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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@ -345,7 +345,7 @@ void StreamStatisticianImpl::LastReceiveTimeNtp(uint32_t* secs,
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}
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bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
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const RTPHeader& header, int min_rtt) const {
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const RTPHeader& header, int64_t min_rtt) const {
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CriticalSectionScoped cs(stream_lock_.get());
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if (InOrderPacketInternal(header.sequenceNumber)) {
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return false;
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@ -358,17 +358,16 @@ bool StreamStatisticianImpl::IsRetransmitOfOldPacket(
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// Diff in time stamp since last received in order.
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uint32_t timestamp_diff = header.timestamp - last_received_timestamp_;
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int32_t rtp_time_stamp_diff_ms = static_cast<int32_t>(timestamp_diff) /
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frequency_khz;
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uint32_t rtp_time_stamp_diff_ms = timestamp_diff / frequency_khz;
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int32_t max_delay_ms = 0;
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int64_t max_delay_ms = 0;
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if (min_rtt == 0) {
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// Jitter standard deviation in samples.
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float jitter_std = sqrt(static_cast<float>(jitter_q4_ >> 4));
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// 2 times the standard deviation => 95% confidence.
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// And transform to milliseconds by dividing by the frequency in kHz.
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max_delay_ms = static_cast<int32_t>((2 * jitter_std) / frequency_khz);
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max_delay_ms = static_cast<int64_t>((2 * jitter_std) / frequency_khz);
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// Min max_delay_ms is 1.
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if (max_delay_ms == 0) {
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