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webrtc_m130/pc
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Jonas Olsson 84df1c724e Make fewer copies when using StringBuilder. 
Replace calls to .str() which copies with .Release which moves in cases where that's safe.

This CL was generated by this command:
git grep -l 'StringBuilder' |
xargs perl -i -0 -pe "s/(rtc::StringBuilder (\S+);.*?return )\\g2.str\(\)/\$1\$2.Release\(\)/sg"

Bug: webrtc:8982
Change-Id: If4dadbeb039df010aaaa9e58da81c1971a84fe8f
Reviewed-on: https://webrtc-review.googlesource.com/100307
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24790}
2018-09-24 09:39:19 +00:00
..
test
Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
2018-09-03 09:00:01 +00:00
audiotrack.cc
…
audiotrack.h
…
BUILD.gn
Fix no_exit_time_destructors in pc.
2018-09-11 09:32:14 +00:00
builtin_video_bitrate_allocator_factory.cc
The default logic for creating video bitrate allocator.
2018-08-23 20:50:32 +00:00
builtin_video_bitrate_allocator_factory.h
The default logic for creating video bitrate allocator.
2018-08-23 20:50:32 +00:00
bundlefilter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bundlefilter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
bundlefilter.h
…
channel_unittest.cc
Cleanup networkroute.h
2018-07-23 19:35:28 +00:00
channel.cc
Remove simple stringstream usages.
2018-09-06 12:53:19 +00:00
channel.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
channelmanager_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
channelmanager.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
channelmanager.h
…
createpeerconnectionfactory.cc
…
datachannel_unittest.cc
Removing warning suppression flags from pc/.
2018-07-04 10:35:27 +00:00
datachannel.cc
Use AsyncInvoker in DataChannel instead of MessageHandler
2018-08-31 22:48:20 +00:00
datachannel.h
Use AsyncInvoker in DataChannel instead of MessageHandler
2018-08-31 22:48:20 +00:00
DEPS
…
dtlssrtptransport_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
dtlssrtptransport.cc
…
dtlssrtptransport.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
dtmfsender_unittest.cc
Add "tones remaining" argument to DTMF ontonechange callback
2018-09-07 17:29:37 +00:00
dtmfsender.cc
Add "tones remaining" argument to DTMF ontonechange callback
2018-09-07 17:29:37 +00:00
dtmfsender.h
Use AsyncInvoker in DtmfSender instead of MessageHandler
2018-08-31 22:49:26 +00:00
externalhmac.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
externalhmac.h
…
iceserverparsing_unittest.cc
Revert "Add SSLConfig object to IceServer."
2018-09-12 10:46:04 +00:00
iceserverparsing.cc
Revert "Add SSLConfig object to IceServer."
2018-09-12 10:46:04 +00:00
iceserverparsing.h
…
jsepicecandidate.cc
Add a missing #include from jsepicecandidate.cc to absl/memory/memory.h
2018-08-15 15:19:07 +00:00
jsepsessiondescription_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
jsepsessiondescription.cc
Add CreateSessionDescription overload which takes a cricket::SessionDescription
2018-07-25 18:03:05 +00:00
jseptransport_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
jseptransport.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
jseptransport.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
jseptransportcontroller_unittest.cc
Add AsyncResolverFactory interface and basic implementation.
2018-08-02 21:20:15 +00:00
jseptransportcontroller.cc
Use AsyncInvoker in JsepTransportController instead of MessageHandler
2018-08-31 22:12:45 +00:00
jseptransportcontroller.h
Use AsyncInvoker in JsepTransportController instead of MessageHandler
2018-08-31 22:12:45 +00:00
localaudiosource_unittest.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
localaudiosource.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
localaudiosource.h
…
mediaconstraintsinterface_unittest.cc
…
mediasession_unittest.cc
Make fewer copies when using StringBuilder.
2018-09-24 09:39:19 +00:00
mediasession.cc
Use C++11 for range loop in pc/mediasession.cc (and test)
2018-09-10 20:27:34 +00:00
mediasession.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastream.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
mediastreamobserver.cc
…
mediastreamobserver.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
mediastreamtrack.h
…
OWNERS
Adding shampson (me) as an owner to pc/ & api/.
2018-07-03 20:39:17 +00:00
peerconnection_bundle_unittest.cc
Add CreateIceCandidate overload which takes a cricket::Candidate
2018-07-23 23:49:44 +00:00
peerconnection_crypto_unittest.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
peerconnection_datachannel_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
peerconnection_histogram_unittest.cc
Modified peerconnection's "observer" slot to be nulled on close.
2018-08-01 09:44:57 +00:00
peerconnection_ice_unittest.cc
Remove simple stringstream usages.
2018-09-06 12:53:19 +00:00
peerconnection_integrationtest.cc
Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
2018-09-03 09:00:01 +00:00
peerconnection_jsep_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
peerconnection_media_unittest.cc
Add support for enabling simulcast in "Plan B" using MediaConstraints.
2018-08-24 09:55:59 +00:00
peerconnection_rampup_tests.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
peerconnection_rtp_unittest.cc
[Unified Plan] Implement FiredDirection for RtpTransceiver
2018-07-17 23:56:04 +00:00
peerconnection_signaling_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
peerconnection.cc
Make fewer copies when using StringBuilder.
