From 5a0665bea4985b1797506024b6f33ea67801a9a9 Mon Sep 17 00:00:00 2001
From: Johannes Kron <kron@webrtc.org>
Date: Mon, 8 Apr 2019 10:35:50 +0200
Subject: [PATCH] Make UDP receive buffer size configurable via field trial
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Bug: chromium:939340
Change-Id: I2ab18554d12a1e9c62f5d3d8f8237cc4d0a1a78c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/131395
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27476}
---
 media/engine/webrtc_video_engine.cc          | 15 +++++-
 media/engine/webrtc_video_engine_unittest.cc | 53 ++++++++++++++++++++
 2 files changed, 67 insertions(+), 1 deletion(-)

diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 52ffcca97b..229a6c8cf9 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -1532,8 +1532,21 @@ void WebRtcVideoChannel::SetInterface(
   MediaChannel::SetInterface(iface, media_transport);
   // Set the RTP recv/send buffer to a bigger size.
 
+  // The group should be a positive integer with an explicit size, in
+  // which case that is used as UDP recevie buffer size. All other values shall
+  // result in the default value being used.
+  const std::string group_name =
+      webrtc::field_trial::FindFullName("WebRTC-IncreasedReceivebuffers");
+  int recv_buffer_size = kVideoRtpRecvBufferSize;
+  if (!group_name.empty() &&
+      (sscanf(group_name.c_str(), "%d", &recv_buffer_size) != 1 ||
+       recv_buffer_size <= 0)) {
+    RTC_LOG(LS_WARNING) << "Invalid receive buffer size: " << group_name;
+    recv_buffer_size = kVideoRtpRecvBufferSize;
+  }
+
   MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_RCVBUF,
-                          kVideoRtpRecvBufferSize);
+                          recv_buffer_size);
 
   // Speculative change to increase the outbound socket buffer size.
   // In b/15152257, we are seeing a significant number of packets discarded
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 58c97ac3a3..d56b3b6892 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -1497,6 +1497,59 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSendSetsTransportBufferSizes) {
   EXPECT_EQ(256 * 1024, network_interface_.recvbuf_size());
 }
 
+// Test that we properly set the send and recv buffer sizes when overriding
+// via field trials.
+TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSize) {
+  // Set field trial to override the default recv buffer size, and then re-run
+  // setup where the interface is created and configured.
+  const int kCustomRecvBufferSize = 123456;
+  webrtc::test::ScopedFieldTrials field_trial(
+      "WebRTC-IncreasedReceivebuffers/123456/");
+  SetUp();
+
+  EXPECT_TRUE(SetOneCodec(DefaultCodec()));
+  EXPECT_TRUE(SetSend(true));
+  EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
+  EXPECT_EQ(kCustomRecvBufferSize, network_interface_.recvbuf_size());
+}
+
+// Test that we properly set the send and recv buffer sizes when overriding
+// via field trials with suffix.
+TEST_F(WebRtcVideoChannelBaseTest, OverridesRecvBufferSizeWithSuffix) {
+  // Set field trial to override the default recv buffer size, and then re-run
+  // setup where the interface is created and configured.
+  const int kCustomRecvBufferSize = 123456;
+  webrtc::test::ScopedFieldTrials field_trial(
+      "WebRTC-IncreasedReceivebuffers/123456_Dogfood/");
+  SetUp();
+
+  EXPECT_TRUE(SetOneCodec(DefaultCodec()));
+  EXPECT_TRUE(SetSend(true));
+  EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
+  EXPECT_EQ(kCustomRecvBufferSize, network_interface_.recvbuf_size());
+}
+
+// Test that we properly set the send and recv buffer sizes when overriding
+// via field trials that don't make any sense.
+TEST_F(WebRtcVideoChannelBaseTest, InvalidRecvBufferSize) {
+  // Set bogus field trial values to override the default recv buffer size, and
+  // then re-run setup where the interface is created and configured. The
+  // default value should still be used.
+
+  for (std::string group : {" ", "NotANumber", "-1", "0"}) {
+    std::string field_trial_string = "WebRTC-IncreasedReceivebuffers/";
+    field_trial_string += group;
+    field_trial_string += "/";
+    webrtc::test::ScopedFieldTrials field_trial(field_trial_string);
+    SetUp();
+
+    EXPECT_TRUE(SetOneCodec(DefaultCodec()));
+    EXPECT_TRUE(SetSend(true));
+    EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
+    EXPECT_EQ(256 * 1024, network_interface_.recvbuf_size());
+  }
+}
+
 // Test that stats work properly for a 1-1 call.
 TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
   const int kDurationSec = 3;