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stats: rename RTCInboundRTPStreamStats and RTCOutboundRTPStreamStats

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to RTCInboundRtpStreamStats and RTCOutboundRtpStreamStats respectively
which follows the camel-casing convention used elsewhere.

The old name is kept around as an alias for a limited amount of time
to allow upgrading dependencies.

BUG=webrtc:14973

Change-Id: Ibf4e65933fd6cc2e7e89955042f6f8fb0f6c7853
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/296261
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#39497}
This commit is contained in:
Philipp Hancke 2023-03-07 09:53:59 +01:00 committed by WebRTC LUCI CQ
parent ffdecdce2c
commit 1f98b466b8
18 changed files with 151 additions and 147 deletions
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20
api/stats/rtcstats_objects.h
View File

@ -406,14 +406,14 @@ class RTC_EXPORT RTCSentRtpStreamStats : public RTCRtpStreamStats {
};
// https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*
class RTC_EXPORT RTCInboundRTPStreamStats final
class RTC_EXPORT RTCInboundRtpStreamStats final
: public RTCReceivedRtpStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCInboundRTPStreamStats(std::string id, Timestamp timestamp);
RTCInboundRTPStreamStats(const RTCInboundRTPStreamStats& other);
~RTCInboundRTPStreamStats() override;
RTCInboundRtpStreamStats(std::string id, Timestamp timestamp);
RTCInboundRtpStreamStats(const RTCInboundRtpStreamStats& other);
~RTCInboundRtpStreamStats() override;
// TODO(https://crbug.com/webrtc/14174): Implement trackIdentifier and kind.
@ -493,16 +493,18 @@ class RTC_EXPORT RTCInboundRTPStreamStats final
// The former googMinPlayoutDelayMs (in seconds).
RTCNonStandardStatsMember<double> min_playout_delay;
};
// TODO(bugs.webrtc.org/14973): remove name alias.
using RTCInboundRTPStreamStats = RTCInboundRtpStreamStats;
// https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*
class RTC_EXPORT RTCOutboundRTPStreamStats final
class RTC_EXPORT RTCOutboundRtpStreamStats final
: public RTCSentRtpStreamStats {
public:
WEBRTC_RTCSTATS_DECL();
RTCOutboundRTPStreamStats(std::string id, Timestamp timestamp);
RTCOutboundRTPStreamStats(const RTCOutboundRTPStreamStats& other);
~RTCOutboundRTPStreamStats() override;
RTCOutboundRtpStreamStats(std::string id, Timestamp timestamp);
RTCOutboundRtpStreamStats(const RTCOutboundRtpStreamStats& other);
~RTCOutboundRtpStreamStats() override;
RTCStatsMember<std::string> media_source_id;
RTCStatsMember<std::string> remote_id;
@ -544,6 +546,8 @@ class RTC_EXPORT RTCOutboundRTPStreamStats final
power_efficient_encoder;
RTCStatsMember<std::string> scalability_mode;
};
// TODO(bugs.webrtc.org/14973): remove name alias.
using RTCOutboundRTPStreamStats = RTCOutboundRtpStreamStats;
// https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*
class RTC_EXPORT RTCRemoteInboundRtpStreamStats final

4
pc/peer_connection_field_trial_tests.cc
View File

@ -266,8 +266,8 @@ TEST_F(PeerConnectionFieldTrialTest, ApplyFakeNetworkConfig) {
// Send packets for kDefaultTimeoutMs
WAIT(false, kDefaultTimeoutMs);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtp_stats =
caller->GetStats()->GetStatsOfType<RTCOutboundRTPStreamStats>();
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtp_stats =
caller->GetStats()->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_GE(outbound_rtp_stats.size(), 1u);
ASSERT_TRUE(outbound_rtp_stats[0]->target_bitrate.is_defined());
// Link capacity is limited to 500k, so BWE is expected to be close to 500k.

20
pc/peer_connection_integrationtest.cc
View File

@ -1350,7 +1350,7 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
caller()->NewGetStats();
ASSERT_TRUE(caller_report);
auto outbound_stream_stats =
caller_report->GetStatsOfType<webrtc::RTCOutboundRTPStreamStats>();
caller_report->GetStatsOfType<webrtc::RTCOutboundRtpStreamStats>();
ASSERT_EQ(outbound_stream_stats.size(), 4u);
std::vector<std::string> outbound_track_ids;
for (const auto& stat : outbound_stream_stats) {
@ -1375,7 +1375,7 @@ TEST_P(PeerConnectionIntegrationTest, NewGetStatsManyAudioAndManyVideoStreams) {
callee()->NewGetStats();
ASSERT_TRUE(callee_report);
auto inbound_stream_stats =
callee_report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
callee_report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
ASSERT_EQ(4u, inbound_stream_stats.size());
std::vector<std::string> inbound_track_ids;
for (const auto& stat : inbound_stream_stats) {
@ -1414,7 +1414,7 @@ TEST_P(PeerConnectionIntegrationTest,
callee()->NewGetStats();
ASSERT_NE(nullptr, report);
auto inbound_stream_stats =
report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
ASSERT_EQ(1U, inbound_stream_stats.size());
ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined());
ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U);
@ -1463,7 +1463,7 @@ TEST_P(PeerConnectionIntegrationTest,
ASSERT_NE(nullptr, report);
auto inbound_rtps =
report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
auto index = FindFirstMediaStatsIndexByKind("audio", inbound_rtps);
ASSERT_GE(index, 0);
EXPECT_TRUE(inbound_rtps[index]->audio_level.is_defined());
@ -1483,7 +1483,7 @@ void ModifySsrcs(cricket::SessionDescription* desc) {
// Test that the "DEPRECATED_RTCMediaStreamTrackStats" object is updated
// correctly when SSRCs are unsignaled, and the SSRC of the received (audio)
// stream changes. This should result in two "RTCInboundRTPStreamStats", but
// stream changes. This should result in two "RTCInboundRtpStreamStats", but
// only one "DEPRECATED_RTCMediaStreamTrackStats", whose counters go up
// continuously rather than being reset to 0 once the SSRC change occurs.
//
@ -1558,13 +1558,13 @@ TEST_P(PeerConnectionIntegrationTest,
*track_stats[0]->total_samples_received *
kAcceptableConcealedSamplesPercentage);
// Also ensure that we have two "RTCInboundRTPStreamStats" as expected, as a
// Also ensure that we have two "RTCInboundRtpStreamStats" as expected, as a
// sanity check that the SSRC really changed.
// TODO(deadbeef): This isn't working right now, because we're not returning
// *any* stats for the inactive stream. Uncomment when the bug is completely
// fixed.
// auto inbound_stream_stats =
// report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
// report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
// ASSERT_EQ(2U, inbound_stream_stats.size());
}
@ -2989,7 +2989,7 @@ TEST_P(PeerConnectionIntegrationTest, DisableAndEnableAudioPlayout) {
double GetAudioEnergyStat(PeerConnectionIntegrationWrapper* pc) {
auto report = pc->NewGetStats();
auto inbound_rtps =
report->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>();
report->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>();
RTC_CHECK(!inbound_rtps.empty());
auto* inbound_rtp = inbound_rtps[0];
if (!inbound_rtp->total_audio_energy.is_defined()) {
@ -3811,7 +3811,7 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndRtpSenderVideoEncoderSelector) {
int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) {
rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
auto sender_stats = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
auto sender_stats = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
if (sender_stats.size() != 1) {
ADD_FAILURE();
return 0;
@ -3824,7 +3824,7 @@ int NacksReceivedCount(PeerConnectionIntegrationWrapper& pc) {
int NacksSentCount(PeerConnectionIntegrationWrapper& pc) {
rtc::scoped_refptr<const webrtc::RTCStatsReport> report = pc.NewGetStats();
auto receiver_stats = report->GetStatsOfType<RTCInboundRTPStreamStats>();
auto receiver_stats = report->GetStatsOfType<RTCInboundRtpStreamStats>();
if (receiver_stats.size() != 1) {
ADD_FAILURE();
return 0;