2018-09-24 09:39:19 +00:00
peerconnection.h
Revert "Use AsyncInvoker in PeerConnection instead of MessageHandler"
2018-09-05 21:12:18 +00:00
peerconnectionendtoend_unittest.cc
Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
2018-09-03 09:00:01 +00:00
peerconnectionfactory_unittest.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
peerconnectionfactory.cc
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
peerconnectionfactory.h
Delete almost all use of MediaConstraintsInterface in the PeerConnection API
2018-08-23 07:14:37 +00:00
peerconnectioninterface_unittest.cc
Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
2018-09-03 09:00:01 +00:00
peerconnectioninternal.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
peerconnectionwrapper.cc
Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
2018-07-12 21:35:47 +00:00
peerconnectionwrapper.h
Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
2018-07-12 21:35:47 +00:00
proxy_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
remoteaudiosource.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
remoteaudiosource.h
…
rtcpmuxfilter_unittest.cc
…
rtcpmuxfilter.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
rtcpmuxfilter.h
…
rtcstats_integrationtest.cc
getStats: add kind alias for mediaType
2018-09-03 11:49:30 +00:00
rtcstatscollector_unittest.cc
getStats: add kind alias for mediaType
2018-09-03 11:49:30 +00:00
rtcstatscollector.cc
getStats: add kind alias for mediaType
2018-09-03 11:49:30 +00:00
rtcstatscollector.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
rtcstatstraversal_unittest.cc
…
rtcstatstraversal.cc
…
rtcstatstraversal.h
…
rtpmediautils_unittest.cc
…
rtpmediautils.cc
…
rtpmediautils.h
…
rtpparametersconversion_unittest.cc
Add PeerConnection GetRtpSender/ReceiverCapabilities
2018-06-28 12:40:07 +00:00
rtpparametersconversion.cc
Add PeerConnection GetRtpSender/ReceiverCapabilities
2018-06-28 12:40:07 +00:00
rtpparametersconversion.h
Add PeerConnection GetRtpSender/ReceiverCapabilities
2018-06-28 12:40:07 +00:00
rtpreceiver.cc
Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel.
2018-09-11 20:10:44 +00:00
rtpreceiver.h
Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel.
2018-09-11 20:10:44 +00:00
rtpsender.cc
Reland "Add initial support for RtpEncodingParameters max_framerate."
2018-09-13 10:06:33 +00:00
rtpsender.h
Adds integration of the FrameEncryptor/FrameDecryptor into the MediaChannel.
2018-09-11 20:10:44 +00:00
rtpsenderreceiver_unittest.cc
Reland "Add initial support for RtpEncodingParameters max_framerate."
2018-09-13 10:06:33 +00:00
rtptransceiver.cc
[Unified Plan] Implement FiredDirection for RtpTransceiver
2018-07-17 23:56:04 +00:00
rtptransceiver.h
[Unified Plan] Implement FiredDirection for RtpTransceiver
2018-07-17 23:56:04 +00:00
rtptransport_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
rtptransport.h
Delete struct webrtc::PacketTime.
2018-08-07 10:07:15 +00:00
rtptransportinternal.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
rtptransportinternaladapter.h
Remove MetricsObserverInterface.
2018-07-19 23:00:20 +00:00
rtptransporttestutil.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
sctputils_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
sctputils.h
…
sdputils.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
sdputils.h
…
sessiondescription.cc
…
sessiondescription.h
…
srtpfilter_unittest.cc
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
srtpfilter.cc
Move base64.h to the proper location.
2018-07-23 15:40:36 +00:00
srtpfilter.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
srtpsession_unittest.cc
Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
2018-07-12 21:35:47 +00:00
srtpsession.cc
Remove MetricsObserverInterface.
2018-07-19 23:00:20 +00:00
srtpsession.h
Remove MetricsObserverInterface.
2018-07-19 23:00:20 +00:00
srtptestutil.h
…
srtptransport_unittest.cc
Use absl::make_unique and absl::WrapUnique directly
2018-07-05 10:59:49 +00:00
srtptransport.cc
Move base64.h to the proper location.
2018-07-23 15:40:36 +00:00
srtptransport.h
Remove MetricsObserverInterface.
2018-07-19 23:00:20 +00:00
statscollector_unittest.cc
Reimplement rtc::ToString and rtc::FromString without streams.
2018-08-16 16:14:01 +00:00
statscollector.cc
Reimplement rtc::ToString and rtc::FromString without streams.
2018-08-16 16:14:01 +00:00
statscollector.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
streamcollection.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00
trackmediainfomap_unittest.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
trackmediainfomap.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
transportstats.cc
…
transportstats.h
…
videocapturertracksource_unittest.cc
Reimplement rtc::ToString and rtc::FromString without streams.
2018-08-16 16:14:01 +00:00
videocapturertracksource.cc
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
videocapturertracksource.h
Move sigslot to proper third_party directory
2018-07-25 14:53:33 +00:00
videotrack_unittest.cc
…
videotrack.cc
…
videotrack.h
…
videotracksource.cc
…
videotracksource.h
Replace rtc::Optional with absl::optional in pc
2018-06-19 20:55:07 +00:00
webrtcsdp_unittest.cc
Make HasAttribute handle partial matching of attribute names.
2018-09-10 08:43:52 +00:00
webrtcsdp.cc
Make HasAttribute handle partial matching of attribute names.
2018-09-10 08:43:52 +00:00
webrtcsdp.h
…
webrtcsessiondescriptionfactory.cc
Reland "Delete leftover includes and declarations for MediaConstraintsInterface"
2018-09-03 09:00:01 +00:00
webrtcsessiondescriptionfactory.h
Reformat the WebRTC code base
2018-06-19 14:00:39 +00:00