Add flag enabling more packets to be retransmittable.

webrtc/modules/rtp_rtcp/source/rtp_sender_video.h

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
index 9fb4648b82e2138e2bfdb0e75e2d87a816710b37..ed5f1aa7e18db6b28708f97606bd2e993e60e5af 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_video.h
@@ -11,9 +11,8 @@
 #ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
 
-#include <list>
+#include <map>
 #include <memory>
-#include <vector>
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_sender.h"
@@ -32,10 +31,13 @@
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
+class RtpPacketizer;
 class RtpPacketToSend;
 
 class RTPSenderVideo {
  public:
+  static constexpr int64_t kTLRateWindowSizeMs = 2500;
+
   RTPSenderVideo(Clock* clock,
                  RTPSender* rtpSender,
                  FlexfecSender* flexfec_sender);
@@ -55,7 +57,8 @@ class RTPSenderVideo {
                  const uint8_t* payload_data,
                  size_t payload_size,
                  const RTPFragmentationHeader* fragmentation,
-                 const RTPVideoHeader* video_header);
+                 const RTPVideoHeader* video_header,
+                 int64_t expected_retransmission_time_ms);
 
   void SetVideoCodecType(RtpVideoCodecTypes type);
 
@@ -76,7 +79,22 @@ class RTPSenderVideo {
   int SelectiveRetransmissions() const;
   void SetSelectiveRetransmissions(uint8_t settings);
 
+ protected:
+  uint8_t GetTemporalId(const RTPVideoHeader& header) const;

[danilchap, 2017/09/04 09:03:05]

look like this one can be static or free function inside .cc

[sprang_webrtc, 2017/09/04 11:09:28]

Made it static. Call from unit test.

+  StorageType GetStorageType(uint8_t temporal_id,
+                             int32_t retransmission_settings,
+                             RtpPacketizer* packetizer,
+                             int64_t expected_retransmission_time);
+
  private:
+  struct TemporalLayerStats {
+    TemporalLayerStats()
+        : frame_rate_fp1000s(kTLRateWindowSizeMs, 1000 * 1000),

[brandtr, 2017/09/04 10:29:22]

Nice unit :)

[sprang_webrtc, 2017/09/04 11:09:28]

Hehe, thanks :)
Maybe I'll add a comment explicitly stating it is to increase precision at low
frame rates.

+          last_frame_time(0) {}
+    RateStatistics frame_rate_fp1000s;
+    int64_t last_frame_time;

[danilchap, 2017/09/04 09:03:05]

add _ms

[sprang_webrtc, 2017/09/04 11:09:28]

Done.

+  };
+
   size_t CalculateFecPacketOverhead() const EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   void SendVideoPacket(std::unique_ptr<RtpPacketToSend> packet,
@@ -103,6 +121,10 @@ class RTPSenderVideo {
 
   bool flexfec_enabled() const { return flexfec_sender_ != nullptr; }
 
+  bool UpdateConditionalRetransmit(uint8_t temporal_id,
+                                   int64_t expected_retransmission_time_ms)
+      EXCLUSIVE_LOCKS_REQUIRED(stats_crit_);
+
   RTPSender* const rtp_sender_;
   Clock* const clock_;
 
@@ -131,6 +153,10 @@ class RTPSenderVideo {
   RateStatistics fec_bitrate_ GUARDED_BY(stats_crit_);
   // Bitrate used for video payload and RTP headers.
   RateStatistics video_bitrate_ GUARDED_BY(stats_crit_);
+
+  std::map<int, TemporalLayerStats> frame_stats_by_temporal_layer_
+      GUARDED_BY(stats_crit_);
+
   OneTimeEvent first_frame_sent_;
 };