webrtc/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc - Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce

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

Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master

Patch Set: Resync Created 5 years, 4 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 ad491fb78a4298799674ab99ccf4ee41b4e08cd0..490fe58e9979d1e88dc031f7a77beb35cd55b527 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

@@ -287,7 +287,7 @@ 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.

@@ -357,7 +357,8 @@ int32_t AudioConferenceMixerImpl::SetOutputFrequency(

CriticalSectionScoped cs(_crit.get());

_outputFrequency = frequency;

- _sampleSize = (_outputFrequency*kProcessPeriodicityInMs) / 1000;

+ _sampleSize =

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

return 0;

}