webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.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/fix/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/fix/source/decode_bwe.c

diff --git a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c

index b1f5d10a6551d49307f56d4d80e5483372039329..316f59a5e2fe49b9d5b768095bb16edadefe6591 100644

--- a/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c

+++ b/webrtc/modules/audio_coding/codecs/isac/fix/source/decode_bwe.c

@@ -26,13 +26,13 @@

int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,

Bitstr_dec *streamdata,

- int32_t packet_size,

+ size_t packet_size,

uint16_t rtp_seq_number,

uint32_t send_ts,

uint32_t arr_ts)

{

int16_t index;

- int16_t frame_samples;

+ size_t frame_samples;

int err;

/* decode framelength */

@@ -53,10 +53,10 @@ int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,

err = WebRtcIsacfix_UpdateUplinkBwImpl(

bwest_str,

rtp_seq_number,

- frame_samples * 1000 / FS,

+ (int16_t)(frame_samples * 1000 / FS),

send_ts,

arr_ts,

- (int16_t) packet_size, /* in bytes */

+ packet_size, /* in bytes */

index);

/* error check */