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Increase the UDP receive buffer for video

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Lost packets have been seen in high-bitrate applications and increasing
the UDP receive buffer reduced the problems.

Bug: b/115713113
Change-Id: I671f528afeaea525150fdc2013f2b245778e5d16
Reviewed-on: https://webrtc-review.googlesource.com/c/107580
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25328}
This commit is contained in:
Johannes Kron 2018-10-23 11:31:19 +02:00 committed by Commit Bot
parent f0c449e3ff
commit d38a2b860b
4 changed files with 7 additions and 70 deletions
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3
media/engine/constants.cc
View File

@ -14,5 +14,6 @@ namespace cricket {
const int kMinVideoBitrateBps = 30000;
const int kVideoMtu = 1200;
const int kVideoRtpBufferSize = 65536;
const int kVideoRtpSendBufferSize = 65536;
const int kVideoRtpRecvBufferSize = 262144;
} // namespace cricket

3
media/engine/constants.h
View File

@ -14,7 +14,8 @@
namespace cricket {
extern const int kVideoMtu;
extern const int kVideoRtpBufferSize;
extern const int kVideoRtpSendBufferSize;
extern const int kVideoRtpRecvBufferSize;
extern const char kH264CodecName[];

16
media/engine/webrtcvideoengine.cc
View File

@ -1461,27 +1461,15 @@ void WebRtcVideoChannel::SetInterface(
MediaChannel::SetInterface(iface, media_transport);
// Set the RTP recv/send buffer to a bigger size.
// The group here can be either a positive integer with an explicit size, in
// which case that is used as 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 = kVideoRtpBufferSize;
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 = kVideoRtpBufferSize;
}
MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_RCVBUF,
recv_buffer_size);
kVideoRtpRecvBufferSize);
// Speculative change to increase the outbound socket buffer size.
// In b/15152257, we are seeing a significant number of packets discarded
// due to lack of socket buffer space, although it's not yet clear what the
// ideal value should be.
MediaChannel::SetOption(NetworkInterface::ST_RTP, rtc::Socket::OPT_SNDBUF,
kVideoRtpBufferSize);
kVideoRtpSendBufferSize);
}
void WebRtcVideoChannel::SetFrameDecryptor(

55
media/engine/webrtcvideoengine_unittest.cc
View File

@ -1485,60 +1485,7 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSendSetsTransportBufferSizes) {
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
EXPECT_EQ(64 * 1024, network_interface_.sendbuf_size());
EXPECT_EQ(64 * 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(64 * 1024, network_interface_.recvbuf_size());
}
EXPECT_EQ(256 * 1024, network_interface_.recvbuf_size());
}
// Test that stats work properly for a 1-1 call.