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

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

View File

@ -14,18 +14,13 @@
#include "webrtc/system_wrappers/interface/logging.h"
namespace webrtc {
namespace {
// Used in determining whether a frame is decodable.
enum {kRttThreshold = 100}; // Not decodable if Rtt is lower than this.
// Do not decode frames if the number of packets is between these two
// thresholds.
static const float kLowPacketPercentageThreshold = 0.2f;
static const float kHighPacketPercentageThreshold = 0.8f;
namespace {
uint16_t BufferToUWord16(const uint8_t* dataBuffer) {
return (dataBuffer[0] << 8) | dataBuffer[1];
}
} // namespace
VCMSessionInfo::VCMSessionInfo()
@ -233,6 +228,12 @@ void VCMSessionInfo::UpdateDecodableSession(const FrameData& frame_data) {
return;
// TODO(agalusza): Account for bursty loss.
// TODO(agalusza): Refine these values to better approximate optimal ones.
// Do not decode frames if the RTT is lower than this.
const int64_t kRttThreshold = 100;
// Do not decode frames if the number of packets is between these two
// thresholds.
const float kLowPacketPercentageThreshold = 0.2f;
const float kHighPacketPercentageThreshold = 0.8f;
if (frame_data.rtt_ms < kRttThreshold
|| frame_type_ == kVideoFrameKey
|| !HaveFirstPacket()