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Unified Diff: webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc

Issue 1168753002: Match existing type usage better. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: Review comments + resync Created 5 years, 6 months ago
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Index: webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
index f6bfcdcd8368c594f7dbd27113062d1ed44bdf9b..47e2e676d7413e3235d871e2b40188329a05b5f4 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ b/webrtc/modules/audio_coding/codecs/isac/main/test/ReleaseTest-API/ReleaseTest-API.cc
@@ -53,7 +53,7 @@ int main(int argc, char* argv[]) {
int16_t stream_len = 0;
int declen = 0, declenTC = 0;
- int16_t lostFrame = 0;
+ bool lostFrame = false;
int16_t shortdata[SWBFRAMESAMPLES_10ms];
int16_t vaddata[SWBFRAMESAMPLES_10ms * 3];
@@ -697,7 +697,7 @@ int main(int argc, char* argv[]) {
if (!lostFrame) {
lostFrame = ((rand() % 100) < packetLossPercent);
} else {
- lostFrame = 0;
+ lostFrame = false;
}
// RED.

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