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Unified Diff: webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c

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/source/bandwidth_estimator.c
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
index 940e8f50c76a740aada28079eeab9e42a04c09a8..51da3f7c76ce870fe0c9b80c7e01db6f08b2a450 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.c
@@ -142,7 +142,7 @@ int16_t WebRtcIsac_UpdateBandwidthEstimator(
const int32_t frame_length,
const uint32_t send_ts,
const uint32_t arr_ts,
- const int32_t pksize
+ const size_t pksize
/*, const uint16_t Index*/)
{
float weight = 0.0f;

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