webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc - Issue 1224123002: Update audio code to use size_t more correctly, webrtc/modules/ portion.

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Unified Diff: webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc

Issue 1224123002: Update audio code to use size_t more correctly, webrtc/modules/ portion. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master

Patch Set: Created 5 years, 5 months ago

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Index: webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc

diff --git a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc

index 3ee2a0863402a495e332ecd6a6eb5ddfcc19ec19..8267aee34d096bd6187d55b38817c1b77e441caf 100644

--- a/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc

+++ b/webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc

@@ -305,7 +305,7 @@ int32_t AudioConferenceMixerImpl::Process() {

AudioFrame::kNormalSpeech,

AudioFrame::kVadPassive, num_mixed_channels);

- _timeStamp += _sampleSize;

+ _timeStamp += static_cast<uint32_t>(_sampleSize);

// We only use the limiter if it supports the output sample rate and

// we're actually mixing multiple streams.

@@ -401,7 +401,8 @@ int32_t AudioConferenceMixerImpl::SetOutputFrequency(

CriticalSectionScoped cs(_crit.get());

_outputFrequency = frequency;

- _sampleSize = (_outputFrequency*kProcessPeriodicityInMs) / 1000;

+ _sampleSize =

+ static_cast<size_t>((_outputFrequency*kProcessPeriodicityInMs) / 1000);

return 0;

}