aboutsummaryrefslogtreecommitdiff
path: root/call/call.cc
diff options
context:
space:
mode:
Diffstat (limited to 'call/call.cc')
-rw-r--r--call/call.cc75
1 files changed, 12 insertions, 63 deletions
diff --git a/call/call.cc b/call/call.cc
index fa5d14d204..6b975edc6a 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -184,7 +184,7 @@ class Call final : public webrtc::Call,
public BitrateAllocator::LimitObserver {
public:
Call(Clock* clock,
- const Call::Config& config,
+ const CallConfig& config,
std::unique_ptr<RtpTransportControllerSendInterface> transport_send,
TaskQueueFactory* task_queue_factory);
~Call() override;
@@ -355,7 +355,7 @@ class Call final : public webrtc::Call,
const int num_cpu_cores_;
const std::unique_ptr<CallStats> call_stats_;
const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
- const Call::Config config_ RTC_GUARDED_BY(worker_thread_);
+ const CallConfig config_ RTC_GUARDED_BY(worker_thread_);
// Maps to config_.trials, can be used from any thread via `trials()`.
const FieldTrialsView& trials_;
@@ -388,11 +388,6 @@ class Call final : public webrtc::Call,
RTC_NO_UNIQUE_ADDRESS SequenceChecker receive_11993_checker_;
- // TODO(bugs.webrtc.org/11993): Move receive_rtp_config_ over to the
- // network thread.
- std::map<uint32_t, ReceiveStreamInterface*> receive_rtp_config_
- RTC_GUARDED_BY(&receive_11993_checker_);
-
// Audio and Video send streams are owned by the client that creates them.
// TODO(bugs.webrtc.org/11993): `audio_send_ssrcs_` and `video_send_ssrcs_`
// should be accessed on the network thread.
@@ -478,20 +473,22 @@ std::string Call::Stats::ToString(int64_t time_ms) const {
return ss.str();
}
-Call* Call::Create(const Call::Config& config) {
+std::unique_ptr<Call> Call::Create(const CallConfig& config) {
Clock* clock = Clock::GetRealTimeClock();
return Create(config, clock,
RtpTransportControllerSendFactory().Create(
config.ExtractTransportConfig(), clock));
}
-Call* Call::Create(const Call::Config& config,
- Clock* clock,
- std::unique_ptr<RtpTransportControllerSendInterface>
- transportControllerSend) {
+std::unique_ptr<Call> Call::Create(
+ const CallConfig& config,
+ Clock* clock,
+ std::unique_ptr<RtpTransportControllerSendInterface>
+ transportControllerSend) {
RTC_DCHECK(config.task_queue_factory);
- return new internal::Call(clock, config, std::move(transportControllerSend),
- config.task_queue_factory);
+ return std::make_unique<internal::Call>(clock, config,
+ std::move(transportControllerSend),
+ config.task_queue_factory);
}
// This method here to avoid subclasses has to implement this method.
@@ -658,7 +655,7 @@ void Call::SendStats::SetMinAllocatableRate(BitrateAllocationLimits limits) {
}
Call::Call(Clock* clock,
- const Call::Config& config,
+ const CallConfig& config,
std::unique_ptr<RtpTransportControllerSendInterface> transport_send,
TaskQueueFactory* task_queue_factory)
: clock_(clock),
@@ -844,11 +841,6 @@ webrtc::AudioReceiveStreamInterface* Call::CreateAudioReceiveStream(
// to live on the network thread.
receive_stream->RegisterWithTransport(&audio_receiver_controller_);
- // TODO(bugs.webrtc.org/11993): Update the below on the network thread.
- // We could possibly set up the audio_receiver_controller_ association up
- // as part of the async setup.
- RegisterReceiveStream(config.rtp.remote_ssrc, receive_stream);
-
ConfigureSync(config.sync_group);
auto it = audio_send_ssrcs_.find(config.rtp.local_ssrc);
@@ -883,8 +875,6 @@ void Call::DestroyAudioReceiveStream(
// for this sync_group.
