Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h"
Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class
that only handles SRTP configuration to a more generic structure that can be
used and extended for all per peer connection CryptoOptions that can be on a
given PeerConnection.
Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be
accessed as crypto_options.srtp.whatever_option_name. This is more inline with
other structures we have in WebRTC such as VideoConfig. As additional features
are added over time this will allow the structure to remain compartmentalized
and concerned components can only request a subset of the overall configuration
structure e.g:
void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config);
In addition to this it made little sense for sslstreamadapter.h to hold all
Srtp related configuration options. The header has become loo large and takes on
too many responsibilities and spilting this up will lead to more maintainable
code going forward.
This will be used in a future CL to enable configuration options for the newly
supported Frame Crypto.
Reland Fix:
- cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional
root level configuration.
- peerconnectionfactory - If this optional is set will now overwrite the
underyling value.
This along with the other field will be deprecated once dependent projects
are updated.
TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org
Bug: webrtc:9681
Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d
Reviewed-on: https://webrtc-review.googlesource.com/c/105560
Reviewed-by: Benjamin Wright <benwright@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25135}
diff --git a/pc/channelmanager.cc b/pc/channelmanager.cc
index dab622a..1c80719 100644
--- a/pc/channelmanager.cc
+++ b/pc/channelmanager.cc
@@ -159,7 +159,7 @@
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
const AudioOptions& options) {
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<VoiceChannel*>(RTC_FROM_HERE, [&] {
@@ -226,7 +226,7 @@
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
- const rtc::CryptoOptions& crypto_options,
+ const webrtc::CryptoOptions& crypto_options,
const VideoOptions& options) {
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<VideoChannel*>(RTC_FROM_HERE, [&] {
@@ -291,7 +291,7 @@
rtc::Thread* signaling_thread,
const std::string& content_name,
bool srtp_required,
- const rtc::CryptoOptions& crypto_options) {
+ const webrtc::CryptoOptions& crypto_options) {
if (!worker_thread_->IsCurrent()) {
return worker_thread_->Invoke<RtpDataChannel*>(RTC_FROM_HERE, [&] {
return CreateRtpDataChannel(media_config, rtp_transport, signaling_thread,