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Delete unused method PayloadRouter::MaxPayloadLength.

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Documentation was also unclear, it seems it returned the RTP packet
size including RTP headers.

BUG=None.

Review-Url: https://codereview.webrtc.org/2588343002
Cr-Commit-Position: refs/heads/master@{#15707}
This commit is contained in:
nisse 2016-12-20 03:12:04 -08:00 committed by Commit bot
parent 8bab796db7
commit 5206667dad
3 changed files with 0 additions and 56 deletions
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16
webrtc/video/payload_router.cc
View File

@ -98,11 +98,6 @@ PayloadRouter::PayloadRouter(const std::vector<RtpRtcp*>& rtp_modules,
PayloadRouter::~PayloadRouter() {}
size_t PayloadRouter::DefaultMaxPayloadLength() {
const size_t kIpUdpSrtpLength = 44;
return IP_PACKET_SIZE - kIpUdpSrtpLength;
}
void PayloadRouter::SetActive(bool active) {
rtc::CritScope lock(&crit_);
if (active_ == active)
@ -149,17 +144,6 @@ EncodedImageCallback::Result PayloadRouter::OnEncodedImage(
return Result(Result::OK, frame_id);
}
size_t PayloadRouter::MaxPayloadLength() const {
size_t min_payload_length = DefaultMaxPayloadLength();
rtc::CritScope lock(&crit_);
for (size_t i = 0; i < rtp_modules_.size(); ++i) {
size_t module_payload_length = rtp_modules_[i]->MaxDataPayloadLength();
if (module_payload_length < min_payload_length)
min_payload_length = module_payload_length;
}
return min_payload_length;
}
void PayloadRouter::OnBitrateAllocationUpdated(
const BitrateAllocation& bitrate) {
rtc::CritScope lock(&crit_);

6
webrtc/video/payload_router.h
View File

@ -36,8 +36,6 @@ class PayloadRouter : public EncodedImageCallback {
int payload_type);
~PayloadRouter();
static size_t DefaultMaxPayloadLength();
// PayloadRouter will only route packets if being active, all packets will be
// dropped otherwise.
void SetActive(bool active);
@ -50,10 +48,6 @@ class PayloadRouter : public EncodedImageCallback {
const CodecSpecificInfo* codec_specific_info,
const RTPFragmentationHeader* fragmentation) override;
// Returns the maximum allowed data payload length, given the configured MTU
// and RTP headers.
size_t MaxPayloadLength() const;
void OnBitrateAllocationUpdated(const BitrateAllocation& bitrate);
private:

34
webrtc/video/payload_router_unittest.cc
View File

@ -149,40 +149,6 @@ TEST(PayloadRouterTest, SendSimulcast) {
.error);
}
TEST(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;
NiceMock<MockRtpRtcp> rtp_1;
NiceMock<MockRtpRtcp> rtp_2;
std::vector<RtpRtcp*> modules;
modules.push_back(&rtp_1);
modules.push_back(&rtp_2);
PayloadRouter payload_router(modules, 42);
EXPECT_EQ(kDefaultMaxLength, PayloadRouter::DefaultMaxPayloadLength());
std::vector<VideoStream> streams(2);
// 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());
}
TEST(PayloadRouterTest, SimulcastTargetBitrate) {
NiceMock<MockRtpRtcp> rtp_1;
NiceMock<MockRtpRtcp> rtp_2;