In order to eliminate the WebRTC Subtree mirror in Chromium,
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org
Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
I used a command like this to update the paths:
perl -pi -e "s/webrtc\/base/webrtc\/rtc_base/g" `find webrtc/rtc_base -name "*.cc" -o -name "*.h"`
The only manual edit is to add an include of webrtc/rtc_base/checks.h in
webrtc/modules/audio_device/android/opensles_common.h, which likely
was needed due to changed include paths due to 'git cl format'.
BUG=webrtc:7634
NOTRY=True
NOPRESUBMIT=True
Review-Url: https://codereview.webrtc.org/2969653002
Cr-Commit-Position: refs/heads/master@{#18871}
RTP packets can be padded with extra data at the end of the payload. The usable
payload length of the packet should then be reduced with the padding length,
since the padding must be discarded. This was not the case; instead, the entire
payload, including padding data, was forwarded to the audio channel and in the
end to the decoder.
A special case of padding is packets which are empty except for the padding.
That is, they carry no usable payload. These packets are sometimes used for
probing the network and were discarded in
RTPReceiverAudio::ParseAudioCodecSpecific. The result is that NetEq never sees
those empty packets, just the holes in the sequence number series; this can
throw off the target buffer calculations.
With this change, the empty (after removing the padding) packets are let through,
all the way down to NetEq, to a new method called NetEq::InsertEmptyPacket. This
method notifies the DelayManager that an empty packet was received.
BUG=webrtc:7610, webrtc:7625
Review-Url: https://codereview.webrtc.org/2870043003
Cr-Commit-Position: refs/heads/master@{#18083}
With this change, the calculations inside AverageIAT are changed to be
in double-precision floating point instead of in fixed point. Also,
the method's name is changed to EstimatedClockDriftPpm to better
reflect what it returns.
A few unit tests had to be updated because of minor numerical
differences.
Also removing the UBSan suppression related to this issue.
BUG=webrtc:5889
Review-Url: https://codereview.webrtc.org/2408653002
Cr-Commit-Position: refs/heads/master@{#14628}
NetEqDecoder is still used in the external interfaces, but this change
opens up the ability to use SdpAudioFormats directly, once appropriate
interfaces have been added.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2355503002
Cr-Commit-Position: refs/heads/master@{#14368}
This change replaces packet_iat_count_ms_ and max_timer_ms_, two
time-counting member variables in DelayManager, with Stopwatch objects
obtained from a TickTimer.
BUG=webrtc:5608
Review-Url: https://codereview.webrtc.org/1929863002
Cr-Commit-Position: refs/heads/master@{#12554}
We must remove dependency on Chromium, i.e. we can't use Chromium's base/logging.h. That means we need to define these macros in WebRTC also when doing Chromium builds. And this causes redefinition.
* DISALLOW_ASSIGN -> RTC_DISALLOW_ASSIGN
* DISALLOW_COPY_AND_ASSIGN -> RTC_DISALLOW_COPY_AND_ASSIGN
* DISALLOW_IMPLICIT_CONSTRUCTORS -> RTC_DISALLOW_IMPLICIT_CONSTRUCTORS
Related CL: https://codereview.webrtc.org/1335923002/
BUG=chromium:468375
NOTRY=true
Review URL: https://codereview.webrtc.org/1345433002
Cr-Commit-Position: refs/heads/master@{#9953}
