18 Commits

Author SHA1 Message Date
Niels Möller
5304a32a94 Delete StreamStatistician::IsRetransmitOfOldPacket
Return value always passed as the |retransmitted| argument to
ReceiveStatistics::IncomingPacket. The implementation of this method,
StreamStatisticianImpl::IncomingPacket, can call its own
IsRetransmitOfOldPacket, which is demoted to a private method.

Bug: webrtc:7135
Change-Id: I904db676738689c7a1db4caa588f70e64e3c357d
Reviewed-on: https://webrtc-review.googlesource.com/95649
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24494}
2018-08-30 11:00:13 +00:00
Niels Möller
fa2b2d62d7 Delete use of RtpPayloadRegistry.
Use in voe::Channel replaced by a std::map storing payload type frequencies.
This is a followup to
https://webrtc-review.googlesource.com/c/src/+/93820.

Bug: webrtc:7135
Change-Id: I874b706aee19fdc2d841db42a540e4f7aa2725f1
Reviewed-on: https://webrtc-review.googlesource.com/94508
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24315}
2018-08-16 14:44:25 +00:00
Niels Möller
30b4839d10 Refactor voe::Channel to not use RtpReceiver.
Analogous to https://webrtc-review.googlesource.com/c/src/+/92398, for
RtpVideoStreamReceiver.

Bug: webrtc:7135
Change-Id: I0639f9982da2ed80edbcf900cf14f8ae982ef80c
Reviewed-on: https://webrtc-review.googlesource.com/93820
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24309}
2018-08-16 10:18:20 +00:00
Sebastian Jansson
9701e0ce2f Makes treatment of received reports of packets lost signed.
Bug: webrtc:9598
Change-Id: I0f6ffe348585b8ec69753089652812da516d33d8
Reviewed-on: https://webrtc-review.googlesource.com/93021
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24291}
2018-08-15 14:27:23 +00:00
Niels Möller
fa4e185684 Delete class voe::RtcEventLogProxy
Bug: None
Change-Id: Ic0c380e2f7f844a0e06c8c2a3d8bcb42ecee1eba
Reviewed-on: https://webrtc-review.googlesource.com/94040
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24287}
2018-08-15 09:59:15 +00:00
Niels Möller
848d6d300e Change Channel::GetRtpRtcp to return only RtpRtcp, not RtpReceiver.
For use in AudiReceiveStream, introduce a new method GetSyncInfo. This
change is analogous to https://webrtc-review.googlesource.com/91123,
doing the same for RtpVideoStreamReceiver. It's a preparation for
bypassing the RtpReceiver class.

Bug: webrtc:7135
Change-Id: I87c1c6f0a1f28b0baebe07c4181f6f0427afa314
Reviewed-on: https://webrtc-review.googlesource.com/93022
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24228}
2018-08-08 11:45:21 +00:00
Niels Möller
ab4a530b87 Delete telephone-event handling from RTPReceiverAudio.
Bug: webrtc:7135
Change-Id: Ic8b96f44ba25ff9265570dd43d3c76ed0177abfb
Reviewed-on: https://webrtc-review.googlesource.com/91125
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24172}
2018-08-02 12:55:40 +00:00
Danil Chapovalov
64b17c2aca Remove StreamStatistician::IsPacketInOrder
this function is now only used in combination with StreamStatistician::IsRetransmitOfOldPacket
but IsRetransmitOfOldPacket internally checks if packet is in_order, thus making extra check unnecessary

In addition to making code simpler, removing this checks avoids
taking two extra CritSection on common code path of incoming rtp packet.

Bug: webrtc:8016
Change-Id: I050004e256b5698ce700e3416aa86b55f446a270
Reviewed-on: https://webrtc-review.googlesource.com/85361
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23762}
2018-06-28 08:44:40 +00:00
Danil Chapovalov
b9b146c9fe Replace rtc::Optional with absl::optional in audio, call and video
This is a no-op change because rtc::Optional is an alias to absl::optional

This CL generated by running script with parameters 'audio call video':
#!/bin/bash
find $@ -type f \( -name \*.h -o -name \*.cc \) \
-exec sed -i 's|rtc::Optional|absl::optional|g' {} \+ \
-exec sed -i 's|rtc::nullopt|absl::nullopt|g' {} \+ \
-exec sed -i 's|#include "api/optional.h"|#include "absl/types/optional.h"|' {} \+

find $@ -type f -name BUILD.gn \
-exec sed -r -i 's|"(../)*api:optional"|"//third_party/abseil-cpp/absl/types:optional"|' {} \+;

git cl format

Bug: webrtc:9078
Change-Id: I02c5db956846a88a268a300ba086703a02d62e36
Reviewed-on: https://webrtc-review.googlesource.com/83722
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23628}
2018-06-15 12:09:49 +00:00
Niels Möller
f782492948 Delete RtpFeedback. The ssrc for a receive stream should be known at
configuration time.

Bug: webrtc:8995
Change-Id: I3d63a76e472a8948c98c98450e96d3301fa2688b
Reviewed-on: https://webrtc-review.googlesource.com/78701
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23409}
2018-05-28 11:05:19 +00:00
Niels Möller
ef99888bca Delete OnIncomingCSRCChanged and related code.
Bug: webrtc:8995
Change-Id: I1987d1527cce5a0c315b2d15cfffa76e7343b1fa
Reviewed-on: https://webrtc-review.googlesource.com/64220
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22626}
2018-03-27 13:18:35 +00:00
Steve Anton
bb50ce5bb6 Wire up MID send value to the PeerConnection API
Bug: webrtc:4050
Change-Id: I522cf8621e2cb639f54be2402174befd23e4af59
Reviewed-on: https://webrtc-review.googlesource.com/60962
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22610}
2018-03-26 18:14:30 +00:00
Tommi
5f22365dd7 Remove unnecessary proxy+lock code around RtcpRttStats pointer
Change-Id: I9c7fdc695be1e424488fa46962d459c66cf4d1e7
Bug: webrtc:9068
Reviewed-on: https://webrtc-review.googlesource.com/64721
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22603}
2018-03-26 12:49:00 +00:00
Niels Möller
9cfb18c5b3 Delete obsolete method RtpFeedback::OnInitializeDecoder.
Bug: None
Change-Id: I55e01e5ff1c54c76c43b378414a31fc43c9aa444
Reviewed-on: https://webrtc-review.googlesource.com/62142
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22561}
2018-03-22 12:06:54 +00:00
Karl Wiberg
08126349f5 Pass a real audio codec pair ID to decoders that we create
Bug: webrtc:8941
Change-Id: Ic2aed2ca759eb378164f3f65465e23fd7c13a9f8
Reviewed-on: https://webrtc-review.googlesource.com/63261
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22538}
2018-03-21 13:55:18 +00:00
Niels Möller
1e06289cdb Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.

Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
2018-02-07 10:07:28 +00:00
Niels Möller
f120cba82d Delete AudioMonitor and related code.
Bug: webrtc:8760
Change-Id: I0b11ec66b0f2576f52866864ba046191034a4d2d
Reviewed-on: https://webrtc-review.googlesource.com/39003
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Noah Richards <noahric@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21801}
2018-01-30 09:48:29 +00:00
Fredrik Solenberg
a8b7c7f4c6 Move remaining traces of VoiceEngine
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
  utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.

NOPRESUBMIT=true

Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 13:27:47 +00:00