48
pc/peer_connection_simulcast_unittest.cc
View File

@ -138,7 +138,7 @@ bool IsReliabilityMechanism(const webrtc::RtpCodecCapability& codec) {
std::string GetCurrentCodecMimeType(
rtc::scoped_refptr<const webrtc::RTCStatsReport> report,
const webrtc::RTCOutboundRTPStreamStats& outbound_rtp) {
const webrtc::RTCOutboundRtpStreamStats& outbound_rtp) {
return outbound_rtp.codec_id.is_defined()
? *report->GetAs<webrtc::RTCCodecStats>(*outbound_rtp.codec_id)
->mime_type
@ -151,8 +151,8 @@ struct RidAndResolution {
uint32_t height;
};
const webrtc::RTCOutboundRTPStreamStats* FindOutboundRtpByRid(
const std::vector<const webrtc::RTCOutboundRTPStreamStats*>& outbound_rtps,
const webrtc::RTCOutboundRtpStreamStats* FindOutboundRtpByRid(
const std::vector<const webrtc::RTCOutboundRtpStreamStats*>& outbound_rtps,
const absl::string_view& rid) {
for (const auto* outbound_rtp : outbound_rtps) {
if (outbound_rtp->rid.is_defined() && *outbound_rtp->rid == rid) {
@ -964,8 +964,8 @@ class PeerConnectionSimulcastWithMediaFlowTests
rtc::scoped_refptr<PeerConnectionTestWrapper> pc_wrapper,
size_t num_layers) {
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(pc_wrapper);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRTPStreamStats>();
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
if (outbound_rtps.size() != num_layers) {
return false;
}
@ -983,10 +983,10 @@ class PeerConnectionSimulcastWithMediaFlowTests
std::vector<RidAndResolution> resolutions,
bool log_during_ramp_up) {
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(pc_wrapper);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRTPStreamStats>();
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
for (const RidAndResolution& resolution : resolutions) {
const RTCOutboundRTPStreamStats* outbound_rtp = nullptr;
const RTCOutboundRtpStreamStats* outbound_rtp = nullptr;
if (!resolution.rid.empty()) {
outbound_rtp = FindOutboundRtpByRid(outbound_rtps, resolution.rid);
} else if (outbound_rtps.size() == 1u) {
@ -1102,8 +1102,8 @@ TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
kLongTimeoutForRampingUp.ms());
// Verify codec and scalability mode.
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRTPStreamStats>();
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_THAT(outbound_rtps, SizeIs(1u));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]),
StrCaseEq("video/VP8"));
@ -1142,8 +1142,8 @@ TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
kDefaultTimeout.ms());
// Verify codec and scalability mode.
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRTPStreamStats>();
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_THAT(outbound_rtps, SizeIs(3u));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]),
StrCaseEq("video/VP8"));
@ -1191,8 +1191,8 @@ TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
kDefaultTimeout.ms());
// When `scalability_mode` is not set, VP8 defaults to L1T1.
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRTPStreamStats>();
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_THAT(outbound_rtps, SizeIs(1u));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]),
StrCaseEq("video/VP8"));
@ -1251,8 +1251,8 @@ TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
// GetStats() confirms "L1T2" is used which is different than the "L1T1"
// default or the "L3T3_KEY" that was attempted.
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRTPStreamStats>();
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_THAT(outbound_rtps, SizeIs(1u));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]),
StrCaseEq("video/VP8"));
@ -1293,8 +1293,8 @@ TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
kDefaultTimeout.ms());
// Verify codec and scalability mode.
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRTPStreamStats>();
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_THAT(outbound_rtps, SizeIs(3u));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]),
StrCaseEq("video/H264"));
@ -1345,8 +1345,8 @@ TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
kLongTimeoutForRampingUp.ms());
// Verify codec and scalability mode.
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRTPStreamStats>();
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_THAT(outbound_rtps, SizeIs(1u));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]),
StrCaseEq("video/VP9"));
@ -1404,8 +1404,8 @@ TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
kLongTimeoutForRampingUp.ms());
// Verify codec and scalability mode.
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRTPStreamStats>();
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_THAT(outbound_rtps, SizeIs(1u));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]),
StrCaseEq("video/VP9"));
@ -1457,8 +1457,8 @@ TEST_F(PeerConnectionSimulcastWithMediaFlowTests,
EXPECT_TRUE_WAIT(HasOutboundRtpBytesSent(local_pc_wrapper, 1u),
kLongTimeoutForRampingUp.ms());
rtc::scoped_refptr<const RTCStatsReport> report = GetStats(local_pc_wrapper);
std::vector<const RTCOutboundRTPStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRTPStreamStats>();
std::vector<const RTCOutboundRtpStreamStats*> outbound_rtps =
report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_THAT(outbound_rtps, SizeIs(1u));
EXPECT_THAT(GetCurrentCodecMimeType(report, *outbound_rtps[0]),
StrCaseEq("video/VP9"));

