Move RTP keep-alive config from VideoSendStream::Config to Call::Config
This makes more sense since logically it's a transport level feature,
not a media stream feature. Even if the implementation details forces it
to be an rtp stream detail, for the moment.
BUG=webrtc:7907
Review-Url: https://codereview.webrtc.org/2978503002
Cr-Commit-Position: refs/heads/master@{#18963}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 41f53fe..3966d5e 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -727,7 +727,8 @@
num_cpu_cores_, module_process_thread_.get(), &worker_queue_,
call_stats_.get(), transport_send_.get(), bitrate_allocator_.get(),
video_send_delay_stats_.get(), event_log_, std::move(config),
- std::move(encoder_config), suspended_video_send_ssrcs_);
+ std::move(encoder_config), suspended_video_send_ssrcs_,
+ config_.keepalive_config);
{
WriteLockScoped write_lock(*send_crit_);
diff --git a/webrtc/call/call.h b/webrtc/call/call.h
index 86142f0..f99ba85 100644
--- a/webrtc/call/call.h
+++ b/webrtc/call/call.h
@@ -112,6 +112,12 @@
// RtcEventLog to use for this call. Required.
// Use webrtc::RtcEventLog::CreateNull() for a null implementation.
RtcEventLog* event_log = nullptr;
+
+ // Enables periodic sending if empty keep-alive messages that helps prevent
+ // network time-out events. The packets adhere to RFC6263 section 4.6, and
+ // by default use payload type 20, as described in 3GPP TS 24.229,
+ // Appendix K.5.2.1.
+ RtpKeepAliveConfig keepalive_config;
};
struct Stats {
diff --git a/webrtc/video/video_send_stream.cc b/webrtc/video/video_send_stream.cc
index 3565f9c..9bfc985 100644
--- a/webrtc/video/video_send_stream.cc
+++ b/webrtc/video/video_send_stream.cc
@@ -349,7 +349,8 @@
const VideoSendStream::Config* config,
int initial_encoder_max_bitrate,
std::map<uint32_t, RtpState> suspended_ssrcs,
- VideoEncoderConfig::ContentType content_type);
+ VideoEncoderConfig::ContentType content_type,
+ const RtpKeepAliveConfig& keepalive_config);
~VideoSendStreamImpl() override;
// RegisterProcessThread register |module_process_thread| with those objects
@@ -480,7 +481,8 @@
const VideoSendStream::Config* config,
int initial_encoder_max_bitrate,
const std::map<uint32_t, RtpState>& suspended_ssrcs,
- VideoEncoderConfig::ContentType content_type)
+ VideoEncoderConfig::ContentType content_type,
+ const RtpKeepAliveConfig& keepalive_config)
: send_stream_(send_stream),
done_event_(done_event),
stats_proxy_(stats_proxy),
@@ -493,7 +495,8 @@
config_(config),
initial_encoder_max_bitrate_(initial_encoder_max_bitrate),
suspended_ssrcs_(suspended_ssrcs),
- content_type_(content_type) {}
+ content_type_(content_type),
+ keepalive_config_(keepalive_config) {}
~ConstructionTask() override { done_event_->Set(); }
@@ -503,7 +506,7 @@
stats_proxy_, rtc::TaskQueue::Current(), call_stats_, transport_,
bitrate_allocator_, send_delay_stats_, vie_encoder_, event_log_,
config_, initial_encoder_max_bitrate_, std::move(suspended_ssrcs_),
- content_type_));
+ content_type_, keepalive_config_));
return true;
}
@@ -520,6 +523,7 @@
int initial_encoder_max_bitrate_;
std::map<uint32_t, RtpState> suspended_ssrcs_;
const VideoEncoderConfig::ContentType content_type_;
+ const RtpKeepAliveConfig& keepalive_config_;
};
class VideoSendStream::DestructAndGetRtpStateTask : public rtc::QueuedTask {
@@ -632,7 +636,8 @@
RtcEventLog* event_log,
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
- const std::map<uint32_t, RtpState>& suspended_ssrcs)
+ const std::map<uint32_t, RtpState>& suspended_ssrcs,
+ const RtpKeepAliveConfig& keepalive_config)
: worker_queue_(worker_queue),
thread_sync_event_(false /* manual_reset */, false),
stats_proxy_(Clock::GetRealTimeClock(),
@@ -648,7 +653,7 @@
&send_stream_, &thread_sync_event_, &stats_proxy_, vie_encoder_.get(),
module_process_thread, call_stats, transport, bitrate_allocator,
send_delay_stats, event_log, &config_, encoder_config.max_bitrate_bps,
- suspended_ssrcs, encoder_config.content_type)));
+ suspended_ssrcs, encoder_config.content_type, keepalive_config)));
// Wait for ConstructionTask to complete so that |send_stream_| can be used.
// |module_process_thread| must be registered and deregistered on the thread
@@ -767,7 +772,8 @@
const VideoSendStream::Config* config,
int initial_encoder_max_bitrate,
std::map<uint32_t, RtpState> suspended_ssrcs,
- VideoEncoderConfig::ContentType content_type)
+ VideoEncoderConfig::ContentType content_type,
+ const RtpKeepAliveConfig& keepalive_config)
: send_side_bwe_with_overhead_(
webrtc::field_trial::IsEnabled("WebRTC-SendSideBwe-WithOverhead")),
stats_proxy_(stats_proxy),
@@ -805,7 +811,7 @@
transport->send_side_cc()->GetRetransmissionRateLimiter(),
this,
config_->rtp.ssrcs.size(),
- config_->rtp.keep_alive)),
+ keepalive_config)),
payload_router_(rtp_rtcp_modules_,
config_->encoder_settings.payload_type),
weak_ptr_factory_(this),
diff --git a/webrtc/video/video_send_stream.h b/webrtc/video/video_send_stream.h
index 2f1fec0..59cb797 100644
--- a/webrtc/video/video_send_stream.h
+++ b/webrtc/video/video_send_stream.h
@@ -57,7 +57,8 @@
RtcEventLog* event_log,
VideoSendStream::Config config,
VideoEncoderConfig encoder_config,
- const std::map<uint32_t, RtpState>& suspended_ssrcs);
+ const std::map<uint32_t, RtpState>& suspended_ssrcs,
+ const RtpKeepAliveConfig& keepalive_config);
~VideoSendStream() override;
diff --git a/webrtc/video/video_send_stream_tests.cc b/webrtc/video/video_send_stream_tests.cc
index aa9d2fb..53796b8 100644
--- a/webrtc/video/video_send_stream_tests.cc
+++ b/webrtc/video/video_send_stream_tests.cc
@@ -3415,6 +3415,14 @@
public:
KeepaliveObserver() : SendTest(kDefaultTimeoutMs) {}
+ Call::Config GetSenderCallConfig() override {
+ Call::Config config = SendTest::GetSenderCallConfig();
+ config.keepalive_config.timeout_interval_ms = kTimeoutMs;
+ config.keepalive_config.payload_type =
+ CallTest::kDefaultKeepalivePayloadType;
+ return config;
+ }
+
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTPHeader header;
@@ -3431,15 +3439,6 @@
return SEND_PACKET;
}
- void ModifyVideoConfigs(
- VideoSendStream::Config* send_config,
- std::vector<VideoReceiveStream::Config>* receive_configs,
- VideoEncoderConfig* encoder_config) override {
- send_config->rtp.keep_alive.timeout_interval_ms = kTimeoutMs;
- send_config->rtp.keep_alive.payload_type =
- CallTest::kDefaultKeepalivePayloadType;
- }
-
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for keep-alive packet.";
}
diff --git a/webrtc/video_send_stream.h b/webrtc/video_send_stream.h
index 2b3ee78..c5a1f9b 100644
--- a/webrtc/video_send_stream.h
+++ b/webrtc/video_send_stream.h
@@ -168,8 +168,6 @@
int payload_type = -1;
} rtx;
- RtpKeepAliveConfig keep_alive;
-
// RTCP CNAME, see RFC 3550.
std::string c_name;
} rtp;
