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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/media/base/mediachannel.h
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@@ -681,7 +681,7 @@ struct MediaSenderInfo {
int packets_sent;
int packets_lost;
float fraction_lost;
int rtt_ms;
int64_t rtt_ms;
std::string codec_name;
std::vector<SsrcSenderInfo> local_stats;
std::vector<SsrcReceiverInfo> remote_stats;
@@ -948,7 +948,7 @@ struct BandwidthEstimationInfo {
int actual_enc_bitrate;
int retransmit_bitrate;
int transmit_bitrate;
int bucket_delay;
int64_t bucket_delay;
// The following stats are only valid when
// StatsOptions::include_received_propagation_stats is true.
int total_received_propagation_delta_ms;