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Unified Diff: webrtc/media/engine/fakewebrtccall.cc

Issue 1757683002: Make the audio channel communicate network state changes to the call. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Added end-to-end tests for all network states Created 4 years, 9 months ago
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Index: webrtc/media/engine/fakewebrtccall.cc
diff --git a/webrtc/media/engine/fakewebrtccall.cc b/webrtc/media/engine/fakewebrtccall.cc
index 1af11afd24e63c553de34b8785f6762dbe3ac2e5..bf20380ae8046c91b9baa42304f985b6de96e148 100644
--- a/webrtc/media/engine/fakewebrtccall.cc
+++ b/webrtc/media/engine/fakewebrtccall.cc
@@ -227,7 +227,8 @@ void FakeVideoReceiveStream::SetStats(
FakeCall::FakeCall(const webrtc::Call::Config& config)
: config_(config),
- network_state_(webrtc::kNetworkUp),
+ audio_network_state_(webrtc::kNetworkUp),
+ video_network_state_(webrtc::kNetworkUp),
num_created_send_streams_(0),
num_created_receive_streams_(0) {}
@@ -276,8 +277,16 @@ const FakeAudioReceiveStream* FakeCall::GetAudioReceiveStream(uint32_t ssrc) {
return nullptr;
}
-webrtc::NetworkState FakeCall::GetNetworkState() const {
- return network_state_;
+webrtc::NetworkState FakeCall::GetNetworkState(webrtc::MediaType media) const {
+ switch (media) {
+ case webrtc::MediaType::AUDIO:
+ return audio_network_state_;
+ case webrtc::MediaType::VIDEO:
+ return video_network_state_;
+ default:
pbos-webrtc 2016/03/22 10:17:41 I think you should be able to enumerate all enums
skvlad 2016/03/22 20:01:44 Done.
+ ADD_FAILURE() << "GetNetworkState called with unknown parameter.";
+ return webrtc::kNetworkDown;
+ }
}
webrtc::AudioSendStream* FakeCall::CreateAudioSendStream(
@@ -293,7 +302,7 @@ void FakeCall::DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) {
audio_send_streams_.end(),
static_cast<FakeAudioSendStream*>(send_stream));
if (it == audio_send_streams_.end()) {
- ADD_FAILURE() << "DestroyAudioSendStream called with unknown paramter.";
+ ADD_FAILURE() << "DestroyAudioSendStream called with unknown parameter.";
} else {
delete *it;
audio_send_streams_.erase(it);
@@ -313,7 +322,7 @@ void FakeCall::DestroyAudioReceiveStream(
audio_receive_streams_.end(),
static_cast<FakeAudioReceiveStream*>(receive_stream));
if (it == audio_receive_streams_.end()) {
- ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown paramter.";
+ ADD_FAILURE() << "DestroyAudioReceiveStream called with unknown parameter.";
} else {
delete *it;
audio_receive_streams_.erase(it);
@@ -335,7 +344,7 @@ void FakeCall::DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) {
video_send_streams_.end(),
static_cast<FakeVideoSendStream*>(send_stream));
if (it == video_send_streams_.end()) {
- ADD_FAILURE() << "DestroyVideoSendStream called with unknown paramter.";
+ ADD_FAILURE() << "DestroyVideoSendStream called with unknown parameter.";
} else {
delete *it;
video_send_streams_.erase(it);
@@ -355,7 +364,7 @@ void FakeCall::DestroyVideoReceiveStream(
video_receive_streams_.end(),
static_cast<FakeVideoReceiveStream*>(receive_stream));
if (it == video_receive_streams_.end()) {
- ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown paramter.";
+ ADD_FAILURE() << "DestroyVideoReceiveStream called with unknown parameter.";
} else {
delete *it;
video_receive_streams_.erase(it);
@@ -416,8 +425,19 @@ void FakeCall::SetBitrateConfig(
config_.bitrate_config = bitrate_config;
}
-void FakeCall::SignalNetworkState(webrtc::NetworkState state) {
- network_state_ = state;
+void FakeCall::SignalChannelNetworkState(webrtc::MediaType media,
+ webrtc::NetworkState state) {
+ switch (media) {
+ case webrtc::MediaType::AUDIO:
+ audio_network_state_ = state;
+ break;
+ case webrtc::MediaType::VIDEO:
+ video_network_state_ = state;
+ break;
+ default:
pbos-webrtc 2016/03/22 10:17:41 same here
skvlad 2016/03/22 20:01:44 Done.
+ ADD_FAILURE()
+ << "SignalChannelNetworkState called with unknown parameter.";
+ }
}
void FakeCall::OnSentPacket(const rtc::SentPacket& sent_packet) {
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