state to PLAYING when peer connection state is changed to CONNECTED

This state change is affected by webrtcbin changes based on 1.18 version.

[Version] 0.2.1
[Issue Type] Update

Change-Id: I4b590ccb43e468dc5add1b1282a97ea17083a514
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Update spec file

[Version] 0.1.178
[Issue Type] Improvement

Change-Id: I4e5ccf682e8fccf83ae4132d6f9cdf4b7564999a

Add new API set for media source mute

black frame or silent sound will be transmitted to the remote peer.
Functions are added as below.
-webrtc_set_media_source_mute()
-webrtc_get_media_source_muted()

[Version] 0.1.177
[Issue Type] API

Change-Id: I4134872691f5f4a46cbd0b510a15a4fd521fb8e7

Fix build error at emul

[Version] 0.1.176
[Issue Type] Improvement

Change-Id: I516cbe551b8af0a6c1777596b2594c62770d11e6

webrtc_source: Set zerocopy_enabled variable outside of __create_rest_of_elements()

Warning log is added.
Invalid branch is fixed. (if -> else if)

[Version] 0.1.175
[Issue Type] Refactoring

Change-Id: Ifa4dd69b921ac39f572d34873505626991db6e35
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Check if the extra field of media packet is gstreamer buffer

It is added to avoid failure on 'push-buffer' signal function when the extra
field of the media packet is not a gstreamer buffer.

[Version] 0.1.174
[Issue Type] Improvement

Change-Id: I30643df4cc86bedf0ff116c58eff1893659d947c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc: Add error condition in webrtc_set_display()

The track added callback will be invoked even if the encoded video frame callback
has been set. Within this callback, this patch let webrtc_set_display() return
the invalid operation error by looking the encoded video frame callback is set,
not by _is_owner_of_track_build_context() the previous one.

Doxygen is also updated.

[Version] 0.1.173
[Issue Type] Improvement

Change-Id: If889e51602de878cf369aaadbda1d99ca63b4fe9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Limit expected size of receiving data via data channel

[Version] 0.1.172
[Issue Type] Test application

Change-Id: I18a0ec0bc0f9925ce687859e8d3c771226d4d16c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add API set for media source pause

These correspond to the 'replaceTrack()' method with 'null' or
'track' parameter of the RTCRtpSender.

Functions are added as below.
 - webrtc_set_media_source_pause()
 - webrtc_get_media_source_pause()

Test cases are also added in webrtc_test.
 p. Pause/play media source
 o. Get the pause of media source

[Version] 0.1.171
[Issue Type] API

Change-Id: I7a67eb4dea0c386876b901ece5aa5660968da776
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc: Add invalid operation error return value to webrtc_destroy()

Gstreamer pipeline state change error or resource release error can
occur during destroying webrtc handle. In this case, it will return
WEBRTC_ERROR_INVALID_OPERATION error value.

[Version] 0.1.170
[Issue Type] Improvement

Change-Id: Ia17540054fbba9205820ac533fbbfb936d84f5aa
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Add Macro definition for printing caps and apply it

[Version] 0.1.169
[Issue Type] Improvement

Change-Id: I82a007e7ea059df648e0c4feecb48e932804293e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Reset all the negotiation states in _webrtc_stop()

The webrtc_stop() function changes the handle state to WEBRTC_STATE_IDLE.
It'll be changed to WEBRTC_STATE_NEGOTIATING again after calling
webrtc_start(), hence this patch is needed to set the states to
initial one.

Test case to get all the negotiation states is added.
 gan. Gets all the negotiation change callbacks

[Version] 0.1.168
[Issue Type] Improvement

Change-Id: Ib9aabafcdc2528303e65c58960f55c5451f49ecc
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Get ready for audio/video source pads in a media source

This patch is to prepare a media source that have two source pads.
Currently, we only support two kinds of stream, audio and video.

[Version] 0.1.167
[Issue Type] Improvement

Change-Id: I2fbccfc66d2fff36faa98ae98c01a8051a71216b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Replace some macro definitions to enum types

MLINES_IDX_AUDIO is replaced to AV_IDX_AUDIO.
MLINES_IDX_VIDEO is replaced to AV_IDX_VIDEO.
AV_IDX_MAX is added.