48
pc/rtc_stats_collector.cc
View File

@ -124,7 +124,7 @@ std::string RTCTransportStatsIDFromTransportChannel(
return sb.str();
}
std::string RTCInboundRTPStreamStatsIDFromSSRC(const std::string& transport_id,
std::string RTCInboundRtpStreamStatsIDFromSSRC(const std::string& transport_id,
cricket::MediaType media_type,
uint32_t ssrc) {
char buf[1024];
@ -134,7 +134,7 @@ std::string RTCInboundRTPStreamStatsIDFromSSRC(const std::string& transport_id,
return sb.str();
}
std::string RTCOutboundRTPStreamStatsIDFromSSRC(const std::string& transport_id,
std::string RTCOutboundRtpStreamStatsIDFromSSRC(const std::string& transport_id,
cricket::MediaType media_type,
uint32_t ssrc) {
char buf[1024];
@ -412,7 +412,7 @@ void SetMediaStreamTrackStatsFromMediaStreamTrackInterface(
// Provides the media independent counters (both audio and video).
void SetInboundRTPStreamStatsFromMediaReceiverInfo(
const cricket::MediaReceiverInfo& media_receiver_info,
RTCInboundRTPStreamStats* inbound_stats) {
RTCInboundRtpStreamStats* inbound_stats) {
RTC_DCHECK(inbound_stats);
inbound_stats->ssrc = media_receiver_info.ssrc();
inbound_stats->packets_received =
@ -440,15 +440,15 @@ void SetInboundRTPStreamStatsFromMediaReceiverInfo(
}
}
std::unique_ptr<RTCInboundRTPStreamStats> CreateInboundAudioStreamStats(
std::unique_ptr<RTCInboundRtpStreamStats> CreateInboundAudioStreamStats(
const cricket::VoiceMediaInfo& voice_media_info,
const cricket::VoiceReceiverInfo& voice_receiver_info,
const std::string& transport_id,
const std::string& mid,
Timestamp timestamp,
RTCStatsReport* report) {
auto inbound_audio = std::make_unique<RTCInboundRTPStreamStats>(
/*id=*/RTCInboundRTPStreamStatsIDFromSSRC(
auto inbound_audio = std::make_unique<RTCInboundRtpStreamStats>(
/*id=*/RTCInboundRtpStreamStatsIDFromSSRC(
transport_id, cricket::MEDIA_TYPE_AUDIO, voice_receiver_info.ssrc()),
timestamp);
SetInboundRTPStreamStatsFromMediaReceiverInfo(voice_receiver_info,
@ -537,7 +537,7 @@ std::unique_ptr<RTCRemoteOutboundRtpStreamStats>
CreateRemoteOutboundAudioStreamStats(
const cricket::VoiceReceiverInfo& voice_receiver_info,
const std::string& mid,
const RTCInboundRTPStreamStats& inbound_audio_stats,
const RTCInboundRtpStreamStats& inbound_audio_stats,
const std::string& transport_id) {
if (!voice_receiver_info.last_sender_report_timestamp_ms.has_value()) {
// Cannot create `RTCRemoteOutboundRtpStreamStats` when the RTCP SR arrival
@ -584,7 +584,7 @@ CreateRemoteOutboundAudioStreamStats(
return stats;
}
std::unique_ptr<RTCInboundRTPStreamStats>
std::unique_ptr<RTCInboundRtpStreamStats>
CreateInboundRTPStreamStatsFromVideoReceiverInfo(
const std::string& transport_id,
const std::string& mid,
@ -592,8 +592,8 @@ CreateInboundRTPStreamStatsFromVideoReceiverInfo(
const cricket::VideoReceiverInfo& video_receiver_info,
Timestamp timestamp,
RTCStatsReport* report) {
auto inbound_video = std::make_unique<RTCInboundRTPStreamStats>(
RTCInboundRTPStreamStatsIDFromSSRC(
auto inbound_video = std::make_unique<RTCInboundRtpStreamStats>(
RTCInboundRtpStreamStatsIDFromSSRC(
transport_id, cricket::MEDIA_TYPE_VIDEO, video_receiver_info.ssrc()),
timestamp);
SetInboundRTPStreamStatsFromMediaReceiverInfo(video_receiver_info,
@ -691,7 +691,7 @@ CreateInboundRTPStreamStatsFromVideoReceiverInfo(
// video).
void SetOutboundRTPStreamStatsFromMediaSenderInfo(
const cricket::MediaSenderInfo& media_sender_info,
RTCOutboundRTPStreamStats* outbound_stats) {
RTCOutboundRtpStreamStats* outbound_stats) {
RTC_DCHECK(outbound_stats);
outbound_stats->ssrc = media_sender_info.ssrc();
outbound_stats->packets_sent =
@ -712,7 +712,7 @@ void SetOutboundRTPStreamStatsFromMediaSenderInfo(
}
}
std::unique_ptr<RTCOutboundRTPStreamStats>
std::unique_ptr<RTCOutboundRtpStreamStats>
CreateOutboundRTPStreamStatsFromVoiceSenderInfo(
const std::string& transport_id,
const std::string& mid,
@ -720,8 +720,8 @@ CreateOutboundRTPStreamStatsFromVoiceSenderInfo(
const cricket::VoiceSenderInfo& voice_sender_info,
Timestamp timestamp,
RTCStatsReport* report) {
auto outbound_audio = std::make_unique<RTCOutboundRTPStreamStats>(
RTCOutboundRTPStreamStatsIDFromSSRC(
auto outbound_audio = std::make_unique<RTCOutboundRtpStreamStats>(
RTCOutboundRtpStreamStatsIDFromSSRC(
transport_id, cricket::MEDIA_TYPE_AUDIO, voice_sender_info.ssrc()),
timestamp);
SetOutboundRTPStreamStatsFromMediaSenderInfo(voice_sender_info,
@ -749,7 +749,7 @@ CreateOutboundRTPStreamStatsFromVoiceSenderInfo(
return outbound_audio;
}
std::unique_ptr<RTCOutboundRTPStreamStats>
std::unique_ptr<RTCOutboundRtpStreamStats>
CreateOutboundRTPStreamStatsFromVideoSenderInfo(
const std::string& transport_id,
const std::string& mid,
@ -757,8 +757,8 @@ CreateOutboundRTPStreamStatsFromVideoSenderInfo(
const cricket::VideoSenderInfo& video_sender_info,
Timestamp timestamp,
RTCStatsReport* report) {
auto outbound_video = std::make_unique<RTCOutboundRTPStreamStats>(
RTCOutboundRTPStreamStatsIDFromSSRC(
auto outbound_video = std::make_unique<RTCOutboundRtpStreamStats>(
RTCOutboundRtpStreamStatsIDFromSSRC(
transport_id, cricket::MEDIA_TYPE_VIDEO, video_sender_info.ssrc()),
timestamp);
SetOutboundRTPStreamStatsFromMediaSenderInfo(video_sender_info,
@ -842,7 +842,7 @@ ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
const std::string& transport_id,
const ReportBlockData& report_block_data,
cricket::MediaType media_type,
const std::map<std::string, RTCOutboundRTPStreamStats*>& outbound_rtps,
const std::map<std::string, RTCOutboundRtpStreamStats*>& outbound_rtps,
const RTCStatsReport& report) {
const auto& report_block = report_block_data.report_block();
// RTCStats' timestamp generally refers to when the metric was sampled, but
@ -869,7 +869,7 @@ ProduceRemoteInboundRtpStreamStatsFromReportBlockData(
remote_inbound->round_trip_time_measurements =
report_block_data.num_rtts();
std::string local_id = RTCOutboundRTPStreamStatsIDFromSSRC(
std::string local_id = RTCOutboundRtpStreamStatsIDFromSSRC(
transport_id, media_type, report_block.source_ssrc);
// Look up local stat from `outbound_rtps` where the pointers are non-const.
auto local_id_it = outbound_rtps.find(local_id);
@ -1170,7 +1170,7 @@ ProduceMediaStreamTrackStatsFromVideoReceiverInfo(
// TODO(hbos): When we support receiving simulcast, this should be the total
// number of frames correctly decoded, independent of which SSRC it was
// received from. Since we don't support that, this is correct and is the same
// value as "RTCInboundRTPStreamStats.framesDecoded". https://crbug.com/659137
// value as "RTCInboundRtpStreamStats.framesDecoded". https://crbug.com/659137
video_track_stats->frames_decoded = video_receiver_info.frames_decoded;
video_track_stats->frames_dropped = video_receiver_info.frames_dropped;
@ -1292,7 +1292,7 @@ RTCStatsCollector::CreateReportFilteredBySelector(
// Find outbound-rtp(s) of the sender using ssrc lookup.
auto encodings = sender_selector->GetParametersInternal().encodings;
for (const auto* outbound_rtp :
report->GetStatsOfType<RTCOutboundRTPStreamStats>()) {
report->GetStatsOfType<RTCOutboundRtpStreamStats>()) {
RTC_DCHECK(outbound_rtp->ssrc.is_defined());
auto it = std::find_if(
encodings.begin(), encodings.end(),
@ -1313,7 +1313,7 @@ RTCStatsCollector::CreateReportFilteredBySelector(
worker_thread_->BlockingCall([&] { ssrc = receiver_selector->ssrc(); });
if (ssrc.has_value()) {
for (const auto* inbound_rtp :
report->GetStatsOfType<RTCInboundRTPStreamStats>()) {
report->GetStatsOfType<RTCInboundRtpStreamStats>()) {
RTC_DCHECK(inbound_rtp->ssrc.is_defined());
if (*inbound_rtp->ssrc == *ssrc) {
rtpstream_ids.push_back(inbound_rtp->id());
@ -2049,7 +2049,7 @@ void RTCStatsCollector::ProduceAudioRTPStreamStats_n(
}
}
// Outbound.
std::map<std::string, RTCOutboundRTPStreamStats*> audio_outbound_rtps;
std::map<std::string, RTCOutboundRtpStreamStats*> audio_outbound_rtps;
for (const cricket::VoiceSenderInfo& voice_sender_info :
stats.track_media_info_map.voice_media_info()->senders) {
if (!voice_sender_info.connected())
@ -2134,7 +2134,7 @@ void RTCStatsCollector::ProduceVideoRTPStreamStats_n(
}
}
// Outbound
std::map<std::string, RTCOutboundRTPStreamStats*> video_outbound_rtps;
std::map<std::string, RTCOutboundRtpStreamStats*> video_outbound_rtps;
for (const cricket::VideoSenderInfo& video_sender_info :
stats.track_media_info_map.video_media_info()->senders) {
if (!video_sender_info.connected())

