Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(261)

Side by Side Diff: webrtc/video/send_statistics_proxy_unittest.cc

Issue 1273363005: Add send transports to individual webrtc::Call streams. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: rebase+comment Created 5 years, 3 months 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 unified diff | Download patch
« no previous file with comments | « webrtc/video/rtc_event_log_unittest.cc ('k') | webrtc/video/transport_adapter.cc » ('j') | no next file with comments »
Toggle Intra-line Diffs ('i') | Expand Comments ('e') | Collapse Comments ('c') | Show Comments Hide Comments ('s')
OLDNEW
1 /* 1 /*
2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
3 * 3 *
4 * Use of this source code is governed by a BSD-style license 4 * Use of this source code is governed by a BSD-style license
5 * that can be found in the LICENSE file in the root of the source 5 * that can be found in the LICENSE file in the root of the source
6 * tree. An additional intellectual property rights grant can be found 6 * tree. An additional intellectual property rights grant can be found
7 * in the file PATENTS. All contributing project authors may 7 * in the file PATENTS. All contributing project authors may
8 * be found in the AUTHORS file in the root of the source tree. 8 * be found in the AUTHORS file in the root of the source tree.
9 */ 9 */
10 10
11 // This file includes unit tests for SendStatisticsProxy. 11 // This file includes unit tests for SendStatisticsProxy.
12 #include "webrtc/video/send_statistics_proxy.h" 12 #include "webrtc/video/send_statistics_proxy.h"
13 13
14 #include <map> 14 #include <map>
15 #include <string> 15 #include <string>
16 #include <vector> 16 #include <vector>
17 17
18 #include "testing/gtest/include/gtest/gtest.h" 18 #include "testing/gtest/include/gtest/gtest.h"
19 19
20 namespace webrtc { 20 namespace webrtc {
21 21
22 class SendStatisticsProxyTest : public ::testing::Test { 22 class SendStatisticsProxyTest : public ::testing::Test {
23 public: 23 public:
24 SendStatisticsProxyTest() 24 SendStatisticsProxyTest()
25 : fake_clock_(1234), avg_delay_ms_(0), max_delay_ms_(0) {} 25 : fake_clock_(1234), config_(GetTestConfig()), avg_delay_ms_(0),
26 max_delay_ms_(0) {}
26 virtual ~SendStatisticsProxyTest() {} 27 virtual ~SendStatisticsProxyTest() {}
27 28
28 protected: 29 protected:
29 virtual void SetUp() { 30 virtual void SetUp() {
30 statistics_proxy_.reset( 31 statistics_proxy_.reset(
31 new SendStatisticsProxy(&fake_clock_, GetTestConfig())); 32 new SendStatisticsProxy(&fake_clock_, GetTestConfig()));
32 config_ = GetTestConfig();
33 expected_ = VideoSendStream::Stats(); 33 expected_ = VideoSendStream::Stats();
34 } 34 }
35 35
36 VideoSendStream::Config GetTestConfig() { 36 VideoSendStream::Config GetTestConfig() {
37 VideoSendStream::Config config; 37 VideoSendStream::Config config(nullptr);
38 config.rtp.ssrcs.push_back(17); 38 config.rtp.ssrcs.push_back(17);
39 config.rtp.ssrcs.push_back(42); 39 config.rtp.ssrcs.push_back(42);
40 config.rtp.rtx.ssrcs.push_back(18); 40 config.rtp.rtx.ssrcs.push_back(18);
41 config.rtp.rtx.ssrcs.push_back(43); 41 config.rtp.rtx.ssrcs.push_back(43);
42 return config; 42 return config;
43 } 43 }
44 44
45 void ExpectEqual(VideoSendStream::Stats one, VideoSendStream::Stats other) { 45 void ExpectEqual(VideoSendStream::Stats one, VideoSendStream::Stats other) {
46 EXPECT_EQ(one.input_frame_rate, other.input_frame_rate); 46 EXPECT_EQ(one.input_frame_rate, other.input_frame_rate);
47 EXPECT_EQ(one.encode_frame_rate, other.encode_frame_rate); 47 EXPECT_EQ(one.encode_frame_rate, other.encode_frame_rate);
(...skipping 350 matching lines...) Expand 10 before | Expand all | Expand 10 after
398 VideoSendStream::Stats stats = statistics_proxy_->GetStats(); 398 VideoSendStream::Stats stats = statistics_proxy_->GetStats();
399 EXPECT_EQ(static_cast<int>(bitrate.bitrate_bps), 399 EXPECT_EQ(static_cast<int>(bitrate.bitrate_bps),
400 stats.substreams[config_.rtp.ssrcs[0]].total_bitrate_bps); 400 stats.substreams[config_.rtp.ssrcs[0]].total_bitrate_bps);
401 EXPECT_EQ(static_cast<int>(bitrate.bitrate_bps), 401 EXPECT_EQ(static_cast<int>(bitrate.bitrate_bps),
402 stats.substreams[config_.rtp.ssrcs[0]].retransmit_bitrate_bps); 402 stats.substreams[config_.rtp.ssrcs[0]].retransmit_bitrate_bps);
403 EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[1]].total_bitrate_bps); 403 EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[1]].total_bitrate_bps);
404 EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[1]].retransmit_bitrate_bps); 404 EXPECT_EQ(0, stats.substreams[config_.rtp.ssrcs[1]].retransmit_bitrate_bps);
405 } 405 }
406 406
407 } // namespace webrtc 407 } // namespace webrtc
OLDNEW
« no previous file with comments | « webrtc/video/rtc_event_log_unittest.cc ('k') | webrtc/video/transport_adapter.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698