Calling ModuleRtpRtcpImpl::SetSendingStatus or
ModuleRtpRtcpImpl::SetSendingMediaStatus doesn't change the
current sequence number, which is part of the RTPSender state.
Bug: webrtc:2111
Change-Id: I432d4daa7ab4cc25d87b90c8b1b255359ffbe3a3
Reviewed-on: https://webrtc-review.googlesource.com/c/120401
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26513}
This CL applies clang-tidy's modernize-use-override [1] to the
WebRTC codebase.
All changes in this CL are automatically generated by both clang-tidy
and 'git cl format'.
[1] - https://clang.llvm.org/extra/clang-tidy/checks/modernize-use-override.html
Bug: webrtc:10252
Change-Id: I2bb8bd90fa8adb90aa33861fe7c788132a819a20
Reviewed-on: https://webrtc-review.googlesource.com/c/120412
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26461}
Rids can now be sent using rtp_sender.
Hooking up the rid values in the voice and video engine is still WIP.
Bug: webrtc:10074
Change-Id: I245c7ecb23b67fc0ba65caaa5dbb4fcfd60c81bb
Reviewed-on: https://webrtc-review.googlesource.com/c/114505
Commit-Queue: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26092}
The following APIs on AudioCodingModule are deprecated with this CL:
static int NumberOfCodecs();
static int Codec(int, CodecInst*);
static int Codec(const char*, CodecInst*, int, size_t);
static int Codec(const char*, int, size_t);
absl::optional<CodecInst> SendCodec() const;
bool RegisterReceiveCodec(int, const SdpAudioFormat&);
int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&);
int UnregisterReceiveCodec(uint8_t);
int32_t ReceiveCodec(CodecInst*);
absl::optional<SdpAudioFormat> ReceiveFormat();
As well as this method on RtpRtcp module:
int32_t RegisterSendPayload(const CodecInst&);
Bug: webrtc:7626
Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce
Reviewed-on: https://webrtc-review.googlesource.com/c/113816
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26025}
This will be used in a later CL to use the link capacity field in the
update to control the Opus encoder.
Bug: webrtc:9718
Change-Id: If2ad16a8f4656e8cdf10c33f5fb060ef7ca5caba
Reviewed-on: https://webrtc-review.googlesource.com/c/111510
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25761}
This is a follow up of a comment in
https://webrtc-review.googlesource.com/c/src/+/110105
It was not very useful to split the audio and video report interval since the RTCP module can only either be audio or video.
The recent it was written that way in https://webrtc-review.googlesource.com/c/src/+/43201/ was because that was a straightforward transition from two global constants to two variable.
Bug: webrtc:8789
Change-Id: I2293de14ba5f363351f379a02022ed5dc7b8d458
Reviewed-on: https://webrtc-review.googlesource.com/c/110824
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25741}
Replaced by interface ChannelSendInterface, implemented by ChannelSend
and mock class.
Thread checkers are moved to ChannelSend, which is also moved into
the anonymous namespace and exposed only via a function CreateChannelSend.
Bug: webrtc:9801
Change-Id: I73b2e2bfb67c1a5077709f2379533bf315babad9
Reviewed-on: https://webrtc-review.googlesource.com/c/111240
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25684}
So far ANA was not available for media transport interface. With recent changes to media transport, we can now account for packet overhead, network route (ip/tcp/udp/turn overheads) and we can also use bandwidth estimate from the media transport.
Bug: webrtc:9719
Change-Id: I98c9a09dd418b763c339ee2ee05592e164cf9199
Reviewed-on: https://webrtc-review.googlesource.com/c/110367
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25677}
'configured_bitrate_bps_' is accessed from different threads in
SetBitrate and GetBitrate (one comes back from OnNetworkRouteChange
callback, the other one is used in GetStats()) and so it should be
protected by a critical section.
Bug: webrtc:10010
Change-Id: I029baa729e0203b9f2d180d8835d61add26e6cef
Reviewed-on: https://webrtc-review.googlesource.com/c/111281
Commit-Queue: Peter Slatala <psla@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25675}
This is a preparation for deleting ChannelSendProxy. Signature is
changed on a couple of methods. Unused methods
EnableAudioNetworkAdaptor, DisableAudioNetworkAdaptor,
SetReceiverFrameLengthRange and RtpRtcpModulePtr are deleted. Some
methods are demoted to private: SendData, SendRtp, SendRtcp,
PreferredSampleRate, Sending, and OnOverheadChanged.
Bug: webrtc:9801
Change-Id: I982e72418a32e66fb5de410350b1bfebd9a3219c
Reviewed-on: https://webrtc-review.googlesource.com/c/110605
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25666}
This is a follow up of https://webrtc-review.googlesource.com/c/src/+/43201.