2
pc/rtc_stats_collector.h
View File

@ -211,7 +211,7 @@ class RTCStatsCollector : public rtc::RefCountInterface {
// Produces `RTCAudioPlayoutStats`.
void ProduceAudioPlayoutStats_s(Timestamp timestamp,
RTCStatsReport* report) const;
// Produces `RTCInboundRTPStreamStats`, `RTCOutboundRTPStreamStats`,
// Produces `RTCInboundRtpStreamStats`, `RTCOutboundRtpStreamStats`,
// `RTCRemoteInboundRtpStreamStats`, `RTCRemoteOutboundRtpStreamStats` and any
// referenced `RTCCodecStats`. This has to be invoked after transport stats
// have been created because some metrics are calculated through lookup of

70
pc/rtc_stats_collector_unittest.cc
View File

@ -122,11 +122,11 @@ void PrintTo(const DEPRECATED_RTCMediaStreamTrackStats& stats,
*os << stats.ToJson();
}
void PrintTo(const RTCInboundRTPStreamStats& stats, ::std::ostream* os) {
void PrintTo(const RTCInboundRtpStreamStats& stats, ::std::ostream* os) {
*os << stats.ToJson();
}
void PrintTo(const RTCOutboundRTPStreamStats& stats, ::std::ostream* os) {
void PrintTo(const RTCOutboundRtpStreamStats& stats, ::std::ostream* os) {
*os << stats.ToJson();
}
@ -809,7 +809,7 @@ class RTCStatsCollectorTest : public ::testing::Test {
->cast_to<DEPRECATED_RTCMediaStreamTrackStats>();
EXPECT_EQ(*sender_track.media_source_id, graph.media_source_id);
const auto& outbound_rtp = graph.full_report->Get(graph.outbound_rtp_id)
->cast_to<RTCOutboundRTPStreamStats>();
->cast_to<RTCOutboundRtpStreamStats>();
EXPECT_EQ(*outbound_rtp.media_source_id, graph.media_source_id);
EXPECT_EQ(*outbound_rtp.codec_id, graph.send_codec_id);
EXPECT_EQ(*outbound_rtp.track_id, graph.sender_track_id);
@ -820,7 +820,7 @@ class RTCStatsCollectorTest : public ::testing::Test {
return graph;
}
const auto& inbound_rtp = graph.full_report->Get(graph.inbound_rtp_id)
->cast_to<RTCInboundRTPStreamStats>();
->cast_to<RTCInboundRtpStreamStats>();
EXPECT_EQ(*inbound_rtp.codec_id, graph.recv_codec_id);
EXPECT_EQ(*inbound_rtp.track_id, graph.receiver_track_id);
EXPECT_EQ(*inbound_rtp.transport_id, graph.transport_id);
@ -931,7 +931,7 @@ class RTCStatsCollectorTest : public ::testing::Test {
->cast_to<DEPRECATED_RTCMediaStreamTrackStats>();
EXPECT_EQ(*sender_track.media_source_id, graph.media_source_id);
const auto& outbound_rtp = graph.full_report->Get(graph.outbound_rtp_id)
->cast_to<RTCOutboundRTPStreamStats>();
->cast_to<RTCOutboundRtpStreamStats>();
EXPECT_EQ(*outbound_rtp.media_source_id, graph.media_source_id);
EXPECT_EQ(*outbound_rtp.codec_id, graph.send_codec_id);
EXPECT_EQ(*outbound_rtp.track_id, graph.sender_track_id);
@ -942,7 +942,7 @@ class RTCStatsCollectorTest : public ::testing::Test {
return graph;
}
const auto& inbound_rtp = graph.full_report->Get(graph.inbound_rtp_id)
->cast_to<RTCInboundRTPStreamStats>();
->cast_to<RTCInboundRtpStreamStats>();
EXPECT_EQ(*inbound_rtp.codec_id, graph.recv_codec_id);
EXPECT_EQ(*inbound_rtp.track_id, graph.receiver_track_id);
EXPECT_EQ(*inbound_rtp.transport_id, graph.transport_id);
@ -1079,8 +1079,8 @@ TEST_F(RTCStatsCollectorTest, ValidSsrcCollisionDoesNotCrash) {
// This should not crash (https://crbug.com/1361612).
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
auto inbound_rtps = report->GetStatsOfType<RTCInboundRTPStreamStats>();
auto outbound_rtps = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
auto outbound_rtps = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
EXPECT_EQ(inbound_rtps.size(), 4u);
EXPECT_EQ(outbound_rtps.size(), 4u);
}
@ -2460,7 +2460,7 @@ TEST_F(RTCStatsCollectorTest,
->cast_to<DEPRECATED_RTCMediaStreamTrackStats>());
}
TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio) {
cricket::VoiceMediaInfo voice_media_info;
voice_media_info.receivers.push_back(cricket::VoiceReceiverInfo());
@ -2523,7 +2523,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
report->GetStatsOfType<DEPRECATED_RTCMediaStreamTrackStats>();
ASSERT_EQ(1U, stats_of_track_type.size());
RTCInboundRTPStreamStats expected_audio("ITTransportName1A1",
RTCInboundRtpStreamStats expected_audio("ITTransportName1A1",
report->timestamp());
expected_audio.ssrc = 1;
expected_audio.media_type = "audio";
@ -2565,7 +2565,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(
report->Get(expected_audio.id())->cast_to<RTCInboundRTPStreamStats>(),
report->Get(expected_audio.id())->cast_to<RTCInboundRtpStreamStats>(),
expected_audio);
// Set previously undefined values and "GetStats" again.
@ -2580,14 +2580,14 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio) {
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(
report->Get(expected_audio.id())->cast_to<RTCInboundRTPStreamStats>(),
report->Get(expected_audio.id())->cast_to<RTCInboundRtpStreamStats>(),
expected_audio);
EXPECT_TRUE(report->Get(*expected_audio.track_id));
EXPECT_TRUE(report->Get(*expected_audio.transport_id));
EXPECT_TRUE(report->Get(*expected_audio.codec_id));
}
TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio_PlayoutId) {
TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Audio_PlayoutId) {
cricket::VoiceMediaInfo voice_media_info;
voice_media_info.receivers.push_back(cricket::VoiceReceiverInfo());
@ -2608,7 +2608,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio_PlayoutId) {
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
ASSERT_TRUE(report->Get("ITTransportName1A1"));
auto stats =
report->Get("ITTransportName1A1")->cast_to<RTCInboundRTPStreamStats>();
report->Get("ITTransportName1A1")->cast_to<RTCInboundRtpStreamStats>();
ASSERT_FALSE(stats.playout_id.is_defined());
}
{
@ -2619,13 +2619,13 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Audio_PlayoutId) {
stats_->GetFreshStatsReport();
ASSERT_TRUE(report->Get("ITTransportName1A1"));
auto stats =
report->Get("ITTransportName1A1")->cast_to<RTCInboundRTPStreamStats>();
report->Get("ITTransportName1A1")->cast_to<RTCInboundRtpStreamStats>();
ASSERT_TRUE(stats.playout_id.is_defined());
EXPECT_EQ(*stats.playout_id, "AP");
}
}
TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Video) {
cricket::VideoMediaInfo video_media_info;
video_media_info.receivers.push_back(cricket::VideoReceiverInfo());
@ -2693,7 +2693,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
RTCInboundRTPStreamStats expected_video("ITTransportName1V1",
RTCInboundRtpStreamStats expected_video("ITTransportName1V1",
report->timestamp());
expected_video.ssrc = 1;
expected_video.media_type = "video";