[Version] 0.1.166
[Issue Type] Improvement

Change-Id: Idb3ea3a120730f41b4c58f5984ae287b1937b016
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add test cases for new negotiation callbacks

san. Set all the negotiation change callbacks
uan. Unset all the negotiation change callbacks

[Version] 0.1.165
[Issue Type] Test application

Change-Id: Ib7eb8e7f89806d6fda22d32ecc12d94d2e5501c4
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add API set for the ICE connection state

These correspond to the 'iceConnectionState' and 'oniceconnectionstatechange'
property of the RTCPeerConnection respectively.

Enums are added as below.
 - WEBRTC_ICE_CONNECTION_STATE_NEW
 - WEBRTC_ICE_CONNECTION_STATE_CHECKING
 - WEBRTC_ICE_CONNECTION_STATE_CONNECTED
 - WEBRTC_ICE_CONNECTION_STATE_COMPLETED
 - WEBRTC_ICE_CONNECTION_STATE_FAILED
 - WEBRTC_ICE_CONNECTION_STATE_DISCONNECTED
 - WEBRTC_ICE_CONNECTION_STATE_CLOSED

Functions are added as below.
 - webrtc_set[unset]_ice_connection_state_change_cb()
 - webrtc_get_ice_connection_state()

[Version] 0.1.164
[Issue Type] API

Change-Id: If11019e7d57667d031c676e12c8c688fcc03da30
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add API set for the ICE gathering state

These correspond to the 'iceGatheringState' and 'onicegatheringstatechange'
property of the RTCPeerConnection respectively.

Enums are added as below.
 - WEBRTC_ICE_GATHERING_STATE_NEW
 - WEBRTC_ICE_GATHERING_STATE_GATHERING
 - WEBRTC_ICE_GATHERING_STATE_COMPLETE

Functions are added as below.
 - webrtc_set[unset]_ice_gathering_state_change_cb()
 - webrtc_get_ice_gathering_state()

[Version] 0.1.163
[Issue Type] API

Change-Id: I3076aede623598d191b3792ec9cb2dcf849b325d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add API set for the signaling state

These correspond to the 'signalingState' and 'onsignalingstatechange'
property of the RTCPeerConnection respectively.

Enums are added as below.
 - WEBRTC_SIGNALING_STATE_STABLE
 - WEBRTC_SIGNALING_STATE_HAVE_LOCAL_OFFER
 - WEBRTC_SIGNALING_STATE_HAVE_REMOTE_OFFER
 - WEBRTC_SIGNALING_STATE_HAVE_LOCAL_PRANSWER
 - WEBRTC_SIGNALING_STATE_HAVE_REMOTE_PRANSWER
 - WEBRTC_SIGNALING_STATE_CLOSED

Functions are added as below.
 - webrtc_set[unset]_signaling_state_change_cb()
 - webrtc_get_signaling_state()

[Version] 0.1.162
[Issue Type] API

Change-Id: Ide27303b7905567654fa1a67359f60a237d20bb6
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add API set for the peer connection state

These correspond to the 'connectionState' and 'onconnectionstatechange'
property of the RTCPeerConnection respectively.

Enums are added as below.
 - WEBRTC_PEER_CONNECTION_STATE_NEW
 - WEBRTC_PEER_CONNECTION_STATE_CONNECTING
 - WEBRTC_PEER_CONNECTION_STATE_CONNECTED
 - WEBRTC_PEER_CONNECTION_STATE_DISCONNECTED
 - WEBRTC_PEER_CONNECTION_STATE_FAILED
 - WEBRTC_PEER_CONNECTION_STATE_CLOSED

Functions are added as below.
 - webrtc_set[unset]_peer_connection_state_change_cb()
 - webrtc_get_peer_connection_state()

[Version] 0.1.161
[Issue Type] API

Change-Id: I271413364e4e3704d1df021d9174aac6a18cd225
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add API set for video source resolution

Functions are added as below.
 - webrtc_set_video_source_resolution()
 - webrtc_get_video_source_resolution()

Test cases for these are added in webrtc_test.

