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Unified Diff: webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.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/fix/source/entropy_coding.c
diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
index 5f6e6ac0b1086ae3d7a260d5b1fbee8228513ec4..2379ba50661f2bac040ab2044ddc38a80be8b18a 100644
--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/entropy_coding.c
@@ -1870,7 +1870,7 @@ const uint16_t kFrameLenInitIndex[1] = {1};
int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
- int16_t *framesamples)
+ size_t *framesamples)
{
int err;

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