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Use int64_t more consistently for times, in particular for RTT values.

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Existing code was inconsistent about whether to use uint16_t, int, unsigned int,
or uint32_t, and sometimes silently truncated one to another, or truncated
int64_t.  Because most core time-handling functions use int64_t, being
consistent about using int64_t unless otherwise necessary minimizes the number
of explicit or implicit casts.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, holmer@google.com, tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8045 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
pkasting@chromium.org
2015-01-12 21:51:21 +00:00
parent a7add19cf4
commit 16825b1a82
124 changed files with 422 additions and 417 deletions
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4
talk/app/webrtc/java/jni/peerconnection_jni.cc
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@ -1271,7 +1271,7 @@ class MediaCodecVideoEncoder : public webrtc::VideoEncoder,
webrtc::EncodedImageCallback* callback) OVERRIDE;
virtual int32_t Release() OVERRIDE;
virtual int32_t SetChannelParameters(uint32_t /* packet_loss */,
int /* rtt */) OVERRIDE;
int64_t /* rtt */) OVERRIDE;
virtual int32_t SetRates(uint32_t new_bit_rate, uint32_t frame_rate) OVERRIDE;
// rtc::MessageHandler implementation.
@ -1472,7 +1472,7 @@ int32_t MediaCodecVideoEncoder::Release() {
}
int32_t MediaCodecVideoEncoder::SetChannelParameters(uint32_t /* packet_loss */,
int /* rtt */) {
int64_t /* rtt */) {
return WEBRTC_VIDEO_CODEC_OK;
}