[Version] 0.1.160
[Issue Type] API

Change-Id: Iea0ec96d4e05f900cd4d83e9480e38fb604ad1ab
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Invoke state changed callback in the main context

Some of caller of state changed callback invoked it in a thread context.
It is now fixed that all of the callbacks are called in the main context.

[Version] 0.1.159
[Issue Type] Improvement

Change-Id: Ib3e1047e306247405189bdd2bededed558edec37
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Revise descriptions

Add missing descriptions for enums.
Remove redundant @param.

[Version] 0.1.158
[Issue Type] Doxygen

Change-Id: Id1a2703304d3682f636a3f291ec4809da3abefeb
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Print turn server log in webrtc_foreach_turn_server()

[Version] 0.1.157
[Issue Type] Logs

Change-Id: I1bf0532a90bf471b49d7c9758942eaecb54c865f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Remove unnecessary capsfilter in screen source pipeline

Unnecessary logs are removed.

[Version] 0.1.156
[Issue Type] Improvement

Change-Id: I3dcc19f1bdb07c74cabd367bf4151c0138b4ebd6
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Revise error values

Return WEBRTC_ERROR_INVALID_OPERATION if handle has a member of
not expected value.

[Version] 0.1.155
[Issue Type] Improvement

Change-Id: Ibeb30dc6b0864bb6ecd492af1ddf2444b2cc9bf7
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Fix untrusted conversion from string to number

Use g_ascii_strtoll() instead of atoi().

[Version] 0.1.154
[Issue Type] Improvement

Change-Id: I0e450dd2a7fc6a75bc3a70c997652745bafcdf9f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Add WEBRTC_MEDIA_SOURCE_TYPE_SCREEN to media source type

- Function to use screen as a media source

[Version] 0.1.153
[Issue Type] New API

Change-Id: Iadb687bd4f3b94cfc4b7d5d0555a8a4874184c30
Signed-off-by: Hyunil <hyunil46.park@samsung.com>

webrtc_source: Release gst resources in case of error

[Version] 0.1.152
[Issue Type] Improvement

Change-Id: Ic9a5e8ffadae1384e5d3e2b7787e2df29382fabe
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Change type GstElement to GstBin for bin

[Version] 0.1.151
[Issue Type] Improvement

Change-Id: I10ba655d266ba5c6ce1eaff125ed6283d989cc8d

webrtc_private: Destroy sink pipeline in case of webrtc_stop()

When the webrtc_stop() is called, the state is changed to IDLE.
The state will be changed to NEGOTIATING with following webrtc_start()
again which means the re-negotiation is required.

Therefore, releasing the previous sink pipeline should be performed
inside of webrtc_stop().

[Version] 0.1.150
[Issue Type] Improvement

Change-Id: I1bd03ecf8b8f27d93dec7d1488fa20223299a737
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_data_channel: Revise logs

Printing channel handle logs are added.

[Version] 0.1.149
[Issue Type] Log

Change-Id: I36fd5d292af3817c33f239fe3e697ad5f888110f
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc: Remove state change to READY in webrtc_create_data_channel()

Previously, READY state of webrtcbin was required to create the
data channel. Due to this condition, we raise the state to READY
right before creating a data channel. But this state change invokes
the negotiation needed callback unexpectedly.

In webrtcbin, the state limitation to create a data channel has been
removed.
 : https://review.tizen.org/gerrit/#/c/platform/upstream/gst-plugins-bad/+/256742/

Therefore, this state change is also no longer necessary and removed.

[Version] 0.1.148
[Issue Type] Improvement

Change-Id: Ibefe084aaf68639f2bfb750bac1866d7445e24d8
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add API set for TURN server

 Functions are added as below.
 - webrtc_add_turn_server()
 - webrtc_foreach_turn_server()

 Callback prototype
 - typedef bool (*webrtc_turn_server_cb)(
                 const char *turn_server, void *user_data);

[Version] 0.1.147
[Issue Type] API

Change-Id: Iadfd91db135167556b801ef9dab3a455ca0d3e1f

webrtc_source: Add support for media packet with zerocopy buffer

If the incoming media packet has tbm surface, it should be used to
allocate memory with gst_tizen_allocator_alloc_surface() to have
zerocopy method.