@ -2740,7 +2740,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
ASSERT_TRUE(report->Get(expected_video.id()));
EXPECT_EQ(
report->Get(expected_video.id())->cast_to<RTCInboundRTPStreamStats>(),
report->Get(expected_video.id())->cast_to<RTCInboundRtpStreamStats>(),
expected_video);
// Set previously undefined values and "GetStats" again.
@ -2763,7 +2763,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRTPStreamStats_Video) {
ASSERT_TRUE(report->Get(expected_video.id()));
EXPECT_EQ(
report->Get(expected_video.id())->cast_to<RTCInboundRTPStreamStats>(),
report->Get(expected_video.id())->cast_to<RTCInboundRtpStreamStats>(),
expected_video);
EXPECT_TRUE(report->Get(*expected_video.track_id));
EXPECT_TRUE(report->Get(*expected_video.transport_id));
@ -2825,14 +2825,14 @@ TEST_F(RTCStatsCollectorTest, CollectGoogTimingFrameInfo) {
cricket::MEDIA_TYPE_VIDEO, "RemoteVideoTrackID", "RemoteStreamId", 1);
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
auto inbound_rtps = report->GetStatsOfType<RTCInboundRTPStreamStats>();
auto inbound_rtps = report->GetStatsOfType<RTCInboundRtpStreamStats>();
ASSERT_EQ(inbound_rtps.size(), 1u);
ASSERT_TRUE(inbound_rtps[0]->goog_timing_frame_info.is_defined());
EXPECT_EQ(*inbound_rtps[0]->goog_timing_frame_info,
"1,2,3,4,5,6,7,8,9,10,11,12,13,1,0");
}
TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Audio) {
cricket::VoiceMediaInfo voice_media_info;
voice_media_info.senders.push_back(cricket::VoiceSenderInfo());
@ -2864,7 +2864,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
RTCOutboundRTPStreamStats expected_audio("OTTransportName1A1",
RTCOutboundRtpStreamStats expected_audio("OTTransportName1A1",
report->timestamp());
expected_audio.media_source_id = "SA50";
// `expected_audio.remote_id` should be undefined.
@ -2888,19 +2888,19 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Audio) {
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(
report->Get(expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>(),
report->Get(expected_audio.id())->cast_to<RTCOutboundRtpStreamStats>(),
expected_audio);
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(
report->Get(expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>(),
report->Get(expected_audio.id())->cast_to<RTCOutboundRtpStreamStats>(),
expected_audio);
EXPECT_TRUE(report->Get(*expected_audio.track_id));
EXPECT_TRUE(report->Get(*expected_audio.transport_id));
EXPECT_TRUE(report->Get(*expected_audio.codec_id));
}
TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
cricket::VideoMediaInfo video_media_info;
video_media_info.senders.push_back(cricket::VideoSenderInfo());
@ -2954,13 +2954,13 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
auto stats_of_my_type = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
auto stats_of_my_type = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
ASSERT_EQ(1U, stats_of_my_type.size());
auto stats_of_track_type =
report->GetStatsOfType<DEPRECATED_RTCMediaStreamTrackStats>();
ASSERT_EQ(1U, stats_of_track_type.size());
RTCOutboundRTPStreamStats expected_video(stats_of_my_type[0]->id(),
RTCOutboundRtpStreamStats expected_video(stats_of_my_type[0]->id(),
report->timestamp());
expected_video.media_source_id = "SV50";
// `expected_video.remote_id` should be undefined.
@ -3003,7 +3003,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
ASSERT_TRUE(report->Get(expected_video.id()));
EXPECT_EQ(
report->Get(expected_video.id())->cast_to<RTCOutboundRTPStreamStats>(),
report->Get(expected_video.id())->cast_to<RTCOutboundRtpStreamStats>(),
expected_video);
// Set previously undefined values and "GetStats" again.
@ -3023,7 +3023,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRTPStreamStats_Video) {
ASSERT_TRUE(report->Get(expected_video.id()));
EXPECT_EQ(
report->Get(expected_video.id())->cast_to<RTCOutboundRTPStreamStats>(),
report->Get(expected_video.id())->cast_to<RTCOutboundRtpStreamStats>(),
expected_video);
EXPECT_TRUE(report->Get(*expected_video.track_id));
EXPECT_TRUE(report->Get(*expected_video.transport_id));
@ -3275,7 +3275,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCTransportStatsWithCrypto) {
report->Get(expected_rtp_transport.id())->cast_to<RTCTransportStats>());
}
TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRtpStreamStats_Audio) {
cricket::VoiceMediaInfo voice_media_info;
voice_media_info.senders.push_back(cricket::VoiceSenderInfo());
@ -3307,7 +3307,7 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
rtc::scoped_refptr<const RTCStatsReport> report = stats_->GetStatsReport();
RTCOutboundRTPStreamStats expected_audio("OTTransportName1A1",
RTCOutboundRtpStreamStats expected_audio("OTTransportName1A1",
report->timestamp());
expected_audio.media_source_id = "SA50";
expected_audio.mid = "AudioMid";
@ -3329,7 +3329,7 @@ TEST_F(RTCStatsCollectorTest, CollectNoStreamRTCOutboundRTPStreamStats_Audio) {
ASSERT_TRUE(report->Get(expected_audio.id()));
EXPECT_EQ(
report->Get(expected_audio.id())->cast_to<RTCOutboundRTPStreamStats>(),
report->Get(expected_audio.id())->cast_to<RTCOutboundRtpStreamStats>(),
expected_audio);
EXPECT_TRUE(report->Get(*expected_audio.track_id));
EXPECT_TRUE(report->Get(*expected_audio.transport_id));
@ -3656,7 +3656,7 @@ TEST_P(RTCStatsCollectorTestWithParamKind,
ASSERT_TRUE(report->Get(*expected_remote_inbound_rtp.local_id));
// Lookup works in both directions.
EXPECT_EQ(*report->Get(*expected_remote_inbound_rtp.local_id)
->cast_to<RTCOutboundRTPStreamStats>()
->cast_to<RTCOutboundRtpStreamStats>()
.remote_id,
expected_remote_inbound_rtp.id());
}
@ -4006,7 +4006,7 @@ TEST_F(RTCStatsCollectorTest, RtpIsMissingWhileSsrcIsZero) {
auto tracks = report->GetStatsOfType<DEPRECATED_RTCMediaStreamTrackStats>();
EXPECT_EQ(1U, tracks.size());
auto outbound_rtps = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
auto outbound_rtps = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
EXPECT_TRUE(outbound_rtps.empty());
}
@ -4027,7 +4027,7 @@ TEST_F(RTCStatsCollectorTest, DoNotCrashIfSsrcIsKnownButInfosAreStillMissing) {
std::vector<const DEPRECATED_RTCMediaStreamTrackStats*> track_stats =
report->GetStatsOfType<DEPRECATED_RTCMediaStreamTrackStats>();
EXPECT_EQ(1U, track_stats.size());
auto outbound_rtps = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
auto outbound_rtps = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
EXPECT_TRUE(outbound_rtps.empty());
}

