Remove getting max payload length from default module.
Moving functionality to get max payload length from default RTP module
to the payload router.
I'll make a follow up CL changing asserts to DCHECK in rtp_rtcp_impl.cc.
BUG=769
TEST=New unittest and existing sender mtu test
R=stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/36119004
Cr-Commit-Position: refs/heads/master@{#8345}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8345 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
index 3dcbac5..e81740d 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
@@ -555,31 +555,8 @@
}
uint16_t ModuleRtpRtcpImpl::MaxDataPayloadLength() const {
- // Assuming IP/UDP.
- uint16_t min_data_payload_length = IP_PACKET_SIZE - 28;
-
- if (IsDefaultModule()) {
- // For default we need to update all child modules too.
- CriticalSectionScoped lock(critical_section_module_ptrs_.get());
- std::vector<ModuleRtpRtcpImpl*>::const_iterator it = child_modules_.begin();
- while (it != child_modules_.end()) {
- RtpRtcp* module = *it;
- if (module) {
- uint16_t data_payload_length =
- module->MaxDataPayloadLength();
- if (data_payload_length < min_data_payload_length) {
- min_data_payload_length = data_payload_length;
- }
- }
- it++;
- }
- }
-
- uint16_t data_payload_length = rtp_sender_.MaxDataPayloadLength();
- if (data_payload_length < min_data_payload_length) {
- min_data_payload_length = data_payload_length;
- }
- return min_data_payload_length;
+ assert(!IsDefaultModule());
+ return rtp_sender_.MaxDataPayloadLength();
}
int32_t ModuleRtpRtcpImpl::SetTransportOverhead(
diff --git a/webrtc/video_engine/payload_router.cc b/webrtc/video_engine/payload_router.cc
index f815d13..00c64ad 100644
--- a/webrtc/video_engine/payload_router.cc
+++ b/webrtc/video_engine/payload_router.cc
@@ -12,6 +12,7 @@
#include "webrtc/base/checks.h"
#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
namespace webrtc {
@@ -22,6 +23,11 @@
PayloadRouter::~PayloadRouter() {}
+size_t PayloadRouter::DefaultMaxPayloadLength() {
+ const size_t kIpUdpSrtpLength = 44;
+ return IP_PACKET_SIZE - kIpUdpSrtpLength;
+}
+
void PayloadRouter::SetSendingRtpModules(
const std::list<RtpRtcp*>& rtp_modules) {
CriticalSectionScoped cs(crit_.get());
@@ -47,7 +53,7 @@
uint32_t time_stamp,
int64_t capture_time_ms,
const uint8_t* payload_data,
- size_t payload_size,
+ size_t payload_length,
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_video_hdr) {
CriticalSectionScoped cs(crit_.get());
@@ -62,7 +68,18 @@
stream_idx = rtp_video_hdr->simulcastIdx;
return rtp_modules_[stream_idx]->SendOutgoingData(
frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
- payload_size, fragmentation, rtp_video_hdr) == 0 ? true : false;
+ payload_length, fragmentation, rtp_video_hdr) == 0 ? true : false;
+}
+
+size_t PayloadRouter::MaxPayloadLength() const {
+ size_t min_payload_length = DefaultMaxPayloadLength();
+ CriticalSectionScoped cs(crit_.get());
+ for (auto* rtp_module : rtp_modules_) {
+ size_t module_payload_length = rtp_module->MaxDataPayloadLength();
+ if (module_payload_length < min_payload_length)
+ min_payload_length = module_payload_length;
+ }
+ return min_payload_length;
}
} // namespace webrtc
diff --git a/webrtc/video_engine/payload_router.h b/webrtc/video_engine/payload_router.h
index 3cc0bb0..baad38f5 100644
--- a/webrtc/video_engine/payload_router.h
+++ b/webrtc/video_engine/payload_router.h
@@ -33,6 +33,8 @@
PayloadRouter();
~PayloadRouter();
+ static size_t DefaultMaxPayloadLength();
+
// Rtp modules are assumed to be sorted in simulcast index order.
void SetSendingRtpModules(const std::list<RtpRtcp*>& rtp_modules);
@@ -52,6 +54,10 @@
const RTPFragmentationHeader* fragmentation,
const RTPVideoHeader* rtp_video_hdr);
+ // Returns the maximum allowed data payload length, given the configured MTU
+ // and RTP headers.
+ size_t MaxPayloadLength() const;
+
private:
scoped_ptr<CriticalSectionWrapper> crit_;
diff --git a/webrtc/video_engine/payload_router_unittest.cc b/webrtc/video_engine/payload_router_unittest.cc
index b4d20bf..abab965 100644
--- a/webrtc/video_engine/payload_router_unittest.cc
+++ b/webrtc/video_engine/payload_router_unittest.cc
@@ -127,5 +127,38 @@
&payload_2, 1, NULL, &rtp_hdr_2));
}
+TEST_F(PayloadRouterTest, MaxPayloadLength) {
+ // Without any limitations from the modules, verify we get the max payload
+ // length for IP/UDP/SRTP with a MTU of 150 bytes.
