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Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"

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The breaking tests in Chromium have been temporarily disabled, they will be fixed and reenabled soon.

Original CLs: https://codereview.webrtc.org/1748403002/, https://codereview.webrtc.org/2107253002/ and https://codereview.webrtc.org/2106103003/.

TBR=solenberg@webrtc.org,tommi@webrtc.org,stefan@webrtc.org,terelius@webrtc.org,tkchin@webrtc.org
BUG=webrtc:4741, webrtc:5603, chromium:609749

Review-Url: https://codereview.webrtc.org/2110113003
Cr-Commit-Position: refs/heads/master@{#13379}
This commit is contained in:
ivoc
2016-07-04 07:06:55 -07:00
committed by Commit bot
parent 77ad394fa6
commit 14d5dbe5b3
61 changed files with 428 additions and 359 deletions
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9
webrtc/modules/bitrate_controller/include/bitrate_controller.h
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@ -55,8 +55,11 @@ class BitrateController : public Module {
// TODO(perkj): BitrateObserver has been deprecated and is not used in WebRTC.
// Remove this method once other other projects does not use it.
static BitrateController* CreateBitrateController(Clock* clock,
BitrateObserver* observer);
static BitrateController* CreateBitrateController(Clock* clock);
BitrateObserver* observer,
RtcEventLog* event_log);
static BitrateController* CreateBitrateController(Clock* clock,
RtcEventLog* event_log);
virtual ~BitrateController() {}
@ -76,8 +79,6 @@ class BitrateController : public Module {
virtual void UpdateDelayBasedEstimate(uint32_t bitrate_bps) = 0;
virtual void SetEventLog(RtcEventLog* event_log) = 0;
// Gets the available payload bandwidth in bits per second. Note that
// this bandwidth excludes packet headers.
virtual bool AvailableBandwidth(uint32_t* bandwidth) const = 0;