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webrtc_m130/webrtc/call/DEPS

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Reland "Add check_deps rules in DEPS files." Relanding https://codereview.webrtc.org/1796413002/ without the change to the openmax_dl include path (which broke downstream code). TBR=tommi@webrtc.org BUG=webrtc:5623 TESTED=Passing runs using: buildtools/checkdeps/checkdeps.py --root=. talk buildtools/checkdeps/checkdeps.py --root=. webrtc Review URL: https://codereview.webrtc.org/1804333002 . Cr-Commit-Position: refs/heads/master@{#12031}
2016-03-17 12:05:36 +01:00
include_rules = [
"+webrtc/audio",
"+webrtc/base",
Moved RtcEventLog files from call/ to logging/ The RtcEventLog headers need to be accessible from any place which needs logging, and the implementation needs access to data structures that are logged. After a discussion in the code review, we all agreed to move the RtcEventLog implementation into its own top level directory - which I called "logging/" in expectation that other types of logging may have similar requirements. The directory contains two main build targets - "rtc_event_log_api", which is just rtc_event_log.h, that has no external dependencies and can be used from anywhere, and "rtc_event_log_impl" which contains the rest of the implementation and has many dependencies (more in the future). The "api" target can be referenced from anywhere, while the "impl" target is only needed at the place of instantiation (currently Call, soon to be moved to PeerConnection by https://codereview.webrtc.org/2353033005/). This change allows using RtcEventLog in the p2p/ directory, so that we can log STUN pings and ICE state transitions. BUG=webrtc:6393 R=kjellander@webrtc.org, kwiberg@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, terelius@webrtc.org Review URL: https://codereview.webrtc.org/2380683005 . Cr-Commit-Position: refs/heads/master@{#14485}
2016-10-03 18:31:22 -07:00
"+webrtc/logging/rtc_event_log",
Reland "Add check_deps rules in DEPS files." Relanding https://codereview.webrtc.org/1796413002/ without the change to the openmax_dl include path (which broke downstream code). TBR=tommi@webrtc.org BUG=webrtc:5623 TESTED=Passing runs using: buildtools/checkdeps/checkdeps.py --root=. talk buildtools/checkdeps/checkdeps.py --root=. webrtc Review URL: https://codereview.webrtc.org/1804333002 . Cr-Commit-Position: refs/heads/master@{#12031}
2016-03-17 12:05:36 +01:00
"+webrtc/modules/audio_coding",
Added an empty AudioTransportProxy to AudioState. All audio in calls is now routed through AudioTransportProxy. The AudioTransport implemented by VoEBaseImpl is disconnected from AudioDevice and replaced by an empty proxy layer that forwards calls to the old Transport. This is a refactoring CL in preparation for landing https://codereview.webrtc.org/2436033002/, which will connect the new AudioMixer. In the planned configuration, the currently empty AudioTransportProxy will query the new mixer for audio instead of polling data from the old Transport. Mixed audio will be passed to an AudioProcessing interface. AudioTransportProxy is initialized with an AudioProcessing*, which is currently unused. No presubmit since we implement an interface with non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2454373002 Cr-Commit-Position: refs/heads/master@{#15133}
2016-11-17 06:28:59 -08:00
"+webrtc/modules/audio_device",
Passed AudioMixer to AudioState::Config. This is a refactoring change in preparation for enabling AudioMixer with the goal to have a small CL as possible for passing audio through the new audio mixer in WebRTC. The dependent CL https://codereview.webrtc.org/2436033002/ enables the mixer. An object of class AudioState is shared across different webrtc audio connections. It is created in tests and in WebRTCVoiceEngine. AudioState is constructed by passing a Config struct, where one argument is scoped_refptr<AudioMixer>. Populating this field has now been mandatory. Tests and WebRTCVoiceEngine create and pass either a AudioMixerImpl. WebRTCVoiceEngine passes a real AudioMixer, which is currently unused. An alternative would have tests pass a mocked audio mixer. We chose not to do that, because we believe that tests should use the real thing unless there are reasons against it. Construction time is not an issue, because the real mixer is relatively lightweight. We couldn't find a way to test any mixer-related changes in AudioState before the mixes is connected. The next dependent CL https://codereview.webrtc.org/2436033002/ contains unit tests for mixer usage. BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2469743002 Cr-Commit-Position: refs/heads/master@{#15134}
2016-11-17 06:48:48 -08:00
"+webrtc/modules/audio_mixer",
Added an empty AudioTransportProxy to AudioState. All audio in calls is now routed through AudioTransportProxy. The AudioTransport implemented by VoEBaseImpl is disconnected from AudioDevice and replaced by an empty proxy layer that forwards calls to the old Transport. This is a refactoring CL in preparation for landing https://codereview.webrtc.org/2436033002/, which will connect the new AudioMixer. In the planned configuration, the currently empty AudioTransportProxy will query the new mixer for audio instead of polling data from the old Transport. Mixed audio will be passed to an AudioProcessing interface. AudioTransportProxy is initialized with an AudioProcessing*, which is currently unused. No presubmit since we implement an interface with non-const references. NOPRESUBMIT=True BUG=webrtc:6346 Review-Url: https://codereview.webrtc.org/2454373002 Cr-Commit-Position: refs/heads/master@{#15133}
2016-11-17 06:28:59 -08:00
"+webrtc/modules/audio_processing",
Reland "Add check_deps rules in DEPS files." Relanding https://codereview.webrtc.org/1796413002/ without the change to the openmax_dl include path (which broke downstream code). TBR=tommi@webrtc.org BUG=webrtc:5623 TESTED=Passing runs using: buildtools/checkdeps/checkdeps.py --root=. talk buildtools/checkdeps/checkdeps.py --root=. webrtc Review URL: https://codereview.webrtc.org/1804333002 . Cr-Commit-Position: refs/heads/master@{#12031}
2016-03-17 12:05:36 +01:00
"+webrtc/modules/bitrate_controller",
"+webrtc/modules/congestion_controller",
"+webrtc/modules/pacing",
"+webrtc/modules/rtp_rtcp",
"+webrtc/modules/utility",
"+webrtc/system_wrappers",
"+webrtc/voice_engine",
"+webrtc/video",
]
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