Fix two UBSan warnings in NetEq

webrtc/modules/audio_coding/neteq/delay_manager.cc

Index: webrtc/modules/audio_coding/neteq/delay_manager.cc
diff --git a/webrtc/modules/audio_coding/neteq/delay_manager.cc b/webrtc/modules/audio_coding/neteq/delay_manager.cc
index 806d02b8deb538184d542e419b49c5fec37b5da9..593249480ab06add7948ee880b8aae3162dd3f19 100644
--- a/webrtc/modules/audio_coding/neteq/delay_manager.cc
+++ b/webrtc/modules/audio_coding/neteq/delay_manager.cc
@@ -15,6 +15,7 @@
 
 #include <algorithm>  // max, min
 
+#include "webrtc/base/safe_conversions.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/modules/audio_coding/neteq/delay_peak_detector.h"
 #include "webrtc/modules/include/module_common_types.h"
@@ -93,10 +94,11 @@ int DelayManager::Update(uint16_t sequence_number,
     packet_len_ms = packet_len_ms_;
   } else {
     // Calculate timestamps per packet and derive packet length in ms.
-    int packet_len_samp =
+    int64_t packet_len_samp =
         static_cast<uint32_t>(timestamp - last_timestamp_) /
         static_cast<uint16_t>(sequence_number - last_seq_no_);
-    packet_len_ms = (1000 * packet_len_samp) / sample_rate_hz;
+    packet_len_ms =
+        rtc::saturated_cast<int>(1000 * packet_len_samp / sample_rate_hz);

[kwiberg-webrtc, 2016/02/10 15:14:21]

Am I guessing right when I think that this calculation is guaranteed to never
overflow in real life? packet_len_samp is at most 2^32 - 1, and sample_rate_hz >
2000.

Hmm. If so, a checked_cast may be better?

[hlundin-webrtc, 2016/02/10 15:36:13]

Done.

   }
 
   if (packet_len_ms > 0) {