| /* |
| * Copyright (c) 2019 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 <vector> |
| |
| #include "api/transport/field_trial_based_config.h" |
| #include "modules/rtp_rtcp/include/rtp_header_extension_map.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "modules/rtp_rtcp/source/rtp_header_extensions.h" |
| #include "modules/rtp_rtcp/source/rtp_packet_received.h" |
| #include "modules/rtp_rtcp/source/rtp_sender.h" |
| #include "modules/rtp_rtcp/source/rtp_sender_audio.h" |
| #include "test/gmock.h" |
| #include "test/gtest.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| enum : int { // The first valid value is 1. |
| kAudioLevelExtensionId = 1, |
| }; |
| |
| const uint16_t kSeqNum = 33; |
| const uint32_t kSsrc = 725242; |
| const uint8_t kAudioLevel = 0x5a; |
| const uint64_t kStartTime = 123456789; |
| |
| using ::testing::_; |
| using ::testing::ElementsAreArray; |
| |
| class LoopbackTransportTest : public webrtc::Transport { |
| public: |
| LoopbackTransportTest() { |
| receivers_extensions_.Register(kRtpExtensionAudioLevel, |
| kAudioLevelExtensionId); |
| } |
| |
| bool SendRtp(const uint8_t* data, |
| size_t len, |
| const PacketOptions& /*options*/) override { |
| sent_packets_.push_back(RtpPacketReceived(&receivers_extensions_)); |
| EXPECT_TRUE(sent_packets_.back().Parse(data, len)); |
| return true; |
| } |
| bool SendRtcp(const uint8_t* data, size_t len) override { return false; } |
| const RtpPacketReceived& last_sent_packet() { return sent_packets_.back(); } |
| int packets_sent() { return sent_packets_.size(); } |
| |
| private: |
| RtpHeaderExtensionMap receivers_extensions_; |
| std::vector<RtpPacketReceived> sent_packets_; |
| }; |
| |
| } // namespace |
| |
| class RtpSenderAudioTest : public ::testing::Test { |
| public: |
| RtpSenderAudioTest() |
| : fake_clock_(kStartTime), |
| rtp_sender_([&] { |
| RtpRtcp::Configuration config; |
| config.audio = true; |
| config.clock = &fake_clock_; |
| config.outgoing_transport = &transport_; |
| config.media_send_ssrc = kSsrc; |
| return config; |
| }()), |
| rtp_sender_audio_(&fake_clock_, &rtp_sender_) { |
| rtp_sender_.SetSequenceNumber(kSeqNum); |
| } |
| |
| SimulatedClock fake_clock_; |
| LoopbackTransportTest transport_; |
| RTPSender rtp_sender_; |
| RTPSenderAudio rtp_sender_audio_; |
| }; |
| |
| TEST_F(RtpSenderAudioTest, SendAudio) { |
| const char payload_name[] = "PAYLOAD_NAME"; |
| const uint8_t payload_type = 127; |
| ASSERT_EQ(0, rtp_sender_audio_.RegisterAudioPayload( |
| payload_name, payload_type, 48000, 0, 1500)); |
| uint8_t payload[] = {47, 11, 32, 93, 89}; |
| |
| ASSERT_TRUE(rtp_sender_audio_.SendAudio(AudioFrameType::kAudioFrameCN, |
| payload_type, 4321, payload, |
| sizeof(payload))); |
| |
| auto sent_payload = transport_.last_sent_packet().payload(); |
| EXPECT_THAT(sent_payload, ElementsAreArray(payload)); |
| } |
| |
| TEST_F(RtpSenderAudioTest, SendAudioWithAudioLevelExtension) { |
| EXPECT_EQ(0, rtp_sender_audio_.SetAudioLevel(kAudioLevel)); |
| EXPECT_EQ(0, rtp_sender_.RegisterRtpHeaderExtension(kRtpExtensionAudioLevel, |
| kAudioLevelExtensionId)); |
| |
| const char payload_name[] = "PAYLOAD_NAME"; |