28
pc/rtc_stats_integrationtest.cc
View File

@ -342,8 +342,8 @@ class RTCStatsReportVerifier {
stats_types.insert(DEPRECATED_RTCMediaStreamStats::kType);
stats_types.insert(DEPRECATED_RTCMediaStreamTrackStats::kType);
stats_types.insert(RTCPeerConnectionStats::kType);
stats_types.insert(RTCInboundRTPStreamStats::kType);
stats_types.insert(RTCOutboundRTPStreamStats::kType);
stats_types.insert(RTCInboundRtpStreamStats::kType);
stats_types.insert(RTCOutboundRtpStreamStats::kType);
stats_types.insert(RTCTransportStats::kType);
return stats_types;
}
@ -389,12 +389,12 @@ class RTCStatsReportVerifier {
} else if (stats.type() == RTCPeerConnectionStats::kType) {
verify_successful &= VerifyRTCPeerConnectionStats(
stats.cast_to<RTCPeerConnectionStats>());
} else if (stats.type() == RTCInboundRTPStreamStats::kType) {
} else if (stats.type() == RTCInboundRtpStreamStats::kType) {
verify_successful &= VerifyRTCInboundRtpStreamStats(
stats.cast_to<RTCInboundRTPStreamStats>());
} else if (stats.type() == RTCOutboundRTPStreamStats::kType) {
stats.cast_to<RTCInboundRtpStreamStats>());
} else if (stats.type() == RTCOutboundRtpStreamStats::kType) {
verify_successful &= VerifyRTCOutboundRtpStreamStats(
stats.cast_to<RTCOutboundRTPStreamStats>());
stats.cast_to<RTCOutboundRtpStreamStats>());
} else if (stats.type() == RTCRemoteInboundRtpStreamStats::kType) {
verify_successful &= VerifyRTCRemoteInboundRtpStreamStats(
stats.cast_to<RTCRemoteInboundRtpStreamStats>());
@ -721,8 +721,8 @@ class RTCStatsReportVerifier {
verifier.TestMemberIsDefined(stream.kind);
// Some legacy metrics are only defined for some of the RTP types in the
// hierarcy.
if (stream.type() == RTCInboundRTPStreamStats::kType ||
stream.type() == RTCOutboundRTPStreamStats::kType) {
if (stream.type() == RTCInboundRtpStreamStats::kType ||
stream.type() == RTCOutboundRtpStreamStats::kType) {
verifier.TestMemberIsDefined(stream.media_type);
verifier.TestMemberIsIDReference(
stream.track_id, DEPRECATED_RTCMediaStreamTrackStats::kType);
@ -743,7 +743,7 @@ class RTCStatsReportVerifier {
}
bool VerifyRTCInboundRtpStreamStats(
const RTCInboundRTPStreamStats& inbound_stream) {
const RTCInboundRtpStreamStats& inbound_stream) {
RTCStatsVerifier verifier(report_.get(), &inbound_stream);
VerifyRTCReceivedRtpStreamStats(inbound_stream, verifier);
verifier.TestMemberIsOptionalIDReference(
@ -921,7 +921,7 @@ class RTCStatsReportVerifier {
}
bool VerifyRTCOutboundRtpStreamStats(
const RTCOutboundRTPStreamStats& outbound_stream) {
const RTCOutboundRtpStreamStats& outbound_stream) {
RTCStatsVerifier verifier(report_.get(), &outbound_stream);
VerifyRTCRtpStreamStats(outbound_stream, verifier);
verifier.TestMemberIsDefined(outbound_stream.mid);
@ -1034,7 +1034,7 @@ class RTCStatsReportVerifier {
VerifyRTCReceivedRtpStreamStats(remote_inbound_stream, verifier);
verifier.TestMemberIsDefined(remote_inbound_stream.fraction_lost);
verifier.TestMemberIsIDReference(remote_inbound_stream.local_id,
RTCOutboundRTPStreamStats::kType);
RTCOutboundRtpStreamStats::kType);
verifier.TestMemberIsNonNegative<double>(
remote_inbound_stream.round_trip_time);
verifier.TestMemberIsNonNegative<double>(
@ -1050,7 +1050,7 @@ class RTCStatsReportVerifier {
VerifyRTCRtpStreamStats(remote_outbound_stream, verifier);
VerifyRTCSentRtpStreamStats(remote_outbound_stream, verifier);
verifier.TestMemberIsIDReference(remote_outbound_stream.local_id,
RTCOutboundRTPStreamStats::kType);
RTCOutboundRtpStreamStats::kType);
verifier.TestMemberIsNonNegative<double>(
remote_outbound_stream.remote_timestamp);
verifier.TestMemberIsDefined(remote_outbound_stream.reports_sent);
@ -1196,7 +1196,7 @@ TEST_F(RTCStatsIntegrationTest, GetStatsWithSenderSelector) {
// TODO(hbos): Include RTC[Audio/Video]ReceiverStats when implemented.
// TODO(hbos): Include RTCRemoteOutboundRtpStreamStats when implemented.
// TODO(hbos): Include RTCRtpContributingSourceStats when implemented.
RTCInboundRTPStreamStats::kType,
RTCInboundRtpStreamStats::kType,
RTCPeerConnectionStats::kType,
DEPRECATED_RTCMediaStreamStats::kType,
RTCDataChannelStats::kType,
@ -1215,7 +1215,7 @@ TEST_F(RTCStatsIntegrationTest, GetStatsWithReceiverSelector) {
// TODO(hbos): Include RTC[Audio/Video]SenderStats when implemented.
// TODO(hbos): Include RTCRemoteInboundRtpStreamStats when implemented.
// TODO(hbos): Include RTCRtpContributingSourceStats when implemented.
RTCOutboundRTPStreamStats::kType,
RTCOutboundRtpStreamStats::kType,
RTCPeerConnectionStats::kType,
DEPRECATED_RTCMediaStreamStats::kType,
RTCDataChannelStats::kType,

