diff options
Diffstat (limited to 'webrtc/call/rtc_event_log_unittest.cc')
-rw-r--r-- | webrtc/call/rtc_event_log_unittest.cc | 47 |
1 files changed, 22 insertions, 25 deletions
diff --git a/webrtc/call/rtc_event_log_unittest.cc b/webrtc/call/rtc_event_log_unittest.cc index 1e35733d2c..536b997227 100644 --- a/webrtc/call/rtc_event_log_unittest.cc +++ b/webrtc/call/rtc_event_log_unittest.cc @@ -84,10 +84,12 @@ MediaType GetRuntimeMediaType(rtclog::MediaType media_type) { return ::testing::AssertionFailure() << "Event of type " << type << " has " << (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet"; - if ((type == rtclog::Event::DEBUG_EVENT) != event.has_debug_event()) + if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) != + event.has_audio_playout_event()) return ::testing::AssertionFailure() << "Event of type " << type << " has " - << (event.has_debug_event() ? "" : "no ") << "debug event"; + << (event.has_audio_playout_event() ? "" : "no ") + << "audio_playout event"; if ((type == rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT) != event.has_video_receiver_config()) return ::testing::AssertionFailure() @@ -267,20 +269,15 @@ void VerifyRtcpEvent(const rtclog::Event& event, void VerifyPlayoutEvent(const rtclog::Event& event, uint32_t ssrc) { ASSERT_TRUE(IsValidBasicEvent(event)); - ASSERT_EQ(rtclog::Event::DEBUG_EVENT, event.type()); - const rtclog::DebugEvent& debug_event = event.debug_event(); - ASSERT_TRUE(debug_event.has_type()); - EXPECT_EQ(rtclog::DebugEvent::AUDIO_PLAYOUT, debug_event.type()); - ASSERT_TRUE(debug_event.has_local_ssrc()); - EXPECT_EQ(ssrc, debug_event.local_ssrc()); + ASSERT_EQ(rtclog::Event::AUDIO_PLAYOUT_EVENT, event.type()); + const rtclog::AudioPlayoutEvent& playout_event = event.audio_playout_event(); + ASSERT_TRUE(playout_event.has_local_ssrc()); + EXPECT_EQ(ssrc, playout_event.local_ssrc()); } void VerifyLogStartEvent(const rtclog::Event& event) { ASSERT_TRUE(IsValidBasicEvent(event)); - ASSERT_EQ(rtclog::Event::DEBUG_EVENT, event.type()); - const rtclog::DebugEvent& debug_event = event.debug_event(); - ASSERT_TRUE(debug_event.has_type()); - EXPECT_EQ(rtclog::DebugEvent::LOG_START, debug_event.type()); + EXPECT_EQ(rtclog::Event::LOG_START, event.type()); } /* @@ -399,12 +396,12 @@ void GenerateVideoSendConfig(uint32_t extensions_bitvector, // them back to see if they match. void LogSessionAndReadBack(size_t rtp_count, size_t rtcp_count, - size_t debug_count, + size_t playout_count, uint32_t extensions_bitvector, uint32_t csrcs_count, unsigned random_seed) { ASSERT_LE(rtcp_count, rtp_count); - ASSERT_LE(debug_count, rtp_count); + ASSERT_LE(playout_count, rtp_count); std::vector<rtc::Buffer> rtp_packets; std::vector<rtc::Buffer> rtcp_packets; std::vector<size_t> rtp_header_sizes; @@ -429,8 +426,8 @@ void LogSessionAndReadBack(size_t rtp_count, rtcp_packets.push_back(rtc::Buffer(packet_size)); GenerateRtcpPacket(rtcp_packets[i].data(), packet_size); } - // Create debug_count random SSRCs to use when logging AudioPlayout events. - for (size_t i = 0; i < debug_count; i++) { + // Create playout_count random SSRCs to use when logging AudioPlayout events. + for (size_t i = 0; i < playout_count; i++) { playout_ssrcs.push_back(static_cast<uint32_t>(rand())); } // Create configurations for the video streams. @@ -450,7 +447,7 @@ void LogSessionAndReadBack(size_t rtp_count, rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create()); log_dumper->LogVideoReceiveStreamConfig(receiver_config); log_dumper->LogVideoSendStreamConfig(sender_config); - size_t rtcp_index = 1, debug_index = 1; + size_t rtcp_index = 1, playout_index = 1; for (size_t i = 1; i <= rtp_count; i++) { log_dumper->LogRtpHeader( (i % 2 == 0), // Every second packet is incoming. @@ -464,9 +461,9 @@ void LogSessionAndReadBack(size_t rtp_count, rtcp_packets[rtcp_index - 1].size()); rtcp_index++; } - if (i * debug_count >= debug_index * rtp_count) { - log_dumper->LogAudioPlayout(playout_ssrcs[debug_index - 1]); - debug_index++; + if (i * playout_count >= playout_index * rtp_count) { + log_dumper->LogAudioPlayout(playout_ssrcs[playout_index - 1]); + playout_index++; } if (i == rtp_count / 2) { log_dumper->StartLogging(temp_filename, 10000000); @@ -481,11 +478,11 @@ void LogSessionAndReadBack(size_t rtp_count, // Verify the result. const int event_count = - config_count + debug_count + rtcp_count + rtp_count + 1; + config_count + playout_count + rtcp_count + rtp_count + 1; EXPECT_EQ(event_count, parsed_stream.stream_size()); VerifyReceiveStreamConfig(parsed_stream.stream(0), receiver_config); VerifySendStreamConfig(parsed_stream.stream(1), sender_config); - size_t event_index = config_count, rtcp_index = 1, debug_index = 1; + size_t event_index = config_count, rtcp_index = 1, playout_index = 1; for (size_t i = 1; i <= rtp_count; i++) { VerifyRtpEvent(parsed_stream.stream(event_index), (i % 2 == 0), // Every second packet is incoming. @@ -502,11 +499,11 @@ void LogSessionAndReadBack(size_t rtp_count, event_index++; rtcp_index++; } - if (i * debug_count >= debug_index * rtp_count) { + if (i * playout_count >= playout_index * rtp_count) { VerifyPlayoutEvent(parsed_stream.stream(event_index), - playout_ssrcs[debug_index - 1]); + playout_ssrcs[playout_index - 1]); event_index++; - debug_index++; + playout_index++; } if (i == rtp_count / 2) { VerifyLogStartEvent(parsed_stream.stream(event_index)); |