Make UDP receive buffer size configurable via field trial
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}diff --git a/media/engine/webrtc_video_engine.cc b/media/engine/webrtc_video_engine.cc
index 52ffcca..229a6c8 100644
--- a/media/engine/webrtc_video_engine.cc
+++ b/media/engine/webrtc_video_engine.cc
@@ -1532,8 +1532,21 @@
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 58c97ac..d56b3b6 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -1497,6 +1497,59 @@
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;
