webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h - 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/source/bandwidth_estimator.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/source/bandwidth_estimator.h

diff --git a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h

index 291687608256c2c1738d69b6a2536f407d224326..0704337f7df1bc675fbab439a4b1b2c05b2fd4aa 100644

--- a/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h

+++ b/webrtc/modules/audio_coding/codecs/isac/main/source/bandwidth_estimator.h

@@ -95,7 +95,7 @@ extern "C" {

const int32_t frame_length,

const uint32_t send_ts,

const uint32_t arr_ts,

- const int32_t pksize);

+ const size_t pksize);

/* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */

int16_t WebRtcIsac_UpdateUplinkBwImpl(