Renamed constant and added comments.
Review URL: https://webrtc-codereview.appspot.com/847004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2825 4adac7df-926f-26a2-2b94-8c16560cd09d
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@ -15,6 +15,24 @@
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#include "rtp_utility.h"
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namespace webrtc {
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namespace {
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// Factor that is applied to the target bitrate to calculate the number of
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// bytes that can be transmitted per interval.
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// Increasing this factor will result in lower delays in cases of bitrate
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// overshoots.
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const float kBytesPerIntervalMargin = 1.5f;
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// Time limit in ms between packets within a frame.
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// A packet will be transmitted if the elapsed time since the last transmitted
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// packet (for packets within same frames) has exceed this time limit.
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const int kPacketLimitMs = 5;
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// Time limit factor between frames.
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// A packet in a new frame will be transmitted if the elapsed time since the
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// last transmitted packet in the previous frame has exceeded the time
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// difference for when the packets were stored, multiplied by this factor.
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const float kFrameLimitFactor = 1.2f;
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} // namespace
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TransmissionBucket::TransmissionBucket(RtpRtcpClock* clock)
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: clock_(clock),
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@ -57,9 +75,8 @@ void TransmissionBucket::UpdateBytesPerInterval(
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uint16_t target_bitrate_kbps) {
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webrtc::CriticalSectionScoped cs(*critsect_);
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const float kMargin = 1.5f;
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uint32_t bytes_per_interval =
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kMargin * (target_bitrate_kbps * delta_time_ms / 8);
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kBytesPerIntervalMargin * (target_bitrate_kbps * delta_time_ms / 8);
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if (bytes_rem_interval_ < 0) {
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bytes_rem_interval_ += bytes_per_interval;
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@ -109,7 +126,6 @@ bool TransmissionBucket::SameFrameAndPacketIntervalTimeElapsed(
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// Not same frame.
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return false;
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}
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const int kPacketLimitMs = 5;
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if ((clock_->GetTimeInMS() - last_transmitted_packet_.transmitted_ms) <
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kPacketLimitMs) {
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// Time has not elapsed.
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@ -128,7 +144,6 @@ bool TransmissionBucket::NewFrameAndFrameIntervalTimeElapsed(
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// Not a new frame.
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return false;
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}
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const float kFrameLimitFactor = 1.2f;
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if ((clock_->GetTimeInMS() - last_transmitted_packet_.transmitted_ms) <
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kFrameLimitFactor *
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(current_packet.stored_ms - last_transmitted_packet_.stored_ms)) {
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