Chromium Code Reviews| Index: webrtc/api/rtcstatscollector_unittest.cc |
| diff --git a/webrtc/api/rtcstatscollector_unittest.cc b/webrtc/api/rtcstatscollector_unittest.cc |
| index e03238423870c6dbd2e34ab07a417d31e7077af4..fd57ad5970622276f9bdb8e6d15cddceb8ad8b15 100644 |
| --- a/webrtc/api/rtcstatscollector_unittest.cc |
| +++ b/webrtc/api/rtcstatscollector_unittest.cc |
| @@ -15,9 +15,10 @@ |
| #include <string> |
| #include <vector> |
| +#include "webrtc/api/jsepsessiondescription.h" |
| #include "webrtc/api/mediastream.h" |
| #include "webrtc/api/mediastreamtrack.h" |
| -#include "webrtc/api/jsepsessiondescription.h" |
| +#include "webrtc/api/rtpparameters.h" |
| #include "webrtc/api/stats/rtcstats_objects.h" |
| #include "webrtc/api/stats/rtcstatsreport.h" |
| #include "webrtc/api/test/mock_datachannel.h" |
| @@ -52,6 +53,10 @@ void PrintTo(const RTCCertificateStats& stats, ::std::ostream* os) { |
| *os << stats.ToString(); |
| } |
| +void PrintTo(const RTCCodecStats& stats, ::std::ostream* os) { |
| + *os << stats.ToString(); |
| +} |
| + |
| void PrintTo(const RTCDataChannelStats& stats, ::std::ostream* os) { |
| *os << stats.ToString(); |
| } |
| @@ -806,6 +811,117 @@ TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsSingle) { |
| ExpectReportContainsCertificateInfo(report, *remote_certinfo.get()); |
| } |
| +TEST_F(RTCStatsCollectorTest, CollectRTCCodecStats) { |
| + MockVoiceMediaChannel* voice_media_channel = new MockVoiceMediaChannel(); |
| + cricket::VoiceChannel voice_channel( |
| + test_->worker_thread(), test_->network_thread(), test_->media_engine(), |
| + voice_media_channel, nullptr, "VoiceContentName", false); |
| + |
| + MockVideoMediaChannel* video_media_channel = new MockVideoMediaChannel(); |
| + cricket::VideoChannel video_channel( |
| + test_->worker_thread(), test_->network_thread(), video_media_channel, |
| + nullptr, "VideoContentName", false); |
|
Taylor Brandstetter
2016/11/17 22:01:51
To test the scenario I described where a payload t
|
| + |
| + // Audio |
| + cricket::VoiceMediaInfo voice_media_info; |
| + |
| + RtpCodecParameters inbound_audio_codec; |
| + inbound_audio_codec.payload_type = 1; |
| + inbound_audio_codec.mime_type = "opus"; |
| + inbound_audio_codec.clock_rate = 1337; |
| + voice_media_info.receive_codecs.insert( |
| + std::make_pair(inbound_audio_codec.payload_type, inbound_audio_codec)); |
| + |
| + RtpCodecParameters outbound_audio_codec; |
| + outbound_audio_codec.payload_type = 2; |
| + outbound_audio_codec.mime_type = "isac"; |
| + outbound_audio_codec.clock_rate = 1338; |
| + voice_media_info.send_codecs.insert( |
| + std::make_pair(outbound_audio_codec.payload_type, outbound_audio_codec)); |
| + |
| + EXPECT_CALL(*voice_media_channel, GetStats(_)) |
| + .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true))); |
| + |
| + // Video |
| + cricket::VideoMediaInfo video_media_info; |
| + |
| + RtpCodecParameters inbound_video_codec; |
| + inbound_video_codec.payload_type = 3; |
| + inbound_video_codec.mime_type = "H264"; |
| + inbound_video_codec.clock_rate = 1339; |
| + video_media_info.receive_codecs.insert( |
| + std::make_pair(inbound_video_codec.payload_type, inbound_video_codec)); |
| + |
| + RtpCodecParameters outbound_video_codec; |
| + outbound_video_codec.payload_type = 4; |
| + outbound_video_codec.mime_type = "VP8"; |
| + outbound_video_codec.clock_rate = 1340; |
| + video_media_info.send_codecs.insert( |
| + std::make_pair(outbound_video_codec.payload_type, outbound_video_codec)); |
| + |
| + EXPECT_CALL(*video_media_channel, GetStats(_)) |
| + .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true))); |
| + |
| + SessionStats session_stats; |
| + session_stats.proxy_to_transport["VoiceContentName"] = "TransportName"; |