Issue 43201 didn't do the job properly.
1. The audio rtcp report interval is not properly hooked up.
2. We don't need to propagate audio rtcp interval into video send stream or vice versa.
3. We don't need to propagate rtcp report interval to any receiving streams.
Bug: webrtc:8789
Change-Id: I1f637d6e5173608564ef0702d7eda6fc93b3200f
Reviewed-on: https://webrtc-review.googlesource.com/c/110105
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#25610}
This reverts commit 9a0662ac7e.
Reason for revert: breaks some av sync perf tests
Original change's description:
> Use only first payload timestamp for RTCP SR generation for audio
>
> Since now RTP rate is set correctly for audio, there's no need to
> use the very last data packet rtp/capture timestamps for generating
> RTCP SR packets.
>
> Using only one (first) packet timestamp eliminates the jitter between
> rtp and capture timestamps for audio. This jitter comes from the fact
> that capture timestamp for audio is unknown and we generate bogus
> timestamp at arbitrary, non-constant offset from the real capture time.
>
> Bug: webrtc:9905
> Change-Id: I855556184cfe994be39ab7780836a050f5a38c35
> Reviewed-on: https://webrtc-review.googlesource.com/c/108580
> Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25430}
TBR=danilchap@webrtc.org,ilnik@webrtc.org,ossu@webrtc.org
Change-Id: I208a659379b1075258ee94613e42afd9aebe4754
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9905
Reviewed-on: https://webrtc-review.googlesource.com/c/108623
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25435}
Since now RTP rate is set correctly for audio, there's no need to
use the very last data packet rtp/capture timestamps for generating
RTCP SR packets.
Using only one (first) packet timestamp eliminates the jitter between
rtp and capture timestamps for audio. This jitter comes from the fact
that capture timestamp for audio is unknown and we generate bogus
timestamp at arbitrary, non-constant offset from the real capture time.
Bug: webrtc:9905
Change-Id: I855556184cfe994be39ab7780836a050f5a38c35
Reviewed-on: https://webrtc-review.googlesource.com/c/108580
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25430}
This change corrects a potential race condition when updating a FrameEncryptor
for the audio send channel. If a FrameEncryptor is set on an active audio
stream it is possible for the current FrameEncryptor attached to the audio channel to be deallocated due to
the FrameEncryptors reference count reaching zero before the new FrameEncryptor is set on the
channel.
To address this issue the ChannelSend is now holds a scoped_reftptr<FrameEncryptor>
to only allow deallocation when it is actually set on the encoder queue.
ChannelSend is unique in this respect as the Audio Receiver a long with the
Video Sender and Video Receiver streams all recreate themselves when they have
a configuration change. ChannelSend instead reconfigures itself using the
existing channel object.
Added Seth as TBR as this only introduces mocks.
TBR=shampson@webrtc.org
Bug: webrtc:9907
Change-Id: Ibf391dc9cecdbed1874e0252ff5c2cb92a5c64f4
Reviewed-on: https://webrtc-review.googlesource.com/c/107664
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25374}
This change adds a new subcategory to the public native webrtc::CryptoOptions
structure: webrtc::CryptoOptions::Frame.
This new structure has a single off by default property:
crypto_options.frame.require_frame_encryption.
This new flag if set prevents RtpSenders from sending outgoing payloads unless
a frame_encryptor_ is attached and prevents RtpReceivers from receiving
incoming payloads unless a frame_decryptor_ is attached.
This option is important to enforce no unencrypted data can ever leave the
device or be received.
I have also attached bindings for Java and Objective-C.
I have implemented this functionality for E2EE audio but not E2EE video
since the changes are still in review.
Bug: webrtc:9681
Change-Id: Ie184711190e0cdf5ac781f69e9489ceec904736f
Reviewed-on: https://webrtc-review.googlesource.com/c/105540
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25238}
This change integrates the FrameDecryptorInterface and the FrameEncryptorInterface into
the audio media path. If a FrameEncryptorInterface is set on an outgoing audio RTPSender
then each outgoing audio payload will first pass through the provided FrameEncryptor which
will have a chance to modify the payload contents for the purposes of encryption.
If a FrameDecryptorInterface is set on an incoming audio RtpReceiver then each incoming
audio payload will first pass through the provided FrameDecryptor which have a chance to
modify the payload contents for the purpose of decryption.
While AEAD is supported by the FrameDecryptor/FrameEncryptor interfaces this CL does not
use it and so it is left as null.
Bug: webrtc:9681
Change-Id: Ic383a9dce280528739f9d271357c2220e0a0dccf
Reviewed-on: https://webrtc-review.googlesource.com/c/101702
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25001}
