VoE: Handle empty playout timestamp differently

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 725 matching lines @@
       _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
       _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
       _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
       _outputFileRecording(false),
       _outputExternalMedia(false),
       _inputExternalMediaCallbackPtr(NULL),
       _outputExternalMediaCallbackPtr(NULL),
       _timeStamp(0),  // This is just an offset, RTP module will add it's own
                       // random offset
       ntp_estimator_(Clock::GetRealTimeClock()),
-      jitter_buffer_playout_timestamp_(0),
       playout_timestamp_rtp_(0),
       playout_timestamp_rtcp_(0),
       playout_delay_ms_(0),
       _numberOfDiscardedPackets(0),
       send_sequence_number_(0),
       rtp_ts_wraparound_handler_(new rtc::TimestampWrapAroundHandler()),
       capture_start_rtp_time_stamp_(-1),
       capture_start_ntp_time_ms_(-1),
       _engineStatisticsPtr(NULL),
       _outputMixerPtr(NULL),
@@ skipping 2524 matching lines @@
                  ") != "
                  "fileSamples(%" PRIuS ")",
                  audioFrame.samples_per_channel_, fileSamples);
     return -1;
   }
 
   return 0;
 }
 
 void Channel::UpdatePlayoutTimestamp(bool rtcp) {
-  rtc::Optional<uint32_t> playout_timestamp = audio_coding_->PlayoutTimestamp();
+  jitter_buffer_playout_timestamp_ = audio_coding_->PlayoutTimestamp();
 
-  if (!playout_timestamp) {
-    // This can happen if this channel has not been received any RTP packet. In
-    // this case, NetEq is not capable of computing playout timestamp.
+  if (!jitter_buffer_playout_timestamp_) {
+    // This can happen if this channel has not received any RTP packets. In
+    // this case, NetEq is not capable of computing a playout timestamp.
     return;
   }
 
   uint16_t delay_ms = 0;
   if (_audioDeviceModulePtr->PlayoutDelay(&delay_ms) == -1) {
     WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, _channelId),
                  "Channel::UpdatePlayoutTimestamp() failed to read playout"
                  " delay from the ADM");
     _engineStatisticsPtr->SetLastError(
         VE_CANNOT_RETRIEVE_VALUE, kTraceError,
         "UpdatePlayoutTimestamp() failed to retrieve playout delay");
     return;
   }
 
-  jitter_buffer_playout_timestamp_ = *playout_timestamp;
+  RTC_DCHECK(jitter_buffer_playout_timestamp_);
+  uint32_t playout_timestamp = *jitter_buffer_playout_timestamp_;
 
   // Remove the playout delay.
-  *playout_timestamp -= (delay_ms * (GetPlayoutFrequency() / 1000));
+  playout_timestamp -= (delay_ms * (GetPlayoutFrequency() / 1000));
 
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::UpdatePlayoutTimestamp() => playoutTimestamp = %lu",
-               *playout_timestamp);
+               playout_timestamp);
 
   {
     rtc::CritScope lock(&video_sync_lock_);
     if (rtcp) {
-      playout_timestamp_rtcp_ = *playout_timestamp;
+      playout_timestamp_rtcp_ = playout_timestamp;
     } else {
-      playout_timestamp_rtp_ = *playout_timestamp;
+      playout_timestamp_rtp_ = playout_timestamp;
     }
     playout_delay_ms_ = delay_ms;
   }
 }
 
 // Called for incoming RTP packets after successful RTP header parsing.
 void Channel::UpdatePacketDelay(uint32_t rtp_timestamp,
                                 uint16_t sequence_number) {
   WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::UpdatePacketDelay(timestamp=%lu, sequenceNumber=%u)",
                rtp_timestamp, sequence_number);
 
   // Get frequency of last received payload
   int rtp_receive_frequency = GetPlayoutFrequency();
 
   // |jitter_buffer_playout_timestamp_| updated in UpdatePlayoutTimestamp for
-  // every incoming packet.
-  uint32_t timestamp_diff_ms =
-      (rtp_timestamp - jitter_buffer_playout_timestamp_) /
-      (rtp_receive_frequency / 1000);
-  if (!IsNewerTimestamp(rtp_timestamp, jitter_buffer_playout_timestamp_) ||
-      timestamp_diff_ms > (2 * kVoiceEngineMaxMinPlayoutDelayMs)) {
+  // every incoming packet. May be empty if no valid playout timestamp is
+  // available.
+  uint32_t timestamp_diff_ms = 0;
+  if (jitter_buffer_playout_timestamp_ &&
+      IsNewerTimestamp(rtp_timestamp, *jitter_buffer_playout_timestamp_)) {
     // If |jitter_buffer_playout_timestamp_| is newer than the incoming RTP
     // timestamp, the resulting difference is negative, but is set to zero.

[the sun, 2016/04/05 14:06:29]

AFAICT this comment is wrong now - you're not setting the difference to zero if
it is negative.

[hlundin-webrtc, 2016/04/05 14:35:15]

Implicitly I still do. I changed the negation before IsNewerTimestamp, so now
the code reads as "if the diff is positive, set timestamp_diff_ms to something
other than the initial zero value."

I updated the comment.

     // This can happen when a network glitch causes a packet to arrive late,
     // and during long comfort noise periods with clock drift.
-    timestamp_diff_ms = 0;
+    timestamp_diff_ms = (rtp_timestamp - *jitter_buffer_playout_timestamp_) /
+                        (rtp_receive_frequency / 1000);
+    if (timestamp_diff_ms > (2 * kVoiceEngineMaxMinPlayoutDelayMs)) {
+      // Diff is too large; set it to zero instead.
+      timestamp_diff_ms = 0;
+    }
   }
 
   uint16_t packet_delay_ms =
       (rtp_timestamp - _previousTimestamp) / (rtp_receive_frequency / 1000);
 
   _previousTimestamp = rtp_timestamp;
 
   if (timestamp_diff_ms == 0)
     return;
 
@@ skipping 166 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc