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Unified Diff: webrtc/modules/audio_coding/codecs/isac/main/util/utility.h

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_coding/codecs/isac/main/util/utility.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h b/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h
index f9fba94315b6804257ff76d14007f3a751cece91..1bb6d295b4080ca873433f2ec625cdd367c6f368 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h
+++ b/webrtc/modules/audio_coding/codecs/isac/main/util/utility.h
@@ -99,7 +99,7 @@ extern "C" {
void get_arrival_time(
int current_framesamples, /* samples */
- int packet_size, /* bytes */
+ size_t packet_size, /* bytes */
int bottleneck, /* excluding headers; bits/s */
BottleNeckModel* BN_data,
short senderSampFreqHz,

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