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NetEq jitter calculation now done in uint64_t.

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The timestamps are 32 bit and can (conceivably) be spaced far enough
apart for the calculation, which is done in Q4, to overflow.

BUG=chromium:653268

Review-Url: https://codereview.webrtc.org/2460393002
Cr-Commit-Position: refs/heads/master@{#14856}
This commit is contained in:
ossu 2016-10-31 08:59:26 -07:00 committed by Commit bot
parent 80ac24dd36
commit 6b6c88f184
2 changed files with 4 additions and 4 deletions
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6
webrtc/modules/audio_coding/neteq/rtcp.cc
View File

@ -10,11 +10,11 @@
#include "webrtc/modules/audio_coding/neteq/rtcp.h"
#include <stdlib.h>
#include <string.h>
#include <algorithm>
#include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
@ -46,10 +46,10 @@ void Rtcp::Update(const RTPHeader& rtp_header, uint32_t receive_timestamp) {
// Note that the value in |jitter_| is in Q4.
if (received_packets_ > 1) {
int32_t ts_diff = receive_timestamp - (rtp_header.timestamp - transit_);
ts_diff = WEBRTC_SPL_ABS_W32(ts_diff);
int32_t jitter_diff = (ts_diff << 4) - jitter_;
int64_t jitter_diff = (std::abs(int64_t{ts_diff}) << 4) - jitter_;
// Calculate 15 * jitter_ / 16 + jitter_diff / 16 (with proper rounding).
jitter_ = jitter_ + ((jitter_diff + 8) >> 4);
RTC_DCHECK_GE(jitter_, 0);
}
transit_ = rtp_header.timestamp - receive_timestamp;
}

2
webrtc/modules/audio_coding/neteq/rtcp.h
View File

@ -48,7 +48,7 @@ class Rtcp {
// report was generated.
uint32_t expected_prior_; // Expected number of packets, at the time of the
// last report.
uint32_t jitter_; // Current jitter value.
int64_t jitter_; // Current jitter value in Q4.
int32_t transit_; // Clock difference for previous packet.
RTC_DISALLOW_COPY_AND_ASSIGN(Rtcp);