Index: webrtc/video/call.cc |
diff --git a/webrtc/video/call.cc b/webrtc/video/call.cc |
index a2f07aa590f24e20e09e90db512517cd167bf881..928828f32a12e559618f38325e70ed60db7070cb 100644 |
--- a/webrtc/video/call.cc |
+++ b/webrtc/video/call.cc |
@@ -123,7 +123,6 @@ class Call : public webrtc::Call, public PacketReceiver { |
// and receivers. |
rtc::CriticalSection network_enabled_crit_; |
bool network_enabled_ GUARDED_BY(network_enabled_crit_); |
- TransportAdapter transport_adapter_; |
rtc::scoped_ptr<RWLockWrapper> receive_crit_; |
std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_ |
@@ -160,11 +159,8 @@ Call::Call(const Call::Config& config) |
next_channel_id_(0), |
config_(config), |
network_enabled_(true), |
- transport_adapter_(nullptr), |
receive_crit_(RWLockWrapper::CreateRWLock()), |
send_crit_(RWLockWrapper::CreateRWLock()) { |
- DCHECK(config.send_transport != nullptr); |
- |
DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0); |
DCHECK_GE(config.bitrate_config.start_bitrate_bps, |
config.bitrate_config.min_bitrate_bps); |
@@ -253,7 +249,7 @@ webrtc::VideoSendStream* Call::CreateVideoSendStream( |
// TODO(mflodman): Base the start bitrate on a current bandwidth estimate, if |
// the call has already started. |
VideoSendStream* send_stream = new VideoSendStream( |
- config_.send_transport, overuse_observer_proxy_.get(), num_cpu_cores_, |
+ overuse_observer_proxy_.get(), num_cpu_cores_, |
module_process_thread_.get(), channel_group_.get(), |
rtc::AtomicOps::Increment(&next_channel_id_), config, encoder_config, |
suspended_video_send_ssrcs_); |
@@ -313,7 +309,7 @@ webrtc::VideoReceiveStream* Call::CreateVideoReceiveStream( |
VideoReceiveStream* receive_stream = new VideoReceiveStream( |
num_cpu_cores_, channel_group_.get(), |
rtc::AtomicOps::Increment(&next_channel_id_), config, |
- config_.send_transport, config_.voice_engine); |
+ config_.voice_engine); |
// This needs to be taken before receive_crit_ as both locks need to be held |
// while changing network state. |