[Version] 0.1.146
[Issue Type] Improvement

Change-Id: I1686f25b7140bf47331b7cd036cba0b3b4e5584e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc: Change ownership of the pushed packet

Regarding webrtc_media_packet_source_push_packet(), it is now changed
to take the ownership of the packet. Therefore, a caller should not
release the packet after pushing it using the API from now on.

Description about it is added to the function.
webrtc_test is modified according to this change.

[Version] 0.1.145
[Issue Type] API

Change-Id: I88dfc003365c6167c421feb594c037d79a90e5f6
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add missing gst_buffer_unref()

[Version] 0.1.144
[Issue Type] Test application

Change-Id: I3e6b69854157ec5743475d6422605295dc26f1f5
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Remove unused code

[Version] 0.1.143
[Issue Type] Improvement

Change-Id: I6d6d7fe7f0b572ccbd040fde56955af33b535dd5

webrtc_source: Set 'empty-buffer-timeout' property to 0 if needed

If camerasrc element has this property, set it to 0 not to emit an error
even if there's no buffer consumption.

[Version] 0.1.142
[Issue Type] Improvement

Change-Id: Ie9ecc828c6a856aeb63beddae3aeea0cd7212fc0
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Support zerocopy format

It affects to make it possible to use tizen zerocopy format(SN12, S420)
for camera source and media packet source.

[Version] 0.1.141
[Issue Type] Improvement

Change-Id: I3768aefe382e7dd43139a6e37a14e28671829c03
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add Quit menu to terminate program

[Version] 0.1.140
[Issue Type] Test application

Change-Id: I3cbd5a1a6d9330876dcb365b89391706730e2e97

webrtc: Add new API webrtc_data_channel_get_label()

[Version] 0.1.139
[Issue Type] API

Change-Id: I02e845b4a53ba1e6c1c5c789eb368966a545037a

Add options parameter to webrtc_create_offer()/answer()

This parameter is added to support offer/answer options of
https://www.w3.org/TR/webrtc/#offer-answer-options.

It currently has no effect, because the Gstreamer side also
does not support it yet.

[Version] 0.1.138
[Issue Type] API

Change-Id: Ied29c6a918a7634afaea4716e51358e5666ba506
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Improve __get_hw_encoder_element()

Obtain hardware resources by checking the media type instead of
the source type. Each source can have a different type, but the
media type can only be video or audio. Therefore, this function
can cover all source types even if a new source type is added.

[Version] 0.1.137
[Issue Type] Refactoring

Change-Id: Ia760a5299a6f18568deea175745674011df52bf6
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_options/data_channel: Apply data channel options

Doxygen of webrtc_create_data_channel() is also updated.

[Version] 0.1.136
[Issue Type] Improvement

Change-Id: I8238792a63716e43069f0862c1656448517aca92
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_options: Add functions to handle options

The options parameter is a list of key value pairs made by bundle API.
The functions are added to convert the list to the gststructure with
valid key value pairs and print these for easy to debug.

[Version] 0.1.135
[Issue Type] New feature

Change-Id: If6e7fd4d584b01f0c1f011dd077b830f53808502
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add new parameter for options to webrtc_create_data_channel()

It is added to support configuration options for the data channel.

The 3rd parameter is added as below.
 - int webrtc_create_data_channel(webrtc_h webrtc,
                                  const char *label,
                                  bundle *options,
                                  webrtc_data_channel_h *channel)

[Version] 0.1.134
[Issue Type] API

Change-Id: I21ec0613227e9a039dcb52b5c06ee1c1b9970468
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc : check invalid parameter at the beginning of webrtc_set_display()

[Version] 0.1.133
[Issue Type] Improvement

Change-Id: I1d744d05b270f15af7f4835d63e476d67d8d97d7

Fix SVACE defect

- DEREF_OF_NULL.RET.ALLOC
- PASS_TO_PROC_AFTER_CHECK
- INVARIANT_RESULT.OP_ZERO
- NO_EFFECT
- UNREACHABLE_CODE

[Version] 0.1.132
[Issue Type] Svace

Change-Id: I48afb8b4a5842c3976d57fa190542a04e2b20935

webrtc_private: Print error as string before invoking error callback

[Version] 0.1.131
[Issue Type] Log

Change-Id: If6f588c60fadca60f043a24708ec0cdc289ce804
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private : remove duplicated code

remove simple wrapper functions _webrtcbin_create_offer() and _webrtcbin_create_answer().

