We use Optional in our public API, so its header should be in
webrtc/api/.
BUG=webrtc:8205
Review-Url: https://codereview.webrtc.org/3011943002
Cr-Commit-Position: refs/heads/master@{#19693}
We use ArrayView in our public API, so its header should be in
webrtc/api/.
BUG=none
Review-Url: https://codereview.webrtc.org/3007763002
Cr-Commit-Position: refs/heads/master@{#19658}
Receive logic in voe::Channel attempted to handle RTX if
RTPPayloadRegistry::IsRtx returns true, but the audio code never calls
the config methods (SetRtxSsrc or SetRtxPayloadType) required for that
to ever happen.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/3006913002
Cr-Commit-Position: refs/heads/master@{#19633}
directives in our DEPS files are not needed anymore.
Includes from webrtc/rtc_base are also whitelisted in webrtc/DEPS
so we don't have to whitelist it in all the others DEPS files.
BUG=webrtc:7634
NOTRY=True
Review-Url: https://codereview.webrtc.org/3006583002
Cr-Commit-Position: refs/heads/master@{#19601}
The following fields are logged:
- error, if there was an error.
- elapsed time in milliseconds
BUG=webrtc:8100
Review-Url: https://codereview.webrtc.org/2996933003
Cr-Commit-Position: refs/heads/master@{#19574}
Instead explicitly ignore only the flags we know should be ignored.
BUG=webrtc:7568
Review-Url: https://codereview.webrtc.org/2968003003
Cr-Commit-Position: refs/heads/master@{#19412}
That way, the debug printout will tell us which of x and y that was false.
BUG=none
Review-Url: https://codereview.webrtc.org/2988153003
Cr-Commit-Position: refs/heads/master@{#19297}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2976293002
Cr-Commit-Position: refs/heads/master@{#19094}
All downstream code have been updated to the new location.
In PRESUBMIT.py:
* Remove webrtc/rtc_base from CPP_BLACKLIST
* Add webrtc/rtc_base to LEGACY_API_DIRS
Fix some duplicated paths in
webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
BUG=webrtc:7634
TBR=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2973183002
Cr-Commit-Position: refs/heads/master@{#18948}
This CL removes code that supported the now removed
downstream dependencies in the support for using an
external audio processing module.
BUG=webrtc:7939
Review-Url: https://codereview.webrtc.org/2969213002
Cr-Commit-Position: refs/heads/master@{#18929}
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"`
BUG=webrtc:7634
NOPRESUBMIT=True # cpplint errors that aren't caused by this CL.
Review-Url: https://codereview.webrtc.org/2969623003
Cr-Commit-Position: refs/heads/master@{#18870}
[This CL is a rebase of an original CL by solenberg@:
https://codereview.webrtc.org/2948763002/ which in turn was a
rebase of an original CL by peah@:
https://chromium-review.googlesource.com/c/527032/]
Allow an external audio processing module to be used in WebRTC
This CL adds support for optionally using an externally created audio
processing module in a peerconnection. The ownership is shared
between the peerconnection and the external creator of the module.
As part of this the internal ownership of the audio processing module
is moved from VoiceEngine to WebRtcVoiceEngine.
BUG=webrtc:7775
Review-Url: https://codereview.webrtc.org/2961723004
Cr-Commit-Position: refs/heads/master@{#18837}
And implementation class RtpStreamReceiverController.
It's responsible for demuxing, and acts as factory for
RtpStreamReceiverInterface.
BUG=webrtc:7135
Review-Url: https://codereview.webrtc.org/2886993005
Cr-Commit-Position: refs/heads/master@{#18696}
const int16_t* data() const;
int16_t* mutable_data();
- data() returns a zeroed static buffer on muted frames (to avoid unnecessary zeroing of the member buffer) and directly returns AudioFrame::data_ on unmuted frames.
- mutable_data(), lazily zeroes AudioFrame::data_ if the frame is currently muted, sets muted=false, and returns AudioFrame::data_.
These accessors serve to "force" callers to be aware of the mute state field, i.e. lazy zeroing is not the primary motivation.
This change only optimizes handling of muted frames where it is somewhat trivial to do so. Other improvements requiring more significant structural changes will come later.
BUG=webrtc:7343
TBR=henrika
Review-Url: https://codereview.webrtc.org/2750783004
Cr-Commit-Position: refs/heads/master@{#18543}
Specifically, just like SafeMin() and SafeMax() it handles all
combinations of integer and all
combinations of floating-point arguments by picking a
result type that is guaranteed to be able to hold the result.