8
pc/rtc_stats_traversal.cc
View File

@ -101,17 +101,17 @@ std::vector<const std::string*> GetStatsReferencedIds(const RTCStats& stats) {
AddIdIfDefined(track.media_source_id, &neighbor_ids);
} else if (type == RTCPeerConnectionStats::kType) {
// RTCPeerConnectionStats does not have any neighbor references.
} else if (type == RTCInboundRTPStreamStats::kType) {
} else if (type == RTCInboundRtpStreamStats::kType) {
const auto& inbound_rtp =
static_cast<const RTCInboundRTPStreamStats&>(stats);
static_cast<const RTCInboundRtpStreamStats&>(stats);
AddIdIfDefined(inbound_rtp.remote_id, &neighbor_ids);
AddIdIfDefined(inbound_rtp.track_id, &neighbor_ids);
AddIdIfDefined(inbound_rtp.transport_id, &neighbor_ids);
AddIdIfDefined(inbound_rtp.codec_id, &neighbor_ids);
AddIdIfDefined(inbound_rtp.playout_id, &neighbor_ids);
} else if (type == RTCOutboundRTPStreamStats::kType) {
} else if (type == RTCOutboundRtpStreamStats::kType) {
const auto& outbound_rtp =
static_cast<const RTCOutboundRTPStreamStats&>(stats);
static_cast<const RTCOutboundRtpStreamStats&>(stats);
AddIdIfDefined(outbound_rtp.remote_id, &neighbor_ids);
AddIdIfDefined(outbound_rtp.track_id, &neighbor_ids);
AddIdIfDefined(outbound_rtp.transport_id, &neighbor_ids);

2
pc/test/integration_test_helpers.cc
View File

@ -46,7 +46,7 @@ void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc) {
int FindFirstMediaStatsIndexByKind(
const std::string& kind,
const std::vector<const webrtc::RTCInboundRTPStreamStats*>& inbound_rtps) {
const std::vector<const webrtc::RTCInboundRtpStreamStats*>& inbound_rtps) {
for (size_t i = 0; i < inbound_rtps.size(); i++) {
if (*inbound_rtps[i]->kind == kind) {
return i;

6
pc/test/integration_test_helpers.h
View File

@ -173,7 +173,7 @@ void RemoveSsrcsAndKeepMsids(cricket::SessionDescription* desc);
int FindFirstMediaStatsIndexByKind(
const std::string& kind,
const std::vector<const webrtc::RTCInboundRTPStreamStats*>& inbound_rtps);
const std::vector<const webrtc::RTCInboundRtpStreamStats*>& inbound_rtps);
class TaskQueueMetronome : public webrtc::Metronome {
public:
@ -648,7 +648,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
// Get the baseline numbers for audio_packets and audio_delay.
auto received_stats = NewGetStats();
auto rtp_stats =
received_stats->GetStatsOfType<webrtc::RTCInboundRTPStreamStats>()[0];
received_stats->GetStatsOfType<webrtc::RTCInboundRtpStreamStats>()[0];
ASSERT_TRUE(rtp_stats->relative_packet_arrival_delay.is_defined());
ASSERT_TRUE(rtp_stats->packets_received.is_defined());
ASSERT_TRUE(rtp_stats->track_id.is_defined());
@ -662,7 +662,7 @@ class PeerConnectionIntegrationWrapper : public webrtc::PeerConnectionObserver,
void UpdateDelayStats(std::string tag, int desc_size) {
auto report = NewGetStats();
auto rtp_stats =
report->GetAs<webrtc::RTCInboundRTPStreamStats>(rtp_stats_id_);
report->GetAs<webrtc::RTCInboundRtpStreamStats>(rtp_stats_id_);
ASSERT_TRUE(rtp_stats);
auto delta_packets = *rtp_stats->packets_received - audio_packets_stat_;
auto delta_rpad =

2
pc/test/svc_e2e_tests.cc
View File

@ -209,7 +209,7 @@ class SvcVideoQualityAnalyzer : public DefaultVideoQualityAnalyzer {
absl::string_view pc_label,
const rtc::scoped_refptr<const RTCStatsReport>& report) override {
// Extract the scalability mode reported in the stats.
auto outbound_stats = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
auto outbound_stats = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
for (const auto& stat : outbound_stats) {
if (stat->scalability_mode.is_defined()) {
reported_scalability_mode_ = *stat->scalability_mode;