| + session_stats.proxy_to_transport["VideoContentName"] = "TransportName"; |
| + session_stats.transport_stats["TransportName"].transport_name = |
| + "TransportName"; |
| + |
| + EXPECT_CALL(test_->session(), GetTransportStats(_)) |
| + .WillRepeatedly(DoAll(SetArgPointee<0>(session_stats), Return(true))); |
| + EXPECT_CALL(test_->session(), voice_channel()) |
| + .WillRepeatedly(Return(&voice_channel)); |
| + EXPECT_CALL(test_->session(), video_channel()) |
| + .WillRepeatedly(Return(&video_channel)); |
| + |
| + rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport(); |
| + |
| + RTCCodecStats expected_inbound_audio_codec( |
| + "RTCCodec_InboundAudio_1", report->timestamp_us()); |
| + expected_inbound_audio_codec.payload_type = 1; |
| + expected_inbound_audio_codec.codec = "audio/opus"; |
| + expected_inbound_audio_codec.clock_rate = 1337; |
| + |
| + RTCCodecStats expected_outbound_audio_codec( |
| + "RTCCodec_OutboundAudio_2", report->timestamp_us()); |
| + expected_outbound_audio_codec.payload_type = 2; |
| + expected_outbound_audio_codec.codec = "audio/isac"; |
| + expected_outbound_audio_codec.clock_rate = 1338; |
| + |
| + RTCCodecStats expected_inbound_video_codec( |
| + "RTCCodec_InboundVideo_3", report->timestamp_us()); |
| + expected_inbound_video_codec.payload_type = 3; |
| + expected_inbound_video_codec.codec = "video/H264"; |
| + expected_inbound_video_codec.clock_rate = 1339; |
| + |
| + RTCCodecStats expected_outbound_video_codec( |
| + "RTCCodec_OutboundVideo_4", report->timestamp_us()); |
| + expected_outbound_video_codec.payload_type = 4; |
| + expected_outbound_video_codec.codec = "video/VP8"; |
| + expected_outbound_video_codec.clock_rate = 1340; |
| + |
| + ASSERT(report->Get(expected_inbound_audio_codec.id())); |
| + EXPECT_EQ(expected_inbound_audio_codec, |
| + report->Get(expected_inbound_audio_codec.id())->cast_to< |
| + RTCCodecStats>()); |
| + |
| + ASSERT(report->Get(expected_outbound_audio_codec.id())); |
| + EXPECT_EQ(expected_outbound_audio_codec, |
| + report->Get(expected_outbound_audio_codec.id())->cast_to< |
| + RTCCodecStats>()); |
| + |
| + ASSERT(report->Get(expected_inbound_video_codec.id())); |
| + EXPECT_EQ(expected_inbound_video_codec, |
| + report->Get(expected_inbound_video_codec.id())->cast_to< |
| + RTCCodecStats>()); |
| + |
| + ASSERT(report->Get(expected_outbound_video_codec.id())); |
| + EXPECT_EQ(expected_outbound_video_codec, |
| + report->Get(expected_outbound_video_codec.id())->cast_to< |
| + RTCCodecStats>()); |
| +} |
| + |
| TEST_F(RTCStatsCollectorTest, CollectRTCCertificateStatsMultiple) { |
| std::unique_ptr<CertificateInfo> audio_local_certinfo = |
| CreateFakeCertificateAndInfoFromDers( |
| @@ -1326,6 +1442,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) { |
| voice_media_info.receivers[0].local_stats[0].ssrc = 1; |
| voice_media_info.receivers[0].packets_rcvd = 2; |
| voice_media_info.receivers[0].bytes_rcvd = 3; |
| + voice_media_info.receivers[0].codec_payload_type = rtc::Optional<int>(42); |
| voice_media_info.receivers[0].jitter_ms = 4500; |
| voice_media_info.receivers[0].fraction_lost = 5.5f; |
| EXPECT_CALL(*voice_media_channel, GetStats(_)) |
| @@ -1356,6 +1473,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) { |
| expected_audio.media_type = "audio"; |
| expected_audio.transport_id = "RTCTransport_TransportName_" + |
| rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP); |
| + expected_audio.codec_id = "RTCCodec_InboundAudio_42"; |
| expected_audio.packets_received = 2; |
| expected_audio.bytes_received = 3; |
| expected_audio.jitter = 4.5; |
| @@ -1383,6 +1501,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) { |
| video_media_info.receivers[0].packets_rcvd = 2; |
| video_media_info.receivers[0].bytes_rcvd = 3; |
