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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/base/array_view.h" | |
15 #include "webrtc/common_video/h264/h264_common.h" | 14 #include "webrtc/common_video/h264/h264_common.h" |
16 #include "webrtc/modules/include/module_common_types.h" | 15 #include "webrtc/modules/include/module_common_types.h" |
17 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" | 16 #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" |
18 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | 17 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" |
19 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" | 18 #include "webrtc/modules/rtp_rtcp/source/rtp_format.h" |
20 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h" | 19 #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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