[Version] 0.1.130
[Issue Type] Improvement

Change-Id: I1d586dd7488e36e8c8a5ab6a996b8b1fd9f782ef

webrtc_source : add missing parameter check code

plus, fix wrong return value

[Version] 0.1.129
[Issue Type] Improvement

Change-Id: I70cc965b7074047802edb0cabd36ff73f21b2f37

webrtc_sink: Post error if it failed to build rendering pipeline

[Version] 0.1.128
[Issue Type] Improvement

Change-Id: I68c899e074da9b347284d9598e650423bada0856
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc: Add @remarks for default transceiver direction value

Typos are also fixed.

[Version] 0.1.127
[Issue Type] Doxygen

Change-Id: I4711609fbe04335e380d643d3110473e4ddc1ebd
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc : improve api description

add missing error codes and webrtc state.
remove unnecessary error codes.

[Version] 0.1.126
[Issue Type] Improvement

Change-Id: I573ec8475cb49cd0e87a11c2f535661cf711218c

webrtc : minor code changes

1. return error directly.
2. Add build option for test codes.

[Version] 0.1.125
[Issue Type] Improvement

Change-Id: I8f4fc393d7adf0d658eadef0e40b1346794dd618

webrtc: Add missing error codes to the description

Missing error codes are added to the description of callback prototypes below.
 : webrtc_error_cb()
 : webrtc_data_channel_error_cb()

[Version] 0.1.124
[Issue Type] Doxygen

Change-Id: I0ca78260580a14ad1ed84e1ab477bdf8b92eeff3
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Allow to push media packet of raw format

It is added to check to match the format of pushing packet
with the configured format.

[Version] 0.1.123
[Issue Type] Improvement

Change-Id: Ib443634c62d6f588768a5ecffe6fb1e9e4172612
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Set width/height to media format for encoded frame callback

In case of video format, these will be set if caps has width and height
information in its structure.

[Version] 0.1.122
[Issue Type] Improvement

Change-Id: I77da9d41ec3ac3c3a63d31f34974161511bc4d1e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Change log level in __media_packet_finalize_cb()

The log level is changed to INFO due to the importance of releasing
the packet by app side.

[Version] 0.1.121
[Issue Type] Log

Change-Id: Ic4f71e18bb5e3e0a918741cff8ae94d3c112b704
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

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

- Referred to https://www.webmproject.org/docs/encoder-parameters and
  https://developers.google.com/media/vp9/the-basics#quality_and_speed_settings

[Version] 0.1.120
[Issue Type] Improvement

Change-Id: Ic3647fa6642a1ff06e3839250838d035add69531
Signed-off-by: Hyunil <hyunil46.park@samsung.com>

webrtc_test: Change media test file path in case of TV profile build

To do this, spec and CMakefile are changed.

[Version] 0.1.119
[Issue Type] Test application

Change-Id: I913c8b9066ba92759c7299262d7a3b20b267855c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_resource: Invoke error callback when a resource conflict happens

_post_error_cb_in_idle() is added to invoke the error callback in
the main thread.

[Version] 0.1.118
[Issue Type] Improvement

Change-Id: I35290cb5ae37970cd599fc74e90e499837e50cd9
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Revise _post_state_in_idle()

Rename _post_state_in_idle() to __post_state_cb_in_idle().
Remove prototype in webrtc_private.h.
Allocate userdata to have expansibility.

[Version] 0.1.117
[Issue Type] Refactoring

Change-Id: Ie7e095501dc443474d464d6de78698e54220aed0
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Apply designated initializers to arrays

[Version] 0.1.116
[Issue Type] Refactoring

Change-Id: I514bb42a8b8ee9ef5be1a58a8118608c2ecde42e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add local rendering feature with encoded frame callback

__DEBUG_VALIDATE_ENCODED_FRAME_CB__ definition is added to test
the media packet for H264 encoded data received from the encoded
frame callback with local rendering pipeline.

