Fixed problems in neteq when RTP and decoder timestamps increment with

webrtc/modules/audio_coding/neteq/neteq_impl.cc

Index: webrtc/modules/audio_coding/neteq/neteq_impl.cc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl.cc b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
index 786cb84aa3ed8beffb5d6d6bb9f7db8d74267afc..1dc6cb43b831d503a895ba034b6ff6381758349a 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -594,6 +594,20 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     return packet;
   }());
 
+  if ((rtp_header.header.ssrc != ssrc_) || first_packet_) {

[ossu, 2017/03/13 15:03:17]

It seems to me that update_sample_rate_and_channels captures first_packet or
ssrc change, so perhaps refactor as such:

  bool update_sample_rate_and_channels = first_packet_ ||
(rtp_header.header.ssrc != ssrc_);
  if (update_sample_rate_and_channels) {
    timestamp_scaler_->Reset();
  }

[...] (ToInternal, copy values, etc.)

  if (update_sample_rate_and_channels) {
    rtcp_.Init(...);
  }

+    // Reset timestamp scaling.
+    timestamp_scaler_->Reset();
+  }
+
+  // Scale timestamp to internal domain (only for some codecs).
+  timestamp_scaler_->ToInternal(&packet_list);
+
+  // Store these for later use, since the first packet may very well disappear
+  // before we need these values.
+  const uint32_t main_timestamp = packet_list.front().timestamp;
+  const uint8_t main_payload_type = packet_list.front().payload_type;
+  const uint16_t main_sequence_number = packet_list.front().sequence_number;
+
   bool update_sample_rate_and_channels = false;
   // Reinitialize NetEq if it's needed (changed SSRC or first call).
   if ((rtp_header.header.ssrc != ssrc_) || first_packet_) {
@@ -610,14 +624,10 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     ssrc_ = rtp_header.header.ssrc;
 
     // Update audio buffer timestamp.
-    sync_buffer_->IncreaseEndTimestamp(rtp_header.header.timestamp -
-                                       timestamp_);
+    sync_buffer_->IncreaseEndTimestamp(main_timestamp - timestamp_);
 
     // Update codecs.
-    timestamp_ = rtp_header.header.timestamp;
-
-    // Reset timestamp scaling.
-    timestamp_scaler_->Reset();
+    timestamp_ = main_timestamp;
 
     // Trigger an update of sampling rate and the number of channels.
     update_sample_rate_and_channels = true;
@@ -652,14 +662,6 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   }
 
   RTC_DCHECK(!packet_list.empty());
-  // Store these for later use, since the first packet may very well disappear
-  // before we need these values.
-  const uint32_t main_timestamp = packet_list.front().timestamp;
-  const uint8_t main_payload_type = packet_list.front().payload_type;
-  const uint16_t main_sequence_number = packet_list.front().sequence_number;
-
-  // Scale timestamp to internal domain (only for some codecs).
-  timestamp_scaler_->ToInternal(&packet_list);
 
   // Process DTMF payloads. Cycle through the list of packets, and pick out any
   // DTMF payloads found.