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| author | Sebastian Jansson <srte@webrtc.org> | 2018-02-20 19:38:37 +0100 |
|---|---|---|
| committer | Commit Bot <commit-bot@chromium.org> | 2018-02-21 12:33:02 +0000 |
| commit | df023aa6b434e6845fa06a0dc3be0d6268658063 (patch) | |
| tree | 072e43a9d87d0dcb820e2199a82fe4464e21b409 /call/BUILD.gn | |
| parent | fc8d26bd8ad627a4469bbe2ca6e343ae9fe4e537 (diff) | |
| download | webrtc-df023aa6b434e6845fa06a0dc3be0d6268658063.tar.gz | |
Extracted bitrate configuration from call class.
This separates the bitrate configuration logic from other call specific
logic, creating a greater separation of concern and simplifying testing.
The old call tests are kept but can be removed in the future reducing
the dependencies on rtp transport control interface and congestion
control in the system, which will simplify future refactoring.
This also prepares for moving the bitrate configuration responsibility
to the rtp transport controller in a later CL.
Bug: webrtc:8415
Change-Id: I97126e89f30b63fc9b5d98a0bed1c29f18a6ed44
Reviewed-on: https://webrtc-review.googlesource.com/54401
Reviewed-by: Zach Stein <zstein@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22124}
Diffstat (limited to 'call/BUILD.gn')
| -rw-r--r-- | call/BUILD.gn | 15 |
1 files changed, 15 insertions, 0 deletions
diff --git a/call/BUILD.gn b/call/BUILD.gn index a3d23164ef..caa17b2bb1 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn @@ -100,6 +100,18 @@ rtc_source_set("rtp_sender") { ] } +rtc_source_set("bitrate_configurator") { + sources = [ + "rtp_bitrate_configurator.cc", + "rtp_bitrate_configurator.h", + ] + deps = [ + ":rtp_interfaces", + "../rtc_base:checks", + "../rtc_base:rtc_base_approved", + ] +} + rtc_source_set("bitrate_allocator") { sources = [ "bitrate_allocator.cc", @@ -135,6 +147,7 @@ rtc_static_library("call") { deps = [ ":bitrate_allocator", + ":bitrate_configurator", ":call_interfaces", ":rtp_interfaces", ":rtp_receiver", @@ -201,12 +214,14 @@ if (rtc_include_tests) { "call_unittest.cc", "flexfec_receive_stream_unittest.cc", "rtcp_demuxer_unittest.cc", + "rtp_bitrate_configurator_unittest.cc", "rtp_demuxer_unittest.cc", "rtp_rtcp_demuxer_helper_unittest.cc", "rtx_receive_stream_unittest.cc", ] deps = [ ":bitrate_allocator", + ":bitrate_configurator", ":call", ":call_interfaces", ":mock_rtp_interfaces", |