Index: webrtc/modules/audio_coding/codecs/g711/g711_interface.c |
diff --git a/webrtc/modules/audio_coding/codecs/g711/g711_interface.c b/webrtc/modules/audio_coding/codecs/g711/g711_interface.c |
index b5795209f70bf35c07cd7531cbdb3822fa91f013..5b96a9c5553f41747f9c13d85116d7ec09053b1b 100644 |
--- a/webrtc/modules/audio_coding/codecs/g711/g711_interface.c |
+++ b/webrtc/modules/audio_coding/codecs/g711/g711_interface.c |
@@ -12,40 +12,40 @@ |
#include "g711_interface.h" |
#include "webrtc/typedefs.h" |
-int16_t WebRtcG711_EncodeA(const int16_t* speechIn, |
- int16_t len, |
- uint8_t* encoded) { |
- int n; |
+size_t WebRtcG711_EncodeA(const int16_t* speechIn, |
+ size_t len, |
+ uint8_t* encoded) { |
+ size_t n; |
for (n = 0; n < len; n++) |
encoded[n] = linear_to_alaw(speechIn[n]); |
return len; |
} |
-int16_t WebRtcG711_EncodeU(const int16_t* speechIn, |
- int16_t len, |
- uint8_t* encoded) { |
- int n; |
+size_t WebRtcG711_EncodeU(const int16_t* speechIn, |
+ size_t len, |
+ uint8_t* encoded) { |
+ size_t n; |
for (n = 0; n < len; n++) |
encoded[n] = linear_to_ulaw(speechIn[n]); |
return len; |
} |
-int16_t WebRtcG711_DecodeA(const uint8_t* encoded, |
- int16_t len, |
- int16_t* decoded, |
- int16_t* speechType) { |
- int n; |
+size_t WebRtcG711_DecodeA(const uint8_t* encoded, |
+ size_t len, |
+ int16_t* decoded, |
+ int16_t* speechType) { |
+ size_t n; |
for (n = 0; n < len; n++) |
decoded[n] = alaw_to_linear(encoded[n]); |
*speechType = 1; |
return len; |
} |
-int16_t WebRtcG711_DecodeU(const uint8_t* encoded, |
- int16_t len, |
- int16_t* decoded, |
- int16_t* speechType) { |
- int n; |
+size_t WebRtcG711_DecodeU(const uint8_t* encoded, |
+ size_t len, |
+ int16_t* decoded, |
+ int16_t* speechType) { |
+ size_t n; |
for (n = 0; n < len; n++) |
decoded[n] = ulaw_to_linear(encoded[n]); |
*speechType = 1; |