9fea80f50d
Add audio send and receive streams to CallTest and call the necessary voice engine APIs for the streams to be usable. Verifies the implementation by adding a simple test which monitors outgoing packets and checks that both audio and video is being sent with transport sequence numbers. Audio streams are using a fake audio device with file input. The CallTest implementation is to a big degree based on call_perf_tests.cc and should in the future replace a lot of that code. R=pbos@webrtc.org TBR=kjellander@webrtc.org BUG=webrtc:5263 Review URL: https://codereview.webrtc.org/1542653002 . Cr-Commit-Position: refs/heads/master@{#11171}
92 lines
3.0 KiB
C++
92 lines
3.0 KiB
C++
/*
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* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/test/call_test.h"
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#include "webrtc/test/null_transport.h"
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namespace webrtc {
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class PacketInjectionTest : public test::CallTest {
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protected:
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enum class CodecType {
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kVp8,
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kH264,
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};
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PacketInjectionTest() : rtp_header_parser_(RtpHeaderParser::Create()) {}
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void InjectIncorrectPacket(CodecType codec_type,
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uint8_t packet_type,
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const uint8_t* packet,
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size_t length);
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rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
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};
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void PacketInjectionTest::InjectIncorrectPacket(CodecType codec_type,
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uint8_t payload_type,
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const uint8_t* packet,
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size_t length) {
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CreateSenderCall(Call::Config());
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CreateReceiverCall(Call::Config());
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test::NullTransport null_transport;
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CreateSendConfig(1, 0, &null_transport);
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CreateMatchingReceiveConfigs(&null_transport);
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video_receive_configs_[0].decoders[0].payload_type = payload_type;
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switch (codec_type) {
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case CodecType::kVp8:
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video_receive_configs_[0].decoders[0].payload_name = "VP8";
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break;
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case CodecType::kH264:
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video_receive_configs_[0].decoders[0].payload_name = "H264";
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break;
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}
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CreateVideoStreams();
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RTPHeader header;
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EXPECT_TRUE(rtp_header_parser_->Parse(packet, length, &header));
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EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc)
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<< "Packet should have configured SSRC to not be dropped early.";
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EXPECT_EQ(payload_type, header.payloadType);
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Start();
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EXPECT_EQ(PacketReceiver::DELIVERY_PACKET_ERROR,
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receiver_call_->Receiver()->DeliverPacket(MediaType::VIDEO, packet,
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length, PacketTime()));
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Stop();
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DestroyStreams();
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}
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TEST_F(PacketInjectionTest, StapAPacketWithTruncatedNalUnits) {
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const uint8_t kPacket[] = {0x80,
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0xE5,
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0xE6,
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0x0,
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0x0,
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0xED,
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0x23,
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0x4,
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0x00,
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0xC0,
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0xFF,
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0xED,
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0x58,
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0xCB,
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0xED,
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0xDF};
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InjectIncorrectPacket(CodecType::kH264, 101, kPacket, sizeof(kPacket));
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}
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} // namespace webrtc
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