20
stats/rtcstats_objects.cc
View File

@ -456,7 +456,7 @@ RTCSentRtpStreamStats::~RTCSentRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCInboundRTPStreamStats, RTCReceivedRtpStreamStats, "inbound-rtp",
RTCInboundRtpStreamStats, RTCReceivedRtpStreamStats, "inbound-rtp",
&track_identifier,
&mid,
&remote_id,
@ -515,7 +515,7 @@ WEBRTC_RTCSTATS_IMPL(
&min_playout_delay)
// clang-format on
RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string id,
RTCInboundRtpStreamStats::RTCInboundRtpStreamStats(std::string id,
Timestamp timestamp)
: RTCReceivedRtpStreamStats(std::move(id), timestamp),
playout_id("playoutId"),
@ -583,13 +583,13 @@ RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(std::string id,
total_interruption_duration("totalInterruptionDuration"),
min_playout_delay("minPlayoutDelay") {}
RTCInboundRTPStreamStats::RTCInboundRTPStreamStats(
const RTCInboundRTPStreamStats& other) = default;
RTCInboundRTPStreamStats::~RTCInboundRTPStreamStats() {}
RTCInboundRtpStreamStats::RTCInboundRtpStreamStats(
const RTCInboundRtpStreamStats& other) = default;
RTCInboundRtpStreamStats::~RTCInboundRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(
RTCOutboundRTPStreamStats, RTCSentRtpStreamStats, "outbound-rtp",
RTCOutboundRtpStreamStats, RTCSentRtpStreamStats, "outbound-rtp",
&media_source_id,
&remote_id,
&mid,
@ -622,7 +622,7 @@ WEBRTC_RTCSTATS_IMPL(
&scalability_mode)
// clang-format on
RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(std::string id,
RTCOutboundRtpStreamStats::RTCOutboundRtpStreamStats(std::string id,
Timestamp timestamp)
: RTCSentRtpStreamStats(std::move(id), timestamp),
media_source_id("mediaSourceId"),
@ -657,10 +657,10 @@ RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(std::string id,
power_efficient_encoder("powerEfficientEncoder"),
scalability_mode("scalabilityMode") {}
RTCOutboundRTPStreamStats::RTCOutboundRTPStreamStats(
const RTCOutboundRTPStreamStats& other) = default;
RTCOutboundRtpStreamStats::RTCOutboundRtpStreamStats(
const RTCOutboundRtpStreamStats& other) = default;
RTCOutboundRTPStreamStats::~RTCOutboundRTPStreamStats() {}
RTCOutboundRtpStreamStats::~RTCOutboundRtpStreamStats() {}
// clang-format off
WEBRTC_RTCSTATS_IMPL(

2
test/pc/e2e/analyzer/audio/default_audio_quality_analyzer.cc
View File

@ -39,7 +39,7 @@ void DefaultAudioQualityAnalyzer::Start(std::string test_case_name,
void DefaultAudioQualityAnalyzer::OnStatsReports(
absl::string_view pc_label,
const rtc::scoped_refptr<const RTCStatsReport>& report) {
auto stats = report->GetStatsOfType<RTCInboundRTPStreamStats>();
auto stats = report->GetStatsOfType<RTCInboundRtpStreamStats>();
for (auto& stat : stats) {
if (!stat->kind.is_defined() ||

2
test/pc/e2e/analyzer/video/video_quality_metrics_reporter.cc
View File

@ -68,7 +68,7 @@ void VideoQualityMetricsReporter::OnStatsReports(
const RTCIceCandidatePairStats ice_candidate_pair_stats =
report->Get(selected_ice_id)->cast_to<const RTCIceCandidatePairStats>();
auto outbound_rtp_stats = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
auto outbound_rtp_stats = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
StatsSample sample;
for (auto& s : outbound_rtp_stats) {
if (!s->kind.is_defined()) {

8
test/pc/e2e/cross_media_metrics_reporter.cc
View File

@ -43,8 +43,8 @@ void CrossMediaMetricsReporter::Start(
void CrossMediaMetricsReporter::OnStatsReports(
absl::string_view pc_label,
const rtc::scoped_refptr<const RTCStatsReport>& report) {
auto inbound_stats = report->GetStatsOfType<RTCInboundRTPStreamStats>();
std::map<std::string, std::vector<const RTCInboundRTPStreamStats*>>
auto inbound_stats = report->GetStatsOfType<RTCInboundRtpStreamStats>();
std::map<std::string, std::vector<const RTCInboundRtpStreamStats*>>
sync_group_stats;
for (const auto& stat : inbound_stats) {
if (stat->estimated_playout_timestamp.ValueOrDefault(0.) > 0 &&
@ -63,8 +63,8 @@ void CrossMediaMetricsReporter::OnStatsReports(
continue;
}
auto sync_group = std::string(pair.first);
const RTCInboundRTPStreamStats* audio_stat = pair.second[0];
const RTCInboundRTPStreamStats* video_stat = pair.second[1];
const RTCInboundRtpStreamStats* audio_stat = pair.second[0];
const RTCInboundRtpStreamStats* video_stat = pair.second[1];
RTC_CHECK(pair.second.size() == 2 && audio_stat->kind.is_defined() &&
video_stat->kind.is_defined() &&

4
test/pc/e2e/network_quality_metrics_reporter.cc
View File

@ -76,14 +76,14 @@ void NetworkQualityMetricsReporter::OnStatsReports(
DataSize payload_received = DataSize::Zero();
DataSize payload_sent = DataSize::Zero();
auto inbound_stats = report->GetStatsOfType<RTCInboundRTPStreamStats>();
auto inbound_stats = report->GetStatsOfType<RTCInboundRtpStreamStats>();
for (const auto& stat : inbound_stats) {
payload_received +=
DataSize::Bytes(stat->bytes_received.ValueOrDefault(0ul) +
stat->header_bytes_received.ValueOrDefault(0ul));
}
auto outbound_stats = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
auto outbound_stats = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
for (const auto& stat : outbound_stats) {
payload_sent +=
DataSize::Bytes(stat->bytes_sent.ValueOrDefault(0ul) +

4
test/pc/e2e/stats_based_network_quality_metrics_reporter.cc
View File

@ -296,14 +296,14 @@ void StatsBasedNetworkQualityMetricsReporter::OnStatsReports(
const rtc::scoped_refptr<const RTCStatsReport>& report) {
PCStats cur_stats;
auto inbound_stats = report->GetStatsOfType<RTCInboundRTPStreamStats>();
auto inbound_stats = report->GetStatsOfType<RTCInboundRtpStreamStats>();
for (const auto& stat : inbound_stats) {
cur_stats.payload_received +=
DataSize::Bytes(stat->bytes_received.ValueOrDefault(0ul) +
stat->header_bytes_received.ValueOrDefault(0ul));
}
auto outbound_stats = report->GetStatsOfType<RTCOutboundRTPStreamStats>();
auto outbound_stats = report->GetStatsOfType<RTCOutboundRtpStreamStats>();
for (const auto& stat : outbound_stats) {
cur_stats.payload_sent +=
DataSize::Bytes(stat->bytes_sent.ValueOrDefault(0ul) +