Merge pull request #10 from taste1981/cloudgaming

changes for cloud gaming to enable low latency transport
open-webrtc-toolkit · May 13, 2019 · da8eff5 · da8eff5
2 parents 5d65b55 + f84ce74
commit da8eff5
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Showing 4 changed files with 27 additions and 8 deletions.
diff --git a/call/rtp_transport_controller_send.cc b/call/rtp_transport_controller_send.cc
@@ -80,7 +80,7 @@ RtpTransportControllerSend::RtpTransportControllerSend(
                        TaskQueueExperimentEnabled(), controller_factory);
 
   process_thread_->RegisterModule(&pacer_, RTC_FROM_HERE);
-  process_thread_->RegisterModule(send_side_cc_.get(), RTC_FROM_HERE);
+  //process_thread_->RegisterModule(send_side_cc_.get(), RTC_FROM_HERE);
   if (!LowLatencyStreamingEnabled()) {
     process_thread_->Start();
   } else {
@@ -90,7 +90,7 @@ RtpTransportControllerSend::RtpTransportControllerSend(
 
 RtpTransportControllerSend::~RtpTransportControllerSend() {
   process_thread_->Stop();
-  process_thread_->DeRegisterModule(send_side_cc_.get());
+  //process_thread_->DeRegisterModule(send_side_cc_.get());
   process_thread_->DeRegisterModule(&pacer_);
 }
 

diff --git a/modules/pacing/paced_sender.cc b/modules/pacing/paced_sender.cc
@@ -49,6 +49,10 @@ namespace webrtc {
 const int64_t PacedSender::kMaxQueueLengthMs = 2000;
 const float PacedSender::kDefaultPaceMultiplier = 2.5f;
 
+int64_t capture_timestamp = 0;
+int64_t end_timestamp = 0;
+size_t total_size = 0;
+int64_t frame_count = 0;
 PacedSender::PacedSender(const Clock* clock,
                          PacketSender* packet_sender,
                          RtcEventLog* event_log)
@@ -92,6 +96,7 @@ PacedSender::PacedSender(const Clock* clock,
   UpdateBudgetWithElapsedTime(kMinPacketLimitMs);
   if (low_latency_mode_)
     prober_->SetEnabled(false);
+  recorder = fopen("pacer.txt", "a+");
 }
 
 PacedSender::~PacedSender() {}
@@ -356,6 +361,22 @@ void PacedSender::Process() {
                         << ",cu: " << clock_->TimeInMilliseconds()
                         << ",sz: " << packet.bytes
                         << ", q: " << packets_->SizeInPackets();
+
+      if (capture_timestamp != packet.capture_time_ms) {  // New frame
+        int64_t last_frame_cost = end_timestamp - capture_timestamp;
+        int64_t time_since_last_frame =
+            clock_->TimeInMilliseconds() - end_timestamp;
+        fprintf(recorder, "%lld\t%lld\t%zd\t%lld\n", frame_count, last_frame_cost,
+                total_size, time_since_last_frame);
+        frame_count++;
+        total_size = 0;
+        end_timestamp = clock_->TimeInMilliseconds();
+        capture_timestamp = packet.capture_time_ms;
+	  } else {
+        capture_timestamp = packet.capture_time_ms;
+        total_size += packet.bytes;
+        end_timestamp = clock_->TimeInMilliseconds();
+	  }
       if (is_probing && bytes_sent > recommended_probe_size)
         break;
     } else {
@@ -414,7 +435,7 @@ bool PacedSender::SendPacket(const PacketQueueInterface::Packet& packet,
       packet.retransmission, pacing_info);
   critsect_.Enter();
   RTC_LOG(LS_ERROR) << "a:" << clock_->TimeInMilliseconds();
-  if (success) {
+  if (success && !IsLowLatencyMode()) {
     if (first_sent_packet_ms_ == -1)
       first_sent_packet_ms_ = clock_->TimeInMilliseconds();
     if (!audio_packet || account_for_audio_) {

diff --git a/modules/pacing/paced_sender.h b/modules/pacing/paced_sender.h
@@ -213,6 +213,7 @@ class PacedSender : public Pacer {
 
   int64_t queue_time_limit RTC_GUARDED_BY(critsect_);
   bool account_for_audio_ RTC_GUARDED_BY(critsect_);
+  FILE* recorder;
 };
 }  // namespace webrtc
 #endif  // MODULES_PACING_PACED_SENDER_H_
diff --git a/modules/video_coding/video_sender.cc b/modules/video_coding/video_sender.cc
@@ -256,16 +256,13 @@ int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
   if (_encoder == nullptr)
     return VCM_UNINITIALIZED;
   SetEncoderParameters(encoder_params, encoder_has_internal_source);
-  if (_mediaOpt.DropFrame()) {
+  if (_mediaOpt.DropFrame() && !field_trial::IsEnabled("OWT-LowLatencyMode")) {
     RTC_LOG(LS_VERBOSE) << "Drop Frame "
                         << "target bitrate "
                         << encoder_params.target_bitrate.get_sum_bps()
                         << " loss rate " << encoder_params.loss_rate << " rtt "
                         << encoder_params.rtt << " input frame rate "
                         << encoder_params.input_frame_rate;
-    // For low latency mode we don't drop.
-    if (field_trial::IsEnabled("OWT-LowLatencyMode"))
-      return VCM_OK;
     post_encode_callback_->OnDroppedFrame(
         EncodedImageCallback::DropReason::kDroppedByMediaOptimizations);
     return VCM_OK;
@@ -304,7 +301,7 @@ int32_t VideoSender::AddVideoFrame(const VideoFrame& videoFrame,
       _encoder->Encode(converted_frame, codecSpecificInfo, next_frame_types);
   if (ret < 0) {
     RTC_LOG(LS_ERROR) << "Failed to encode frame. Error code: " << ret;
-    return ret;
+    //return ret;
   }
 
   {