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1 /* | |
2 * Copyright (c) 2015 The WebRTC project authors. All Rights Reserved. | |
3 * | |
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 | |
6 * tree. An additional intellectual property rights grant can be found | |
7 * in the file PATENTS. All contributing project authors may | |
8 * be found in the AUTHORS file in the root of the source tree. | |
9 */ | |
10 | |
11 #include "testing/gtest/include/gtest/gtest.h" | |
12 | |
13 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitra
te_estimator.h" | |
14 #include "webrtc/modules/rtp_rtcp/source/byte_io.h" | |
15 #include "webrtc/video/audio_receive_stream.h" | |
16 | |
17 namespace webrtc { | |
18 | |
19 const size_t kAbsoluteSendTimeLength = 4; | |
20 | |
21 void BuildAbsoluteSendTimeExtension(uint8_t* buffer, | |
22 int id, | |
23 uint32_t abs_send_time) { | |
24 const size_t kRtpOneByteHeaderLength = 4; | |
25 const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE; | |
26 ByteWriter<uint16_t>::WriteBigEndian(buffer, kRtpOneByteHeaderExtensionId); | |
27 | |
28 const uint32_t kPosLength = 2; | |
29 ByteWriter<uint16_t>::WriteBigEndian(buffer + kPosLength, | |
30 kAbsoluteSendTimeLength / 4); | |
31 | |
32 const uint8_t kLengthOfData = 3; | |
33 buffer[kRtpOneByteHeaderLength] = (id << 4) + (kLengthOfData - 1); | |
34 ByteWriter<uint32_t, kLengthOfData>::WriteBigEndian( | |
35 buffer + kRtpOneByteHeaderLength + 1, abs_send_time); | |
36 } | |
37 | |
38 size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header, | |
39 int extension_id, | |
40 uint32_t abs_send_time) { | |
41 header[0] = 0x80; // Version 2. | |
42 header[0] |= 0x10; // Set extension bit. | |
43 header[1] = 100; // Payload type. | |
44 header[1] |= 0x80; // Marker bit is set. | |
45 ByteWriter<uint16_t>::WriteBigEndian(header + 2, 0x1234); // Sequence number. | |
46 ByteWriter<uint32_t>::WriteBigEndian(header + 4, 0x5678); // Timestamp. | |
47 ByteWriter<uint32_t>::WriteBigEndian(header + 8, 0x4321); // SSRC. | |
48 int32_t rtp_header_length = kRtpHeaderSize; | |
49 | |
50 BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id, | |
51 abs_send_time); | |
52 rtp_header_length += kAbsoluteSendTimeLength; | |
53 return rtp_header_length; | |
54 } | |
55 | |
56 TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) { | |
57 MockRemoteBitrateEstimator rbe; | |
58 AudioReceiveStream::Config config; | |
59 config.combined_audio_video_bwe = true; | |
60 config.voe_channel_id = 1; | |
61 const int kAbsSendTimeId = 3; | |
62 config.rtp.extensions.push_back( | |
63 RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId)); | |
64 internal::AudioReceiveStream recv_stream(&rbe, config); | |
65 uint8_t rtp_packet[30]; | |
66 const int kAbsSendTimeValue = 1234; | |
67 CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue); | |
68 PacketTime packet_time(5678000, 0); | |
69 const size_t kExpectedHeaderLength = 20; | |
70 EXPECT_CALL(rbe, IncomingPacket(packet_time.timestamp / 1000, | |
71 sizeof(rtp_packet) - kExpectedHeaderLength, | |
72 testing::_, false)) | |
73 .Times(1); | |
74 EXPECT_TRUE( | |
75 recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time)); | |
76 } | |
77 } // namespace webrtc | |
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