blob: d6cce69dbf09d7b6053abecd43b970819591acd0 [file] [log] [blame]
/*
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/audio/audio_receive_stream.h"
#include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
namespace webrtc {
const size_t kAbsoluteSendTimeLength = 4;
void BuildAbsoluteSendTimeExtension(uint8_t* buffer,
int id,
uint32_t abs_send_time) {
const size_t kRtpOneByteHeaderLength = 4;
const uint16_t kRtpOneByteHeaderExtensionId = 0xBEDE;
ByteWriter<uint16_t>::WriteBigEndian(buffer, kRtpOneByteHeaderExtensionId);
const uint32_t kPosLength = 2;
ByteWriter<uint16_t>::WriteBigEndian(buffer + kPosLength,
kAbsoluteSendTimeLength / 4);
const uint8_t kLengthOfData = 3;
buffer[kRtpOneByteHeaderLength] = (id << 4) + (kLengthOfData - 1);
ByteWriter<uint32_t, kLengthOfData>::WriteBigEndian(
buffer + kRtpOneByteHeaderLength + 1, abs_send_time);
}
size_t CreateRtpHeaderWithAbsSendTime(uint8_t* header,
int extension_id,
uint32_t abs_send_time) {
header[0] = 0x80; // Version 2.
header[0] |= 0x10; // Set extension bit.
header[1] = 100; // Payload type.
header[1] |= 0x80; // Marker bit is set.
ByteWriter<uint16_t>::WriteBigEndian(header + 2, 0x1234); // Sequence number.
ByteWriter<uint32_t>::WriteBigEndian(header + 4, 0x5678); // Timestamp.
ByteWriter<uint32_t>::WriteBigEndian(header + 8, 0x4321); // SSRC.
int32_t rtp_header_length = kRtpHeaderSize;
BuildAbsoluteSendTimeExtension(header + rtp_header_length, extension_id,
abs_send_time);
rtp_header_length += kAbsoluteSendTimeLength;
return rtp_header_length;
}
TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweWithTimestamp) {
MockRemoteBitrateEstimator rbe;
AudioReceiveStream::Config config;
config.combined_audio_video_bwe = true;
config.voe_channel_id = 1;
const int kAbsSendTimeId = 3;
config.rtp.extensions.push_back(
RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
internal::AudioReceiveStream recv_stream(&rbe, config);
uint8_t rtp_packet[30];
const int kAbsSendTimeValue = 1234;
CreateRtpHeaderWithAbsSendTime(rtp_packet, kAbsSendTimeId, kAbsSendTimeValue);
PacketTime packet_time(5678000, 0);
const size_t kExpectedHeaderLength = 20;
EXPECT_CALL(rbe, IncomingPacket(packet_time.timestamp / 1000,
sizeof(rtp_packet) - kExpectedHeaderLength,
testing::_, false))
.Times(1);
EXPECT_TRUE(
recv_stream.DeliverRtp(rtp_packet, sizeof(rtp_packet), packet_time));
}
} // namespace webrtc