| const uint8_t payload_type = 127; |
| ASSERT_EQ(0, rtp_sender_audio_.RegisterAudioPayload( |
| payload_name, payload_type, 48000, 0, 1500)); |
| |
| uint8_t payload[] = {47, 11, 32, 93, 89}; |
| |
| ASSERT_TRUE(rtp_sender_audio_.SendAudio(AudioFrameType::kAudioFrameCN, |
| payload_type, 4321, payload, |
| sizeof(payload))); |
| |
| auto sent_payload = transport_.last_sent_packet().payload(); |
| EXPECT_THAT(sent_payload, ElementsAreArray(payload)); |
| // Verify AudioLevel extension. |
| bool voice_activity; |
| uint8_t audio_level; |
| EXPECT_TRUE(transport_.last_sent_packet().GetExtension<AudioLevel>( |
| &voice_activity, &audio_level)); |
| EXPECT_EQ(kAudioLevel, audio_level); |
| EXPECT_FALSE(voice_activity); |
| } |
| |
| // As RFC4733, named telephone events are carried as part of the audio stream |
| // and must use the same sequence number and timestamp base as the regular |
| // audio channel. |
| // This test checks the marker bit for the first packet and the consequent |
| // packets of the same telephone event. Since it is specifically for DTMF |
| // events, ignoring audio packets and sending kEmptyFrame instead of those. |
| TEST_F(RtpSenderAudioTest, CheckMarkerBitForTelephoneEvents) { |
| const char* kDtmfPayloadName = "telephone-event"; |
| const uint32_t kPayloadFrequency = 8000; |
| const uint8_t kPayloadType = 126; |
| ASSERT_EQ(0, rtp_sender_audio_.RegisterAudioPayload( |
| kDtmfPayloadName, kPayloadType, kPayloadFrequency, 0, 0)); |
| // For Telephone events, payload is not added to the registered payload list, |
| // it will register only the payload used for audio stream. |
| // Registering the payload again for audio stream with different payload name. |
| const char* kPayloadName = "payload_name"; |
| ASSERT_EQ(0, rtp_sender_audio_.RegisterAudioPayload( |
| kPayloadName, kPayloadType, kPayloadFrequency, 1, 0)); |
| // Start time is arbitrary. |
| uint32_t capture_timestamp = fake_clock_.TimeInMilliseconds(); |
| // DTMF event key=9, duration=500 and attenuationdB=10 |
| rtp_sender_audio_.SendTelephoneEvent(9, 500, 10); |
| // During start, it takes the starting timestamp as last sent timestamp. |
| // The duration is calculated as the difference of current and last sent |
| // timestamp. So for first call it will skip since the duration is zero. |
| ASSERT_TRUE(rtp_sender_audio_.SendAudio(AudioFrameType::kEmptyFrame, |
| kPayloadType, capture_timestamp, |
| nullptr, 0)); |
| // DTMF Sample Length is (Frequency/1000) * Duration. |
| // So in this case, it is (8000/1000) * 500 = 4000. |
| // Sending it as two packets. |
| ASSERT_TRUE( |
| rtp_sender_audio_.SendAudio(AudioFrameType::kEmptyFrame, kPayloadType, |
| capture_timestamp + 2000, nullptr, 0)); |
| |
| // Marker Bit should be set to 1 for first packet. |
| EXPECT_TRUE(transport_.last_sent_packet().Marker()); |
| |
| ASSERT_TRUE( |
| rtp_sender_audio_.SendAudio(AudioFrameType::kEmptyFrame, kPayloadType, |
| capture_timestamp + 4000, nullptr, 0)); |
| // Marker Bit should be set to 0 for rest of the packets. |
| EXPECT_FALSE(transport_.last_sent_packet().Marker()); |
| } |
| |
| } // namespace webrtc |