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

webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc

Index: webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
diff --git a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
index 006a5ad542397cb839b2fe70bd5e59c093fbdbc8..ab0b4bf7cd1ff6c2d2b432004786968fc6063e9f 100644
--- a/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
@@ -907,4 +907,42 @@ TEST_F(NetEqImplTest, UnsupportedDecoder) {
   EXPECT_EQ(kChannels, num_channels);
 }
 
+// This test inserts packets until the buffer is flushed. After that, it asks
+// NetEq for the network statistics. The purpose of the test is to make sure
+// that even though the buffer size increment is negative (which it becomes when
+// the packet causing a flush is inserted), the packet length stored in the
+// decision logic remains valid.
+TEST_F(NetEqImplTest, FloodBufferAndGetNetworkStats) {
+  UseNoMocks();
+  CreateInstance();
+
+  const int kPayloadLengthSamples = 80;

[Peter Kasting, 2015/08/27 08:45:33]

Nit: Use size_t here...

[hlundin-webrtc, 2015/08/27 09:54:03]

Done.

+  const size_t kPayloadLengthBytes = 2 * kPayloadLengthSamples;  // PCM 16-bit.
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
+  uint8_t payload[kPayloadLengthBytes] = {0};
+  WebRtcRTPHeader rtp_header;
+  rtp_header.header.payloadType = kPayloadType;
+  rtp_header.header.sequenceNumber = 0x1234;
+  rtp_header.header.timestamp = 0x12345678;
+  rtp_header.header.ssrc = 0x87654321;
+
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->RegisterPayloadType(kDecoderPCM16B, kPayloadType));
+
+  // Insert packets until the buffer flushes.
+  for (size_t i = 0; i <= config_.max_packets_in_buffer; ++i) {
+    EXPECT_EQ(i, packet_buffer_->NumPacketsInBuffer());
+    EXPECT_EQ(NetEq::kOK,
+              neteq_->InsertPacket(rtp_header, payload, kPayloadLengthBytes,
+                                   kReceiveTime));
+    rtp_header.header.timestamp += kPayloadLengthSamples;

[Peter Kasting, 2015/08/27 08:45:33]

Nit: ...and cast to uint32_t here

[hlundin-webrtc, 2015/08/27 09:54:03]

Done.

+    rtp_header.header.sequenceNumber += 1;

[Peter Kasting, 2015/08/27 08:45:33]

Nit: ++rtp_header.header.sequenceNumber;

[hlundin-webrtc, 2015/08/27 09:54:03]

Done.

+  }
+  EXPECT_EQ(1u, packet_buffer_->NumPacketsInBuffer());
+
+  // Ask for network statistics. This should not crash.
+  NetEqNetworkStatistics stats;
+  EXPECT_EQ(NetEq::kOK, neteq_->NetworkStatistics(&stats));
+}
 }  // namespace webrtc