webrtc/modules/audio_coding/codecs/isac/main/util/utility.c - 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_coding/codecs/isac/main/util/utility.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/util/utility.c

diff --git a/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c b/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c

index 0a2256a0365fd1cad9295a336b467f4098843188..d9c4332123cf31265de67e80bf2562f2d4908aaa 100644

--- a/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c

+++ b/webrtc/modules/audio_coding/codecs/isac/main/util/utility.c

@@ -135,7 +135,7 @@ readParamString(

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,