webrtc/modules/audio_device/dummy/file_audio_device.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_device/dummy/file_audio_device.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_device/dummy/file_audio_device.cc

diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.cc b/webrtc/modules/audio_device/dummy/file_audio_device.cc

index 3de5344a8f151a81540e57ca2bc7b25b5305c131..a2eac8767389fd702e576832cf0d99e598a27f1d 100644

--- a/webrtc/modules/audio_device/dummy/file_audio_device.cc

+++ b/webrtc/modules/audio_device/dummy/file_audio_device.cc

@@ -172,7 +172,7 @@ int32_t FileAudioDevice::InitRecording() {

return -1;

}

- _recordingFramesIn10MS = static_cast<uint32_t>(kRecordingFixedSampleRate/100);

+ _recordingFramesIn10MS = static_cast<size_t>(kRecordingFixedSampleRate / 100);

if (_ptrAudioBuffer) {

_ptrAudioBuffer->SetRecordingSampleRate(kRecordingFixedSampleRate);

@@ -190,7 +190,7 @@ int32_t FileAudioDevice::StartPlayout() {

return 0;

}

- _playoutFramesIn10MS = static_cast<uint32_t>(kPlayoutFixedSampleRate/100);

+ _playoutFramesIn10MS = static_cast<size_t>(kPlayoutFixedSampleRate / 100);

_playing = true;

_playoutFramesLeft = 0;