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Side by Side Diff: webrtc/modules/rtp_rtcp/source/rtp_format_h264_unittest.cc

Issue 3007763002: Move array_view.h to webrtc/api/ (Closed)
Patch Set: rebase Created 3 years, 3 months ago
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1 /* 1 /*
2 * Copyright (c) 2014 The WebRTC project authors. All Rights Reserved. 2 * Copyright (c) 2014 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 #include <memory> 11 #include <memory>
12 #include <vector> 12 #include <vector>
13 13
14 #include "webrtc/api/array_view.h"
14 #include "webrtc/common_video/h264/h264_common.h" 15 #include "webrtc/common_video/h264/h264_common.h"
15 #include "webrtc/modules/include/module_common_types.h" 16 #include "webrtc/modules/include/module_common_types.h"
16 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" 17 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
17 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" 18 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
18 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" 19 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
19 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" 20 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
20 #include "webrtc/rtc_base/array_view.h"
21 #include "webrtc/test/gmock.h" 21 #include "webrtc/test/gmock.h"
22 #include "webrtc/test/gtest.h" 22 #include "webrtc/test/gtest.h"
23 23
24 namespace webrtc { 24 namespace webrtc {
25 namespace { 25 namespace {
26 26
27 using ::testing::ElementsAreArray; 27 using ::testing::ElementsAreArray;
28 28
29 constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr; 29 constexpr RtpPacketToSend::ExtensionManager* kNoExtensions = nullptr;
30 const size_t kMaxPayloadSize = 1200; 30 const size_t kMaxPayloadSize = 1200;
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937 EXPECT_EQ(kVideoFrameDelta, payload.frame_type); 937 EXPECT_EQ(kVideoFrameDelta, payload.frame_type);
938 EXPECT_EQ(kH264SingleNalu, h264.packetization_type); 938 EXPECT_EQ(kH264SingleNalu, h264.packetization_type);
939 EXPECT_EQ(kSei, h264.nalu_type); 939 EXPECT_EQ(kSei, h264.nalu_type);
940 ASSERT_EQ(1u, h264.nalus_length); 940 ASSERT_EQ(1u, h264.nalus_length);
941 EXPECT_EQ(static_cast<H264::NaluType>(kSei), h264.nalus[0].type); 941 EXPECT_EQ(static_cast<H264::NaluType>(kSei), h264.nalus[0].type);
942 EXPECT_EQ(-1, h264.nalus[0].sps_id); 942 EXPECT_EQ(-1, h264.nalus[0].sps_id);
943 EXPECT_EQ(-1, h264.nalus[0].pps_id); 943 EXPECT_EQ(-1, h264.nalus[0].pps_id);
944 } 944 }
945 945
946 } // namespace webrtc 946 } // namespace webrtc
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