ConfigureSync(audio_receive_stream->sync_group());
- UnregisterReceiveStream(ssrc);
-
UpdateAggregateNetworkState();
// TODO(bugs.webrtc.org/11993): Consider if deleting `audio_receive_stream`
// on the network thread would be better or if we'd need to tear down the
@@ -1021,15 +1011,6 @@ webrtc::VideoReceiveStreamInterface* Call::CreateVideoReceiveStream(
// TODO(bugs.webrtc.org/11993): Set this up asynchronously on the network
// thread.
receive_stream->RegisterWithTransport(&video_receiver_controller_);
-
- if (receive_stream->rtx_ssrc()) {
- // We record identical config for the rtx stream as for the main
- // stream. Since the transport_send_cc negotiation is per payload
- // type, we may get an incorrect value for the rtx stream, but
- // that is unlikely to matter in practice.
- RegisterReceiveStream(receive_stream->rtx_ssrc(), receive_stream);
- }
- RegisterReceiveStream(receive_stream->remote_ssrc(), receive_stream);
video_receive_streams_.insert(receive_stream);
ConfigureSync(receive_stream->sync_group());
@@ -1048,14 +1029,6 @@ void Call::DestroyVideoReceiveStream(
static_cast<VideoReceiveStream2*>(receive_stream);
// TODO(bugs.webrtc.org/11993): Unregister on the network thread.
receive_stream_impl->UnregisterFromTransport();
-
- // Remove all ssrcs pointing to a receive stream. As RTX retransmits on a
- // separate SSRC there can be either one or two.
- UnregisterReceiveStream(receive_stream_impl->remote_ssrc());
-
- if (receive_stream_impl->rtx_ssrc()) {
- UnregisterReceiveStream(receive_stream_impl->rtx_ssrc());
- }
video_receive_streams_.erase(receive_stream_impl);
ConfigureSync(receive_stream_impl->sync_group());
@@ -1083,8 +1056,6 @@ FlexfecReceiveStream* Call::CreateFlexfecReceiveStream(
// TODO(bugs.webrtc.org/11993): Set this up asynchronously on the network
// thread.
receive_stream->RegisterWithTransport(&video_receiver_controller_);
- RegisterReceiveStream(receive_stream->remote_ssrc(), receive_stream);
-
// TODO(brandtr): Store config in RtcEventLog here.
return receive_stream;
@@ -1100,8 +1071,6 @@ void Call::DestroyFlexfecReceiveStream(FlexfecReceiveStream* receive_stream) {
receive_stream_impl->UnregisterFromTransport();
auto ssrc = receive_stream_impl->remote_ssrc();
- UnregisterReceiveStream(ssrc);
-
// Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
// destroyed.
receive_side_cc_.RemoveStream(ssrc);
@@ -1465,26 +1434,6 @@ void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
receive_side_cc_.OnReceivedPacket(packet, media_type);
}
-bool Call::RegisterReceiveStream(uint32_t ssrc,
- ReceiveStreamInterface* stream) {
- RTC_DCHECK_RUN_ON(&receive_11993_checker_);
- RTC_DCHECK(stream);
- auto inserted = receive_rtp_config_.emplace(ssrc, stream);
- if (!inserted.second) {
- RTC_DLOG(LS_WARNING) << "ssrc already registered: " << ssrc;
- }
- return inserted.second;
-}
-
-bool Call::UnregisterReceiveStream(uint32_t ssrc) {
- RTC_DCHECK_RUN_ON(&receive_11993_checker_);
- size_t erased = receive_rtp_config_.erase(ssrc);
- if (!erased) {
- RTC_DLOG(LS_WARNING) << "ssrc wasn't registered: " << ssrc;
- }
- return erased != 0u;
-}
-
} // namespace internal
} // namespace webrtc
generated by cgit v1.2.3 (git 2.25.1) at 2024-05-23 17:46:40 +0000