Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.
We can then drop the CongestionController and RemoteBitrateEstimator
completely from the receive streams.
BUG=webrtc:6847
Review-Url: https://codereview.webrtc.org/2669463006
Cr-Commit-Position: refs/heads/master@{#16459}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 5e21fe5..e21b076 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -60,6 +60,34 @@
const int Call::Config::kDefaultStartBitrateBps = 300000;
+namespace {
+
+// TODO(nisse): This really begs for a shared context struct.
+bool UseSendSideBwe(const std::vector<RtpExtension>& extensions,
+ bool transport_cc) {
+ if (!transport_cc)
+ return false;
+ for (const auto& extension : extensions) {
+ if (extension.uri == RtpExtension::kTransportSequenceNumberUri)
+ return true;
+ }
+ return false;
+}
+
+bool UseSendSideBwe(const VideoReceiveStream::Config& config) {
+ return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
+}
+
+bool UseSendSideBwe(const AudioReceiveStream::Config& config) {
+ return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
+}
+
+bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) {
+ return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc);
+}
+
+} // namespace
+
namespace internal {
class Call : public webrtc::Call,
@@ -199,16 +227,17 @@
struct ReceiveRtpConfig {
ReceiveRtpConfig() = default; // Needed by std::map
ReceiveRtpConfig(const std::vector<RtpExtension>& extensions,
- bool transport_cc)
- : extensions(extensions), transport_cc(transport_cc) {}
+ bool use_send_side_bwe)
+ : extensions(extensions), use_send_side_bwe(use_send_side_bwe) {}
// Registered RTP header extensions for each stream. Note that RTP header
// extensions are negotiated per track ("m= line") in the SDP, but we have
// no notion of tracks at the Call level. We therefore store the RTP header
// extensions per SSRC instead, which leads to some storage overhead.
RtpHeaderExtensionMap extensions;
- // Set if the RTCP feedback message needed for send side BWE was negotiated.
- bool transport_cc = false;
+ // Set if both RTP extension the RTCP feedback message needed for
+ // send side BWE are negotiated.
+ bool use_send_side_bwe = false;
};
std::map<uint32_t, ReceiveRtpConfig> receive_rtp_config_
GUARDED_BY(receive_crit_);
@@ -525,8 +554,7 @@
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
event_log_->LogAudioReceiveStreamConfig(config);
AudioReceiveStream* receive_stream = new AudioReceiveStream(
- &packet_router_,
- congestion_controller_->GetRemoteBitrateEstimator(true), config,
+ &packet_router_, config,
config_.audio_state, event_log_);
{
WriteLockScoped write_lock(*receive_crit_);
@@ -534,7 +562,7 @@
audio_receive_ssrcs_.end());
audio_receive_ssrcs_[config.rtp.remote_ssrc] = receive_stream;
receive_rtp_config_[config.rtp.remote_ssrc] =
- ReceiveRtpConfig(config.rtp.extensions, config.rtp.transport_cc);
+ ReceiveRtpConfig(config.rtp.extensions, UseSendSideBwe(config));
ConfigureSync(config.sync_group);
}
@@ -559,8 +587,10 @@
static_cast<webrtc::internal::AudioReceiveStream*>(receive_stream);
{
WriteLockScoped write_lock(*receive_crit_);
- uint32_t ssrc = audio_receive_stream->config().rtp.remote_ssrc;
-
+ const AudioReceiveStream::Config& config = audio_receive_stream->config();
+ uint32_t ssrc = config.rtp.remote_ssrc;
+ congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config))
+ ->RemoveStream(ssrc);
size_t num_deleted = audio_receive_ssrcs_.erase(ssrc);
RTC_DCHECK(num_deleted == 1);
const std::string& sync_group = audio_receive_stream->config().sync_group;
@@ -658,13 +688,13 @@
flexfec_receive_ssrcs_media_.end();
}
VideoReceiveStream* receive_stream = new VideoReceiveStream(
- num_cpu_cores_, protected_by_flexfec, congestion_controller_.get(),
+ num_cpu_cores_, protected_by_flexfec,
&packet_router_, std::move(configuration), module_process_thread_.get(),
call_stats_.get(), &remb_);
const webrtc::VideoReceiveStream::Config& config = receive_stream->config();
ReceiveRtpConfig receive_config(config.rtp.extensions,
- config.rtp.transport_cc);
+ UseSendSideBwe(config));
{
WriteLockScoped write_lock(*receive_crit_);
RTC_DCHECK(video_receive_ssrcs_.find(config.rtp.remote_ssrc) ==
@@ -714,6 +744,11 @@
RTC_CHECK(receive_stream_impl != nullptr);
ConfigureSync(receive_stream_impl->config().sync_group);
}
+ const VideoReceiveStream::Config& config = receive_stream_impl->config();
+
+ congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config))
+ ->RemoveStream(config.rtp.remote_ssrc);
+
UpdateAggregateNetworkState();
delete receive_stream_impl;
}
@@ -745,7 +780,7 @@
RTC_DCHECK(receive_rtp_config_.find(config.remote_ssrc) ==
receive_rtp_config_.end());
receive_rtp_config_[config.remote_ssrc] =
- ReceiveRtpConfig(config.rtp_header_extensions, config.transport_cc);
+ ReceiveRtpConfig(config.rtp_header_extensions, UseSendSideBwe(config));
}
// TODO(brandtr): Store config in RtcEventLog here.
@@ -765,7 +800,9 @@
{
WriteLockScoped write_lock(*receive_crit_);
- uint32_t ssrc = receive_stream_impl->GetConfig().remote_ssrc;
+ const FlexfecReceiveStream::Config& config =
+ receive_stream_impl->GetConfig();
+ uint32_t ssrc = config.remote_ssrc;
receive_rtp_config_.erase(ssrc);
// Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
@@ -785,6 +822,9 @@
++media_it;
}
+ congestion_controller_->GetRemoteBitrateEstimator(UseSendSideBwe(config))
+ ->RemoveStream(ssrc);
+
flexfec_receive_streams_.erase(receive_stream_impl);
}
@@ -1234,13 +1274,13 @@
void Call::NotifyBweOfReceivedPacket(const RtpPacketReceived& packet,
MediaType media_type) {
auto it = receive_rtp_config_.find(packet.Ssrc());
- bool transport_cc =
- (it != receive_rtp_config_.end()) && it->second.transport_cc;
+ bool use_send_side_bwe =
+ (it != receive_rtp_config_.end()) && it->second.use_send_side_bwe;
RTPHeader header;
packet.GetHeader(&header);
- if (!transport_cc && header.extension.hasTransportSequenceNumber) {
+ if (!use_send_side_bwe && header.extension.hasTransportSequenceNumber) {
// Inconsistent configuration of send side BWE. Do nothing.
// TODO(nisse): Without this check, we may produce RTCP feedback
// packets even when not negotiated. But it would be cleaner to
@@ -1255,7 +1295,7 @@
// FakeNetworkPipe::Process. We need to treat that as video. Tests
// should be fixed to use the same MediaType as the production code.
if (media_type != MediaType::AUDIO ||
- (transport_cc && header.extension.hasTransportSequenceNumber)) {
+ (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
congestion_controller_->OnReceivedPacket(
packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
header);