+ const size_t kDefaultMaxLength = 1500 - 20 - 8 - 12 - 4;
+ EXPECT_EQ(kDefaultMaxLength, payload_router_->DefaultMaxPayloadLength());
+ EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
+
+ MockRtpRtcp rtp_1;
+ MockRtpRtcp rtp_2;
+ std::list<RtpRtcp*> modules;
+ modules.push_back(&rtp_1);
+ modules.push_back(&rtp_2);
+ payload_router_->SetSendingRtpModules(modules);
+
+ // Modules return a higher length than the default value.
+ EXPECT_CALL(rtp_1, MaxDataPayloadLength())
+ .Times(1)
+ .WillOnce(Return(kDefaultMaxLength + 10));
+ EXPECT_CALL(rtp_2, MaxDataPayloadLength())
+ .Times(1)
+ .WillOnce(Return(kDefaultMaxLength + 10));
+ EXPECT_EQ(kDefaultMaxLength, payload_router_->MaxPayloadLength());
+
+ // The modules return a value lower than default.
+ const size_t kTestMinPayloadLength = 1001;
+ EXPECT_CALL(rtp_1, MaxDataPayloadLength())
+ .Times(1)
+ .WillOnce(Return(kTestMinPayloadLength + 10));
+ EXPECT_CALL(rtp_2, MaxDataPayloadLength())
+ .Times(1)
+ .WillOnce(Return(kTestMinPayloadLength));
+ EXPECT_EQ(kTestMinPayloadLength, payload_router_->MaxPayloadLength());
+}
} // namespace webrtc
diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc
index 3afe758..0dd7e91 100644
--- a/webrtc/video_engine/vie_encoder.cc
+++ b/webrtc/video_engine/vie_encoder.cc
@@ -217,7 +217,7 @@
send_padding_ = video_codec.numberOfSimulcastStreams > 1;
}
if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
- default_rtp_rtcp_->MaxDataPayloadLength()) != 0) {
+ PayloadRouter::DefaultMaxPayloadLength()) != 0) {
return false;
}
if (default_rtp_rtcp_->RegisterSendPayload(video_codec) != 0) {
@@ -324,6 +324,7 @@
}
int32_t ViEEncoder::DeRegisterExternalEncoder(uint8_t pl_type) {
+ DCHECK(send_payload_router_ != NULL);
webrtc::VideoCodec current_send_codec;
if (vcm_.SendCodec(¤t_send_codec) == VCM_OK) {
uint32_t current_bitrate_bps = 0;
@@ -340,8 +341,6 @@
// If the external encoder is the current send codec, use vcm internal
// encoder.
if (current_send_codec.plType == pl_type) {
- uint16_t max_data_payload_length =
- default_rtp_rtcp_->MaxDataPayloadLength();
{
CriticalSectionScoped cs(data_cs_.get());
send_padding_ = current_send_codec.numberOfSimulcastStreams > 1;
@@ -351,6 +350,7 @@
// a hack to prevent the following code from crashing. This should be fixed
// for realz. https://code.google.com/p/chromium/issues/detail?id=348222
current_send_codec.extra_options = NULL;
+ size_t max_data_payload_length = send_payload_router_->MaxPayloadLength();
if (vcm_.RegisterSendCodec(¤t_send_codec, number_of_cores_,
max_data_payload_length) != VCM_OK) {
LOG(LS_INFO) << "De-registered the currently used external encoder ("
@@ -363,6 +363,7 @@
}
int32_t ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
+ DCHECK(send_payload_router_ != NULL);
// Setting target width and height for VPM.
if (vpm_.SetTargetResolution(video_codec.width, video_codec.height,
video_codec.maxFramerate) != VPM_OK) {
@@ -379,13 +380,11 @@
video_codec.numberOfSimulcastStreams);
default_rtp_rtcp_->SetTargetSendBitrate(stream_bitrates);
- uint16_t max_data_payload_length =
- default_rtp_rtcp_->MaxDataPayloadLength();
-
{
CriticalSectionScoped cs(data_cs_.get());
send_padding_ = video_codec.numberOfSimulcastStreams > 1;
}
+ size_t max_data_payload_length = send_payload_router_->MaxPayloadLength();
if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
max_data_payload_length) != VCM_OK) {
return -1;
@@ -660,6 +659,7 @@
}
int32_t ViEEncoder::UpdateProtectionMethod(bool enable_nack) {
+ DCHECK(send_payload_router_ != NULL);
bool fec_enabled = false;
uint8_t dummy_ptype_red = 0;
uint8_t dummy_ptypeFEC = 0;
@@ -693,7 +693,6 @@
// The send codec must be registered to set correct MTU.
webrtc::VideoCodec codec;
if (vcm_.SendCodec(&codec) == 0) {
- uint16_t max_pay_load = default_rtp_rtcp_->MaxDataPayloadLength();
uint32_t current_bitrate_bps = 0;
if (vcm_.Bitrate(¤t_bitrate_bps) != 0) {
LOG_F(LS_WARNING) <<
@@ -701,7 +700,9 @@
}
// Convert to start bitrate in kbps.
codec.startBitrate = (current_bitrate_bps + 500) / 1000;
- if (vcm_.RegisterSendCodec(&codec, number_of_cores_, max_pay_load) != 0) {
+ size_t max_payload_length = send_payload_router_->MaxPayloadLength();
+ if (vcm_.RegisterSendCodec(&codec, number_of_cores_,
+ max_payload_length) != 0) {
return -1;
}
}
