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Unified Diff: webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.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/decode_bwe.c
diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c b/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
index 5abe2041f9d639b58bc3008c381f68d81fa76696..019cc895288ba3f767e28d6037d5bfc247796d74 100644
--- a/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/decode_bwe.c
@@ -18,7 +18,7 @@ int
WebRtcIsac_EstimateBandwidth(
BwEstimatorstr* bwest_str,
Bitstr* streamdata,
- int32_t packet_size,
+ size_t packet_size,
uint16_t rtp_seq_number,
uint32_t send_ts,
uint32_t arr_ts,
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