commit 898395d1810654e412d53197fae2c6493828a1f2
author Elad Alon <eladalon@webrtc.org> Wed Apr 10 13:55:00 2019
committer Commit Bot <commit-bot@chromium.org> Wed Apr 10 14:32:00 2019
tree dbef00c93bb5f5f86b28d538a71bac1a2bfa3b94
parent 0810a7c25a156c1945771eb16ca1a6512be4f295

RTPSenderVideo::GetSentRtpPacketInfo() over a set of sequence numbers

Add a version of RTPSenderVideo::GetSentRtpPacketInfo() that operates
over a set of numbers, so as to only grab the lock once.

Bug: webrtc:10501
Change-Id: I9453b0cb44dcd6e2ce196390b2c5c9a7dd6d800a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132014
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27544}

call/BUILD.gn

diff --git a/call/BUILD.gn b/call/BUILD.gn
index 6ddaf6e..bcd3078 100644
--- a/call/BUILD.gn
+++ b/call/BUILD.gn
@@ -124,6 +124,7 @@
   deps = [
     ":bitrate_configurator",
     ":rtp_interfaces",
+    "../api:array_view",
     "../api:fec_controller_api",
     "../api:network_state_predictor_api",
     "../api:transport_api",

call/rtp_video_sender.cc

diff --git a/call/rtp_video_sender.cc b/call/rtp_video_sender.cc
index 94a8fad..019f842 100644
--- a/call/rtp_video_sender.cc
+++ b/call/rtp_video_sender.cc
@@ -745,15 +745,15 @@
   return protection_bitrate_bps_;
 }
 
-absl::optional<RtpSequenceNumberMap::Info> RtpVideoSender::GetSentRtpPacketInfo(
+std::vector<RtpSequenceNumberMap::Info> RtpVideoSender::GetSentRtpPacketInfos(
     uint32_t ssrc,
-    uint16_t seq_num) const {
+    rtc::ArrayView<const uint16_t> sequence_numbers) const {
   for (const auto& rtp_stream : rtp_streams_) {
     if (ssrc == rtp_stream.rtp_rtcp->SSRC()) {
-      return rtp_stream.sender_video->GetSentRtpPacketInfo(seq_num);
+      return rtp_stream.sender_video->GetSentRtpPacketInfos(sequence_numbers);
     }
   }
-  return absl::nullopt;
+  return std::vector<RtpSequenceNumberMap::Info>();
 }
 
 int RtpVideoSender::ProtectionRequest(const FecProtectionParams* delta_params,

call/rtp_video_sender.h

diff --git a/call/rtp_video_sender.h b/call/rtp_video_sender.h
index 8518c91..e17bb49 100644
--- a/call/rtp_video_sender.h
+++ b/call/rtp_video_sender.h
@@ -17,6 +17,7 @@
 #include <vector>
 
 #include "absl/types/optional.h"
+#include "api/array_view.h"
 #include "api/call/transport.h"
 #include "api/fec_controller.h"
 #include "api/video_codecs/video_encoder.h"
@@ -140,9 +141,9 @@
                        size_t height,
                        size_t num_temporal_layers) override;
 
-  absl::optional<RtpSequenceNumberMap::Info> GetSentRtpPacketInfo(
+  std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
       uint32_t ssrc,
-      uint16_t seq_num) const override;
+      rtc::ArrayView<const uint16_t> sequence_numbers) const override;
 
   // From PacketFeedbackObserver.
   void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) override;

call/rtp_video_sender_interface.h

diff --git a/call/rtp_video_sender_interface.h b/call/rtp_video_sender_interface.h
index 51cf56b..3208e94 100644
--- a/call/rtp_video_sender_interface.h
+++ b/call/rtp_video_sender_interface.h
@@ -15,6 +15,7 @@
 #include <vector>
 
 #include "absl/types/optional.h"
+#include "api/array_view.h"
 #include "call/rtp_config.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "modules/rtp_rtcp/source/rtp_sequence_number_map.h"
@@ -57,9 +58,9 @@
   virtual void SetEncodingData(size_t width,
                                size_t height,
                                size_t num_temporal_layers) = 0;
-  virtual absl::optional<RtpSequenceNumberMap::Info> GetSentRtpPacketInfo(
+  virtual std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos(
       uint32_t ssrc,
-      uint16_t seq_num) const = 0;
+      rtc::ArrayView<const uint16_t> sequence_numbers) const = 0;
 };
 }  // namespace webrtc
 #endif  // CALL_RTP_VIDEO_SENDER_INTERFACE_H_