NetEq: Fixing a bug that caused rtc::checked_cast to trigger

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 22e71f7f1d8af82f80d3f46c38cf465c84da7f4e..1afc165bf6bc8dc4a0848059b37fafb5e619a97c 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl.cc
@@ -599,7 +599,8 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   }
 
   // Insert packets in buffer.
-  size_t temp_bufsize = packet_buffer_->NumPacketsInBuffer();
+  const size_t buffer_length_before_insert =
+      packet_buffer_->NumPacketsInBuffer();
   ret = packet_buffer_->InsertPacketList(
       &packet_list,
       *decoder_database_,
@@ -655,14 +656,18 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
   delay_manager_->LastDecoderType(dec_info->codec_type);
   if (delay_manager_->last_pack_cng_or_dtmf() == 0) {
     // Calculate the total speech length carried in each packet.
-    temp_bufsize = packet_buffer_->NumPacketsInBuffer() - temp_bufsize;
-    temp_bufsize *= decoder_frame_length_;
-
-    if ((temp_bufsize > 0) &&

[minyue-webrtc, 2015/08/27 09:09:30]

Since we introduced size_t - size_t here, it should not have been difficult to
detect when changing the datatype. But the preceding CL was huge and might not
be easy.

Hope there are no other errors.

[Peter Kasting, 2015/08/27 09:14:54]

I wouldn't have caught this even had I just done this portion of the change. 
InsertPacketList() sounds as if it will either increase the buffer size or do
nothing, and the "> 0" check here looks as if it's distinguishing from the "0"
case rather than from the negative number case.

(I did look as a result of this CL to make sure InsertPacketList() documents
that it can decrease the buffer size.  I thought about suggesting it be called
InsertPacketsAndMaybeFlush() or something like that, but it seems awkward.)

-        (temp_bufsize != decision_logic_->packet_length_samples())) {
-      decision_logic_->set_packet_length_samples(temp_bufsize);
-      delay_manager_->SetPacketAudioLength(
-          static_cast<int>((1000 * temp_bufsize) / fs_hz_));
+    const size_t buffer_length_after_insert =
+        packet_buffer_->NumPacketsInBuffer();
+
+    if (buffer_length_after_insert > buffer_length_before_insert) {
+      const size_t packet_length_samples =
+          (buffer_length_after_insert - buffer_length_before_insert) *
+          decoder_frame_length_;
+      if (packet_length_samples != decision_logic_->packet_length_samples()) {
+        decision_logic_->set_packet_length_samples(packet_length_samples);
+        delay_manager_->SetPacketAudioLength(
+            rtc::checked_cast<int>((1000 * packet_length_samples) / fs_hz_));
+      }
     }
 
     // Update statistics.