Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(1183)

Unified Diff: webrtc/video_engine/payload_router_unittest.cc

Issue 1506773002: Merge webrtc/video_engine/ into webrtc/video/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase Created 5 years ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
« no previous file with comments | « webrtc/video_engine/payload_router.cc ('k') | webrtc/video_engine/report_block_stats.h » ('j') | no next file with comments »
Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
Index: webrtc/video_engine/payload_router_unittest.cc
diff --git a/webrtc/video_engine/payload_router_unittest.cc b/webrtc/video_engine/payload_router_unittest.cc
deleted file mode 100644
index 11c664b4a1fa0616ccabb5205d9e6c8508191474..0000000000000000000000000000000000000000
--- a/webrtc/video_engine/payload_router_unittest.cc
+++ /dev/null
@@ -1,209 +0,0 @@
-/*
- * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include <list>
-
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
-#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
-#include "webrtc/video_engine/payload_router.h"
-
-using ::testing::_;
-using ::testing::AnyNumber;
-using ::testing::NiceMock;
-using ::testing::Return;
-
-namespace webrtc {
-
-class PayloadRouterTest : public ::testing::Test {
- protected:
- virtual void SetUp() {
- payload_router_.reset(new PayloadRouter());
- }
- rtc::scoped_ptr<PayloadRouter> payload_router_;
-};
-
-TEST_F(PayloadRouterTest, SendOnOneModule) {
- MockRtpRtcp rtp;
- std::list<RtpRtcp*> modules(1, &rtp);
-
- payload_router_->SetSendingRtpModules(modules);
-
- uint8_t payload = 'a';
- FrameType frame_type = kVideoFrameKey;
- int8_t payload_type = 96;
-
- EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
- NULL))
- .Times(0);
- EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
- &payload, 1, NULL, NULL));
-
- payload_router_->set_active(true);
- EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
- NULL))
- .Times(1);
- EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
- &payload, 1, NULL, NULL));
-
- payload_router_->set_active(false);
- EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
- NULL))
- .Times(0);
- EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
- &payload, 1, NULL, NULL));
-
- payload_router_->set_active(true);
- EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
- NULL))
- .Times(1);
- EXPECT_TRUE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
- &payload, 1, NULL, NULL));
-
- modules.clear();
- payload_router_->SetSendingRtpModules(modules);
- EXPECT_CALL(rtp, SendOutgoingData(frame_type, payload_type, 0, 0, _, 1, NULL,
- NULL))
- .Times(0);
- EXPECT_FALSE(payload_router_->RoutePayload(frame_type, payload_type, 0, 0,
- &payload, 1, NULL, NULL));
-}
-
-TEST_F(PayloadRouterTest, SendSimulcast) {
- MockRtpRtcp rtp_1;
- MockRtpRtcp rtp_2;
- std::list<RtpRtcp*> modules;
- modules.push_back(&rtp_1);
- modules.push_back(&rtp_2);
-
- payload_router_->SetSendingRtpModules(modules);
-
- uint8_t payload_1 = 'a';
- FrameType frame_type_1 = kVideoFrameKey;
- int8_t payload_type_1 = 96;
- RTPVideoHeader rtp_hdr_1;
- rtp_hdr_1.simulcastIdx = 0;
-
- payload_router_->set_active(true);
- EXPECT_CALL(rtp_1, SendOutgoingData(frame_type_1, payload_type_1, 0, 0, _, 1,
- NULL, &rtp_hdr_1))
- .Times(1);
- EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
- .Times(0);
- EXPECT_TRUE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
- &payload_1, 1, NULL, &rtp_hdr_1));
-
- uint8_t payload_2 = 'b';
- FrameType frame_type_2 = kVideoFrameDelta;
- int8_t payload_type_2 = 97;
- RTPVideoHeader rtp_hdr_2;
- rtp_hdr_2.simulcastIdx = 1;
- EXPECT_CALL(rtp_2, SendOutgoingData(frame_type_2, payload_type_2, 0, 0, _, 1,
- NULL, &rtp_hdr_2))
- .Times(1);
- EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
- .Times(0);
- EXPECT_TRUE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
- &payload_2, 1, NULL, &rtp_hdr_2));
-
- // Inactive.
- payload_router_->set_active(false);
- EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
- .Times(0);
- EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
- .Times(0);
- EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
- &payload_1, 1, NULL, &rtp_hdr_1));
- EXPECT_FALSE(payload_router_->RoutePayload(frame_type_2, payload_type_2, 0, 0,
- &payload_2, 1, NULL, &rtp_hdr_2));
-
- // Invalid simulcast index.
- payload_router_->set_active(true);
- EXPECT_CALL(rtp_1, SendOutgoingData(_, _, _, _, _, _, _, _))
- .Times(0);
- EXPECT_CALL(rtp_2, SendOutgoingData(_, _, _, _, _, _, _, _))
- .Times(0);
- rtp_hdr_1.simulcastIdx = 2;
- EXPECT_FALSE(payload_router_->RoutePayload(frame_type_1, payload_type_1, 0, 0,
- &payload_1, 1, NULL, &rtp_hdr_1));
-}
-
-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());
-}
-
-TEST_F(PayloadRouterTest, SetTargetSendBitrates) {
- MockRtpRtcp rtp_1;
- MockRtpRtcp rtp_2;
- std::list<RtpRtcp*> modules;
- modules.push_back(&rtp_1);
- modules.push_back(&rtp_2);
- payload_router_->SetSendingRtpModules(modules);
-
- const uint32_t bitrate_1 = 10000;
- const uint32_t bitrate_2 = 76543;
- std::vector<uint32_t> bitrates(2, bitrate_1);
- bitrates[1] = bitrate_2;
- EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
- .Times(1);
- EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
- .Times(1);
- payload_router_->SetTargetSendBitrates(bitrates);
-
- bitrates.resize(1);
- EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
- .Times(0);
- EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
- .Times(0);
- payload_router_->SetTargetSendBitrates(bitrates);
-
- bitrates.resize(3);
- bitrates[1] = bitrate_2;
- bitrates[2] = bitrate_1 + bitrate_2;
- EXPECT_CALL(rtp_1, SetTargetSendBitrate(bitrate_1))
- .Times(1);
- EXPECT_CALL(rtp_2, SetTargetSendBitrate(bitrate_2))
- .Times(1);
- payload_router_->SetTargetSendBitrates(bitrates);
-}
-} // namespace webrtc
« no previous file with comments | « webrtc/video_engine/payload_router.cc ('k') | webrtc/video_engine/report_block_stats.h » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698