| video_media_info.receivers[0].fraction_lost = 4.5f; |
| + video_media_info.receivers[0].codec_payload_type = rtc::Optional<int>(42); |
| EXPECT_CALL(*video_media_channel, GetStats(_)) |
| .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true))); |
| @@ -1404,23 +1523,24 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) { |
| rtc::scoped_refptr<const RTCStatsReport> report = GetStatsReport(); |
| - RTCInboundRTPStreamStats expected_audio( |
| + RTCInboundRTPStreamStats expected_video( |
| "RTCInboundRTPVideoStream_1", report->timestamp_us()); |
| - expected_audio.ssrc = "1"; |
| - expected_audio.is_remote = false; |
| - expected_audio.media_type = "video"; |
| - expected_audio.transport_id = "RTCTransport_TransportName_" + |
| + expected_video.ssrc = "1"; |
| + expected_video.is_remote = false; |
| + expected_video.media_type = "video"; |
| + expected_video.transport_id = "RTCTransport_TransportName_" + |
| rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP); |
| - expected_audio.packets_received = 2; |
| - expected_audio.bytes_received = 3; |
| - expected_audio.fraction_lost = 4.5; |
| + expected_video.codec_id = "RTCCodec_InboundVideo_42"; |
|
hta-webrtc
2016/11/23 07:37:35
Nit: Since IDs are defined by the spec to be opaqu
hbos
2016/11/23 09:40:59
Done here and three other places. (in/out audio/vi
|
| + expected_video.packets_received = 2; |
| + expected_video.bytes_received = 3; |
| + expected_video.fraction_lost = 4.5; |
| - ASSERT(report->Get(expected_audio.id())); |
| + ASSERT(report->Get(expected_video.id())); |
| const RTCInboundRTPStreamStats& audio = report->Get( |
| - expected_audio.id())->cast_to<RTCInboundRTPStreamStats>(); |
| - EXPECT_EQ(audio, expected_audio); |
| + expected_video.id())->cast_to<RTCInboundRTPStreamStats>(); |
| + EXPECT_EQ(audio, expected_video); |
| - EXPECT_TRUE(report->Get(*expected_audio.transport_id)); |
| + EXPECT_TRUE(report->Get(*expected_video.transport_id)); |
| } |
| TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) { |
| @@ -1436,6 +1556,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) { |
| voice_media_info.senders[0].packets_sent = 2; |
| voice_media_info.senders[0].bytes_sent = 3; |
| voice_media_info.senders[0].rtt_ms = 4500; |
| + voice_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42); |
| EXPECT_CALL(*voice_media_channel, GetStats(_)) |
| .WillOnce(DoAll(SetArgPointee<0>(voice_media_info), Return(true))); |
| @@ -1464,6 +1585,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) { |
| expected_audio.media_type = "audio"; |
| expected_audio.transport_id = "RTCTransport_TransportName_" + |
| rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP); |
| + expected_audio.codec_id = "RTCCodec_OutboundAudio_42"; |
|
hta-webrtc
2016/11/23 07:37:35
As above.
hbos
2016/11/23 09:40:59
Done.
|
| expected_audio.packets_sent = 2; |
| expected_audio.bytes_sent = 3; |
| expected_audio.round_trip_time = 4.5; |
| @@ -1492,6 +1614,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) { |
| video_media_info.senders[0].packets_sent = 5; |
| video_media_info.senders[0].bytes_sent = 6; |
| video_media_info.senders[0].rtt_ms = 7500; |
| + video_media_info.senders[0].codec_payload_type = rtc::Optional<int>(42); |
| EXPECT_CALL(*video_media_channel, GetStats(_)) |
| .WillOnce(DoAll(SetArgPointee<0>(video_media_info), Return(true))); |
| @@ -1520,6 +1643,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) { |
| expected_video.media_type = "video"; |
| expected_video.transport_id = "RTCTransport_TransportName_" + |
| rtc::ToString<>(cricket::ICE_CANDIDATE_COMPONENT_RTP); |
| + expected_video.codec_id = "RTCCodec_OutboundVideo_42"; |
| expected_video.fir_count = 2; |
| expected_video.pli_count = 3; |
| expected_video.nack_count = 4; |