Commit 4da102a4 authored by Antonio Gomes's avatar Antonio Gomes Committed by Commit Bot

Replace left over WebString/WebVector use in modules/peerconnection

Now, only cases actually needed (or logical) and test code still
uses these containers in modules/peerconnection. Production code
has been migrated.

BUG=787254
R=haraken@chromium.org

Change-Id: I40e76739853ae99475ed9ef0ecf2f7243011827c
Reviewed-on: https://chromium-review.googlesource.com/c/chromium/src/+/1992280
Commit-Queue: Antonio Gomes <tonikitoo@igalia.com>
Reviewed-by: default avatarKentaro Hara <haraken@chromium.org>
Cr-Commit-Position: refs/heads/master@{#729708}
parent e99120c1
...@@ -142,7 +142,7 @@ void RTCDTMFSender::PlayoutTask() { ...@@ -142,7 +142,7 @@ void RTCDTMFSender::PlayoutTask() {
DispatchEvent(*event.Release()); DispatchEvent(*event.Release());
return; return;
} }
WebString this_tone = tone_buffer_.Substring(0, 1); String this_tone = tone_buffer_.Substring(0, 1);
tone_buffer_ = tone_buffer_.Substring(1, tone_buffer_.length() - 1); tone_buffer_ = tone_buffer_.Substring(1, tone_buffer_.length() - 1);
// InsertDTMF handles both tones and ",", and calls DidPlayTone after // InsertDTMF handles both tones and ",", and calls DidPlayTone after
// the specified delay. // the specified delay.
......
...@@ -318,7 +318,7 @@ static cricket::IceParameters ConvertIceParameters( ...@@ -318,7 +318,7 @@ static cricket::IceParameters ConvertIceParameters(
static WebVector<webrtc::PeerConnectionInterface::IceServer> ConvertIceServers( static WebVector<webrtc::PeerConnectionInterface::IceServer> ConvertIceServers(
const HeapVector<Member<RTCIceServer>>& ice_servers) { const HeapVector<Member<RTCIceServer>>& ice_servers) {
Vector<webrtc::PeerConnectionInterface::IceServer> converted_ice_servers; WebVector<webrtc::PeerConnectionInterface::IceServer> converted_ice_servers;
for (const RTCIceServer* ice_server : ice_servers) { for (const RTCIceServer* ice_server : ice_servers) {
converted_ice_servers.emplace_back(ConvertIceServer(ice_server)); converted_ice_servers.emplace_back(ConvertIceServer(ice_server));
} }
......
...@@ -394,7 +394,7 @@ RTCRtpSendParameters* RTCRtpSender::getParameters() { ...@@ -394,7 +394,7 @@ RTCRtpSendParameters* RTCRtpSender::getParameters() {
// TODO(orphis): Forward missing fields from the WebRTC library: // TODO(orphis): Forward missing fields from the WebRTC library:
// codecPayloadType, dtx, ptime, maxFramerate, scaleResolutionDownBy. // codecPayloadType, dtx, ptime, maxFramerate, scaleResolutionDownBy.
RTCRtpEncodingParameters* encoding = RTCRtpEncodingParameters::Create(); RTCRtpEncodingParameters* encoding = RTCRtpEncodingParameters::Create();
encoding->setRid(WebString::FromUTF8(webrtc_encoding.rid)); encoding->setRid(String::FromUTF8(webrtc_encoding.rid));
encoding->setActive(webrtc_encoding.active); encoding->setActive(webrtc_encoding.active);
if (webrtc_encoding.max_bitrate_bps) { if (webrtc_encoding.max_bitrate_bps) {
encoding->setMaxBitrate(webrtc_encoding.max_bitrate_bps.value()); encoding->setMaxBitrate(webrtc_encoding.max_bitrate_bps.value());
......
...@@ -27,7 +27,7 @@ v8::Local<v8::Value> RTCStatsToValue(ScriptState* script_state, ...@@ -27,7 +27,7 @@ v8::Local<v8::Value> RTCStatsToValue(ScriptState* script_state,
std::unique_ptr<RTCStatsMember> member = stats->GetMember(i); std::unique_ptr<RTCStatsMember> member = stats->GetMember(i);
if (!member->IsDefined()) if (!member->IsDefined())
continue; continue;
WebString name = member->GetName(); String name = member->GetName();
switch (member->GetType()) { switch (member->GetType()) {
case webrtc::RTCStatsMemberInterface::kBool: case webrtc::RTCStatsMemberInterface::kBool:
builder.AddBoolean(name, member->ValueBool()); builder.AddBoolean(name, member->ValueBool());
......
Markdown is supported
0%
or
You are about to add 0 people to the discussion. Proceed with caution.
Finish editing this message first!
Please register or to comment