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Wire up CallStats to provide modules with correct RTT.

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BUG=769
TEST=Manual test since there is no ViE APi to get RTT for receive channels.

Review URL: https://webrtc-codereview.appspot.com/937027

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3163 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
mflodman@webrtc.org
2012-11-26 12:40:15 +00:00
parent 5ba3decc94
commit 7c894b7cc7
11 changed files with 123 additions and 42 deletions
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12
webrtc/modules/video_coding/main/source/jitter_buffer.cc
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@@ -25,6 +25,9 @@
namespace webrtc {
// Use this rtt if no value has been reported.
static uint32_t kDefaultRtt = 200;
// Predicates used when searching for frames in the frame buffer list
class FrameSmallerTimestamp {
public:
@@ -88,7 +91,7 @@ VCMJitterBuffer::VCMJitterBuffer(TickTimeBase* clock,
num_discarded_packets_(0),
jitter_estimate_(vcm_id, receiver_id),
inter_frame_delay_(clock_->MillisecondTimestamp()),
rtt_ms_(0),
rtt_ms_(kDefaultRtt),
nack_mode_(kNoNack),
low_rtt_nack_threshold_ms_(-1),
high_rtt_nack_threshold_ms_(-1),
@@ -190,7 +193,7 @@ void VCMJitterBuffer::Start() {
first_packet_ = true;
nack_seq_nums_length_ = 0;
waiting_for_key_frame_ = false;
rtt_ms_ = 0;
rtt_ms_ = kDefaultRtt;
num_not_decodable_packets_ = 0;
WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
@@ -796,6 +799,11 @@ void VCMJitterBuffer::SetNackMode(VCMNackMode mode,
assert(low_rtt_nack_threshold_ms > -1 || high_rtt_nack_threshold_ms == -1);
low_rtt_nack_threshold_ms_ = low_rtt_nack_threshold_ms;
high_rtt_nack_threshold_ms_ = high_rtt_nack_threshold_ms;
// Don't set a high start rtt if high_rtt_nack_threshold_ms_ is used, to not
// disable NACK in hybrid mode.
if (rtt_ms_ == kDefaultRtt && high_rtt_nack_threshold_ms_ != -1) {
rtt_ms_ = 0;
}
if (nack_mode_ == kNoNack) {
jitter_estimate_.ResetNackCount();
}