[Version] 0.1.115
[Issue Type] Test application

Change-Id: I27c697737226d4715576a7057b4c2d66c6bbef79
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Add support for media packet with gstbuffer pointer

In case of using media packet source, now we support two kinds of
media packet. One is normal buffer which memory is allocated inside
of the packet, the other one is having an external buffer without
allocation. In the latter case, we assume that extra data of the
media packet has a gstreamer buffer pointer.

A test case for this is also added in webrtc_test.c.

[Version] 0.1.114
[Issue Type] Improvement

Change-Id: I391de40b6f5217e6a797e8a19d97a5640e093632
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Revise media packet for encoded frame callback

The gstbuffer data field is set as external memory pointer of the
media packet. The pointer of gstbuffer is set to extra data of the
media packet. If the caps of the pad has 'codec_data' field, it is
set to the media packet.

[Version] 0.1.113
[Issue Type] Improvement

Change-Id: Ib361a0a9a2600b262c60486c22f27850a21cc5d0
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Set 'do-timestamp' to TRUE to appsrc in case of media packet source

Setting 'emit-signals' to TRUE is also removed because it is default value.

[Version] 0.1.112
[Issue Type] Improvement

Change-Id: I390449180fd7ad91cde380d078a1a28b61ebc2a4
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_ini: Add log for FEC setting

[Version] 0.1.111
[Issue Type] Log

Change-Id: I5783988171c0c23e6d846ebad91314e3b24d2e94
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc: Add missing return check when changing pipeline state to READY/PLAYING

[Version] 0.1.110
[Issue Type] Bug fix

Change-Id: I349a7da7e39a95bf0c93e7a81d9f97165e75c309
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private: Set bundle-policy to max-bundle

[Version] 0.1.109
[Issue Type] Improvement

Change-Id: I49cf86977d7abb2574eaff77e951adc6e665aa5e
Signed-off-by: Sangchul Lee <sangchul1011@gmail.com>

webrtc_source: Add missing unref call of the media format when an error occurs

[Version] 0.1.108
[Issue Type] Bug fix

Change-Id: I79fcbe483146e4463196588e30a27af6231afd0a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Apply network related features to webrtc_create() API

One of features below must be supported to use this webrtc API set.
 - http://tizen.org/feature/network.wifi
 - http://tizen.org/feature/network.telephony
 - http://tizen.org/feature/network.ethernet

[Version] 0.1.107
[Issue Type] Feature

Change-Id: I9719036e4346f0f56919e4ebb6299c1676a3424c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc: Print warning log before overwriting user callback address

[Version] 0.1.106
[Issue Type] Log

Change-Id: I80848724b4ef90ac1f1d357c166a5ac9a4a29017
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_sink: Add capsfilter to apply stream-format and alignment in case of H264/H265

In case of H264/H265 encoded frame callback, it is added to apply
'byte-stream' of stream format and 'au' of alignment, as default values.

[Version] 0.1.105
[Issue Type] Improvement

Change-Id: I352962743b714d3e45f4854faeb9984d357d304e
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add test cases for encoded audio/video frame callback

[Version] 0.1.104
[Issue Type] Test application

Change-Id: Ide975cd7fc5e5abf1029e4f98b2c1cce12ad70d3
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Add API set for encoded audio/video frame callback

Functions are added as below.
 - webrtc_set_encoded_audio_frame_cb()
 - webrtc_unset_encoded_audio_frame_cb()
 - webrtc_set_encoded_video_frame_cb()
 - webrtc_unset_encoded_video_frame_cb()

Callback prototype
 - typedef void (*webrtc_encoded_frame_cb)(webrtc_h webrtc,
                  webrtc_media_type_e type, unsigned int track_id,
                  media_packet_h packet, void *user_data)

[Version] 0.1.103
[Issue Type] API

Change-Id: I96c897f735a549fc84514b02727158b7e6892819
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_private/sink: Add forwarding sink to support the encoded frame callback

[Version] 0.1.102
[Issue Type] New feature

Change-Id: I680cd9c3e00df22fe0e98c6de1df427f9e4d2924
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add H264 test case for media packet source

__DEBUG_VALIDATE_MEDIA_PACKET__ definition is added to test
the media packet for H264 encoded data only with local rendering
pipeline.

