Chromium Code Reviews| Index: webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc |
| diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc |
| index a75da49b2248a3f271ab34e8f238328b0d17a33d..aedf067278f6efc3e884ccc065edc8bdeff289a2 100644 |
| --- a/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc |
| +++ b/webrtc/modules/audio_coding/neteq/tools/neteq_delay_analyzer.cc |
| @@ -14,6 +14,8 @@ |
| #include <limits> |
| #include <utility> |
| +#include <webrtc/base/checks.h> |
| + |
| namespace webrtc { |
| namespace test { |
| namespace { |
| @@ -115,8 +117,9 @@ void NetEqDelayAnalyzer::CreateGraphs( |
| // This loop traverses data_ and populates rtp_timestamps_ms as well as |
| // calculates the base offset. |
| for (auto& d : data_) { |
| - rtp_timestamps_ms.push_back(unwrapper.Unwrap(d.first) / |
| - (last_sample_rate_hz_ / 1000.f)); |
|
ivoc
2017/06/12 10:19:19
Are you sure this change has any effect? I tried t
hlundin-webrtc
2017/06/12 11:54:58
The machine instructions change quite a bit: https
|
| + rtp_timestamps_ms.push_back( |
| + unwrapper.Unwrap(d.first) / |
| + rtc::CheckedDivExact(last_sample_rate_hz_, 1000)); |
| offset = |
| std::min(offset, d.second.arrival_time_ms - rtp_timestamps_ms.back()); |
| } |