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
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@ -71,7 +71,7 @@ class RawRtpPacket {
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class LostPackets {
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public:
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LostPackets(Clock* clock, uint32_t rtt_ms)
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LostPackets(Clock* clock, int64_t rtt_ms)
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: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
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debug_file_(fopen("PacketLossDebug.txt", "w")),
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loss_count_(0),
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@ -180,7 +180,7 @@ class LostPackets {
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int loss_count_;
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RtpPacketList packets_;
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Clock* clock_;
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uint32_t rtt_ms_;
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int64_t rtt_ms_;
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DISALLOW_IMPLICIT_CONSTRUCTORS(LostPackets);
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};
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@ -323,7 +323,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
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RtpPlayerImpl(PayloadSinkFactoryInterface* payload_sink_factory,
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const PayloadTypes& payload_types, Clock* clock,
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scoped_ptr<test::RtpFileReader>* packet_source,
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float loss_rate, uint32_t rtt_ms, bool reordering)
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float loss_rate, int64_t rtt_ms, bool reordering)
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: ssrc_handlers_(payload_sink_factory, payload_types),
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clock_(clock),
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next_rtp_time_(0),
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@ -468,7 +468,7 @@ class RtpPlayerImpl : public RtpPlayerInterface {
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RtpPlayerInterface* Create(const std::string& input_filename,
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PayloadSinkFactoryInterface* payload_sink_factory, Clock* clock,
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const PayloadTypes& payload_types, float loss_rate, uint32_t rtt_ms,
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const PayloadTypes& payload_types, float loss_rate, int64_t rtt_ms,
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bool reordering) {
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scoped_ptr<test::RtpFileReader> packet_source(test::RtpFileReader::Create(
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test::RtpFileReader::kRtpDump, input_filename));
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