This CL also replaces a bunch of std::min + std:max call pairs with
calls to SafeClamp()---the ones that could easily be found by grep
because "min" and "max" were on the same line. :-)
BUG=webrtc:7459
Review-Url: https://codereview.webrtc.org/2808513003
Cr-Commit-Position: refs/heads/master@{#18542}
When using the SetEncoder interface, there's no actual CodecInst to return from Channel::GetSendCodec. Before this CL, this was done by calling the ACM, which has functionality for generating a CodecInst with the necessary values even when handed an external encoder. Unfortunately, this call takes a lock and does some extra processing which isn't strictly necessary in this case. Since GetSendCodec is called inside the audio input callback code, this can cause problems.
This CL instead generates a CodecInst in the SetEncoder call and has GetSendCodec simply return that one if it's available. If it isn't the value from codec_manager_ is returned instead, as was the case before injectable audio codec related changes were added to Channel.
BUG=b/38018041
Review-Url: https://codereview.webrtc.org/2924363004
Cr-Commit-Position: refs/heads/master@{#18515}
Biggest change is to Remove MediaType as argument to RtcEventLog::LogRtpHeader and RtcEventLog::LogRtcpHeader.
Since the type is used by tools, these tools are rewritten to figure out the media type from the configurations instead.
BUG=webrtc:7538
TBR=solenberg@webrtc.org // For call.cc and voiceengine.cc
Review-Url: https://codereview.webrtc.org/2855143002
Cr-Commit-Position: refs/heads/master@{#18324}
Plumbed AudioEncoderFactory up into CreatePeerConnectionFactory.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2799033006
Cr-Commit-Position: refs/heads/master@{#17977}
Reason for revert:
Fourth attempt to land.
Waiting for https://codereview.webrtc.org/2845013003 to
avoid conflicts on webrtc/modules/audio_coding:neteq_unittest_tools.
Original issue's description:
> Revert of Enable GN check for webrtc/base (patchset #13 id:240001 of https://codereview.webrtc.org/2717083002/ )
>
> Reason for revert:
> Breaks Chromium because in Chromium we import WebRTC with rtc_include_tests=false (https://bugs.chromium.org/p/chromium/issues/detail?id=713179#c6).
>
> Chromium uses webrtc/test/fuzzers and this CL adds test dependencies to neteq_rtc_fuzzer.
>
> Original issue's description:
> > Enable GN check for webrtc/base
> >
> > It's not possible to enable it for the rtc_base_approved
> > target but since a larger refactoring is ongoing for webrtc/base
> > this CL doesn't attempt to fix that.
> >
> > Changes made:
> > * Move webrtc/system_wrappers/include/stringize_macros.h into
> > webrtc/base:rtc_base_approved_unittests (and corresponding
> > unit test to rtc_base_approved_unittests).
> > * Move md5digest.* from rtc_base_approved to rtc_base_test_utils target.
> > * Move webrtc/system_wrappers/include/stringize_macros.h (+test) into
> > webrtc/base.
> > * Remove unused use include of webrtc/base/fileutils.h in
> > webrtc/base/pathutils.cc
> >
> > BUG=webrtc:6828, webrtc:3806, webrtc:7480
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2717083002
> > Cr-Commit-Position: refs/heads/master@{#17766}
> > Committed: ed754e71ae
>
> TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:6828, webrtc:3806, webrtc:7480
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2838683002
> Cr-Commit-Position: refs/heads/master@{#17849}
> Committed: 11ed366c48TBR=perkj@webrtc.org,tommi@webrtc.org,nisse@webrtc.org,kjellander@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:6828, webrtc:3806, webrtc:7480
Review-Url: https://codereview.webrtc.org/2852663002
Cr-Commit-Position: refs/heads/master@{#17927}
These are the changes made to WebRtcVoiceEngine and surrounding
code. It still contains some things that are inelegant, like how
AudioCodecSpec and AudioFormatInfo is ferried around in
SendCodecSpec. This should probably be resolved before landing.
There are also a few test still that are disabled. They should be
removed or fixed, as the case may be.
I've put this CL up to get a better overview of the changes made and
how reviewable they are.
BUG=webrtc:5806
Review-Url: https://codereview.webrtc.org/2705093002
Cr-Commit-Position: refs/heads/master@{#17904}