[Version] 0.1.101
[Issue Type] Test application

Change-Id: I8f3d4ffd9113ec61f1d224f9a43f7d6942591067
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Revise caps information of appsrc in case of H264 format

To make negotiation with incoming media packet source of H264 byte stream data
caps information set to appsrc is revised.

[Version] 0.1.100
[Issue Type] Improvement

Change-Id: I0a95e4f5bdb1a02e195095ce1b862409fd65c64d
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_source: Remove H263-related codes

These will not be used.

[Version] 0.1.99
[Issue Type] Clean-up

Change-Id: Idac647008161b5d7ea95e9c499b7eea7e9e34976
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Apply camera/microphone feature to webrtc_add_media_source() API

For WEBRTC_MEDIA_SOURCE_TYPE_CAMERA type
  feature: http://tizen.org/feature/camera

For WEBRTC_MEDIA_SOURCE_TYPE_MIC type
  feature: http://tizen.org/feature/microphone

[Version] 0.1.98
[Issue Type] Feature

Change-Id: Ife9d500d3c2ee743aa133a21134105429f5f0c61
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_restriction: Add function to check feature

RET_ERR_IF_FEATURE_IS_NOT_SUPPORTED macro is also added in
webrtc_private.h.

[Version] 0.1.97
[Issue Type] Feature

Change-Id: I6b154da971830d2392fe691370fe88ec88afddfc
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_test: Add test cases to use signaling server/client API

[Version] 0.1.96
[Issue Type] Test application

Change-Id: I72388c8c15babfbec032ca85b956c0c29031480a
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_signaling_client: Parse message and forward it to user callback

A condition to allow only SDP or ICE candidate message is also added
in webrtc_signaling_send_message().

[Version] 0.1.95
[Issue Type] Implementation

Change-Id: Ia5c4de309c25825b5b94d81f0ecc30ed848255c1
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_signaling_client: Add internal API set for client of signaling server

Functions are added as below.
 - webrtc_signaling_connect()
 - webrtc_signaling_request_session()
 - webrtc_signaling_send_message()
 - webrtc_signaling_get_id()
 - webrtc_signaling_disconnect()

[Version] 0.1.94
[Issue Type] API

Change-Id: I95fe182d70cb6f255abee96a7e99793b3777a269
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_signaling_server: Handle received SDP or ICE candidate message

[Version] 0.1.93
[Issue Type] Implementation

Change-Id: Ic265706a4c6fbec017444ad35178bdc7850d7441
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_signaling_server: Handle request session message

Codes in LWS_CALLBACK_ESTABLISHED case are also revised
with new function.

[Version] 0.1.92
[Issue Type] Implementation

Change-Id: I710cd45fe1bdbc1c76c2a14f98f6990fddffe04c
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_signaling_server: Assign new id to incoming client

[Version] 0.1.91
[Issue Type] Implementation

Change-Id: Ia16f762883f232fab937ee3fb2c055f2f1102a7b
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

webrtc_signaling_server: Add internal API set for signaling server

This signaling server is only for private network.
Handling messages between server and client will be added with
following patches.

Functions are adde as below.
 - webrtc_signaling_server_create()
 - webrtc_signaling_server_start()
 - webrtc_signaling_server_stop()
 - webrtc_signaling_server_destroy()

[Version] 0.1.90
[Issue Type] API

Change-Id: I7719cacb9bcb1a0639b0e425067beba5fcb4a881
Signed-off-by: Sangchul Lee <sc11.lee@samsung.com>

Apply h/w resource management

- In case of TV profile, functions are disabled.

[Version] 0.1.89
[Issue Type] New feature

Change-Id: Ida0e5049b667c2cb14756d03dae25d9d757178bf
Signed-off-by: Hyunil <hyunil46.park@samsung.com>