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Diffstat (limited to 'third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc')
-rw-r--r--third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc25
1 files changed, 11 insertions, 14 deletions
diff --git a/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc b/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc
index 37821ac829..055be6fe99 100644
--- a/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc
+++ b/third_party/libwebrtc/pc/rtc_stats_collector_unittest.cc
@@ -263,9 +263,9 @@ class FakeAudioTrackForStats : public MediaStreamTrack<AudioTrackInterface> {
std::string kind() const override {
return MediaStreamTrackInterface::kAudioKind;
}
- webrtc::AudioSourceInterface* GetSource() const override { return nullptr; }
- void AddSink(webrtc::AudioTrackSinkInterface* sink) override {}
- void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override {}
+ AudioSourceInterface* GetSource() const override { return nullptr; }
+ void AddSink(AudioTrackSinkInterface* sink) override {}
+ void RemoveSink(AudioTrackSinkInterface* sink) override {}
bool GetSignalLevel(int* level) override { return false; }
rtc::scoped_refptr<AudioProcessorInterface> GetAudioProcessor() override {
return processor_;
@@ -2030,7 +2030,7 @@ TEST_F(RTCStatsCollectorTest, CollectRTCIceCandidatePairStats) {
EXPECT_TRUE(report->Get(*expected_pair.transport_id));
// Set bandwidth and "GetStats" again.
- webrtc::Call::Stats call_stats;
+ Call::Stats call_stats;
const int kSendBandwidth = 888;
call_stats.send_bandwidth_bps = kSendBandwidth;
const int kRecvBandwidth = 999;
@@ -2339,12 +2339,9 @@ TEST_F(RTCStatsCollectorTest, CollectRTCInboundRtpStreamStats_Video) {
video_media_info.receivers[0].key_frames_decoded = 3;
video_media_info.receivers[0].frames_dropped = 13;
video_media_info.receivers[0].qp_sum = absl::nullopt;
- video_media_info.receivers[0].total_decode_time =
- webrtc::TimeDelta::Seconds(9);
- video_media_info.receivers[0].total_processing_delay =
- webrtc::TimeDelta::Millis(600);
- video_media_info.receivers[0].total_assembly_time =
- webrtc::TimeDelta::Millis(500);
+ video_media_info.receivers[0].total_decode_time = TimeDelta::Seconds(9);
+ video_media_info.receivers[0].total_processing_delay = TimeDelta::Millis(600);
+ video_media_info.receivers[0].total_assembly_time = TimeDelta::Millis(500);
video_media_info.receivers[0].frames_assembled_from_multiple_packets = 23;
video_media_info.receivers[0].total_inter_frame_delay = 0.123;
video_media_info.receivers[0].total_squared_inter_frame_delay = 0.00456;
@@ -2617,12 +2614,12 @@ TEST_F(RTCStatsCollectorTest, CollectRTCOutboundRtpStreamStats_Video) {
video_media_info.senders[0].key_frames_encoded = 3;
video_media_info.senders[0].total_encode_time_ms = 9000;
video_media_info.senders[0].total_encoded_bytes_target = 1234;
- video_media_info.senders[0].total_packet_send_delay =
- webrtc::TimeDelta::Seconds(10);
+ video_media_info.senders[0].total_packet_send_delay = TimeDelta::Seconds(10);
video_media_info.senders[0].quality_limitation_reason =
QualityLimitationReason::kBandwidth;
- video_media_info.senders[0].quality_limitation_durations_ms
- [webrtc::QualityLimitationReason::kBandwidth] = 300;
+ video_media_info.senders[0]
+ .quality_limitation_durations_ms[QualityLimitationReason::kBandwidth] =
+ 300;
video_media_info.senders[0].quality_limitation_resolution_changes = 56u;
video_media_info.senders[0].qp_sum = absl::nullopt;
video_media_info.senders[0].content_type = VideoContentType::UNSPECIFIED;