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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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