| Index: webrtc/api/webrtcsession.cc
|
| diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc
|
| index 2c414a9761fca2d9d7e001e71f2a99076763615e..57f80c7d133a35a8952236b7808a932df94eeb00 100644
|
| --- a/webrtc/api/webrtcsession.cc
|
| +++ b/webrtc/api/webrtcsession.cc
|
| @@ -19,7 +19,6 @@
|
|
|
| #include "webrtc/api/jsepicecandidate.h"
|
| #include "webrtc/api/jsepsessiondescription.h"
|
| -#include "webrtc/api/mediaconstraintsinterface.h"
|
| #include "webrtc/api/peerconnectioninterface.h"
|
| #include "webrtc/api/sctputils.h"
|
| #include "webrtc/api/webrtcsessiondescriptionfactory.h"
|
| @@ -422,25 +421,6 @@ static std::string MakeTdErrorString(const std::string& desc) {
|
| return MakeErrorString(kPushDownTDFailed, desc);
|
| }
|
|
|
| -// Set |option| to the highest-priority value of |key| in the optional
|
| -// constraints if the key is found and has a valid value.
|
| -template <typename T>
|
| -static void SetOptionFromOptionalConstraint(
|
| - const MediaConstraintsInterface* constraints,
|
| - const std::string& key,
|
| - rtc::Optional<T>* option) {
|
| - if (!constraints) {
|
| - return;
|
| - }
|
| - std::string string_value;
|
| - T value;
|
| - if (constraints->GetOptional().FindFirst(key, &string_value)) {
|
| - if (rtc::FromString(string_value, &value)) {
|
| - *option = rtc::Optional<T>(value);
|
| - }
|
| - }
|
| -}
|
| -
|
| uint32_t ConvertIceTransportTypeToCandidateFilter(
|
| PeerConnectionInterface::IceTransportsType type) {
|
| switch (type) {
|
| @@ -546,7 +526,6 @@ WebRtcSession::~WebRtcSession() {
|
|
|
| bool WebRtcSession::Initialize(
|
| const PeerConnectionFactoryInterface::Options& options,
|
| - const MediaConstraintsInterface* constraints,
|
| rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
|
| const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
|
| bundle_policy_ = rtc_configuration.bundle_policy;
|
| @@ -564,46 +543,33 @@ bool WebRtcSession::Initialize(
|
|
|
| SetIceConfig(ParseIceConfig(rtc_configuration));
|
|
|
| - // TODO(perkj): Take |constraints| into consideration. Return false if not all
|
| - // mandatory constraints can be fulfilled. Note that |constraints|
|
| - // can be null.
|
| - bool value;
|
| -
|
| if (options.disable_encryption) {
|
| dtls_enabled_ = false;
|
| } else {
|
| // Enable DTLS by default if we have an identity store or a certificate.
|
| dtls_enabled_ = (dtls_identity_store || certificate);
|
| - // |constraints| can override the default |dtls_enabled_| value.
|
| - if (FindConstraint(constraints, MediaConstraintsInterface::kEnableDtlsSrtp,
|
| - &value, nullptr)) {
|
| - dtls_enabled_ = value;
|
| + // |rtc_configuration| can override the default |dtls_enabled_| value.
|
| + if (rtc_configuration.enable_dtls_srtp) {
|
| + dtls_enabled_ = *(rtc_configuration.enable_dtls_srtp);
|
| }
|
| }
|
|
|
| // Enable creation of RTP data channels if the kEnableRtpDataChannels is set.
|
| // It takes precendence over the disable_sctp_data_channels
|
| // PeerConnectionFactoryInterface::Options.
|
| - if (FindConstraint(
|
| - constraints, MediaConstraintsInterface::kEnableRtpDataChannels,
|
| - &value, NULL) && value) {
|
| - LOG(LS_INFO) << "Allowing RTP data engine.";
|
| + if (rtc_configuration.enable_rtp_data_channel) {
|
| data_channel_type_ = cricket::DCT_RTP;
|
| } else {
|
| // DTLS has to be enabled to use SCTP.
|
| if (!options.disable_sctp_data_channels && dtls_enabled_) {
|
| - LOG(LS_INFO) << "Allowing SCTP data engine.";
|
| data_channel_type_ = cricket::DCT_SCTP;
|
| }
|
| }
|
|
|
| - SetOptionFromOptionalConstraint(constraints,
|
| - MediaConstraintsInterface::kScreencastMinBitrate,
|
| - &video_options_.screencast_min_bitrate_kbps);
|
| -
|
| - SetOptionFromOptionalConstraint(constraints,
|
| - MediaConstraintsInterface::kCombinedAudioVideoBwe,
|
| - &audio_options_.combined_audio_video_bwe);
|
| + video_options_.screencast_min_bitrate_kbps =
|
| + rtc_configuration.screencast_min_bitrate;
|
| + audio_options_.combined_audio_video_bwe =
|
| + rtc_configuration.combined_audio_video_bwe;
|
|
|
| audio_options_.audio_jitter_buffer_max_packets =
|
| rtc::Optional<int>(rtc_configuration.audio_jitter_buffer_max_packets);
|
| @@ -698,10 +664,8 @@ void WebRtcSession::CreateOffer(
|
|
|
| void WebRtcSession::CreateAnswer(
|
| CreateSessionDescriptionObserver* observer,
|
| - const MediaConstraintsInterface* constraints,
|
| const cricket::MediaSessionOptions& session_options) {
|
| - webrtc_session_desc_factory_->CreateAnswer(observer, constraints,
|
| - session_options);
|
| + webrtc_session_desc_factory_->CreateAnswer(observer, session_options);
|
| }
|
|
|
| bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc,
|
|
|