Fix problem with late packets in NetEq

Since r7255, it could happen that an old packet would block the decoding
process until enough packet was received for the buffer to flush. This
CL fixes that by:
- Partially reverting r7255;
- Remove recent old packets before taking a decision for GetAudio;
- Remove all old packets after a packet has been extracted for decoding;
- Adding tests for reordered packets.

BUG=chrome:423985
R=tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25079004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7612 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc

@@ -32,9 +32,11 @@ using ::testing::Return;
 using ::testing::ReturnNull;
 using ::testing::_;
 using ::testing::SetArgPointee;
+using ::testing::SetArrayArgument;
 using ::testing::InSequence;
 using ::testing::Invoke;
 using ::testing::WithArg;
+using ::testing::Pointee;
 
 namespace webrtc {
 
@@ -494,4 +496,95 @@ TEST_F(NetEqImplTest, VerifyTimestampPropagation) {
             static_cast<int>(sync_buffer->FutureLength()));
 }
 
+TEST_F(NetEqImplTest, ReorderedPacket) {
+  UseNoMocks();
+  CreateInstance();
+
+  const uint8_t kPayloadType = 17;   // Just an arbitrary number.
+  const uint32_t kReceiveTime = 17;  // Value doesn't matter for this test.
+  const int kSampleRateHz = 8000;
+  const int kPayloadLengthSamples = 10 * kSampleRateHz / 1000;  // 10 ms.
+  const size_t kPayloadLengthBytes = kPayloadLengthSamples;
+  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;
+
+  // Create a mock decoder object.
+  MockAudioDecoder mock_decoder;
+  EXPECT_CALL(mock_decoder, Init()).WillRepeatedly(Return(0));
+  EXPECT_CALL(mock_decoder, IncomingPacket(_, kPayloadLengthBytes, _, _, _))
+      .WillRepeatedly(Return(0));
+  int16_t dummy_output[kPayloadLengthSamples] = {0};
+  // The below expectation will make the mock decoder write
+  // |kPayloadLengthSamples| zeros to the output array, and mark it as speech.
+  EXPECT_CALL(mock_decoder, Decode(Pointee(0), kPayloadLengthBytes, _, _))
+      .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
+                                          dummy_output + kPayloadLengthSamples),
+                      SetArgPointee<3>(AudioDecoder::kSpeech),
+                      Return(kPayloadLengthSamples)));
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->RegisterExternalDecoder(
+                &mock_decoder, kDecoderPCM16B, kPayloadType));
+
+  // Insert one packet.
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->InsertPacket(
+                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+
+  // Pull audio once.
+  const int kMaxOutputSize = 10 * kSampleRateHz / 1000;
+  int16_t output[kMaxOutputSize];
+  int samples_per_channel;
+  int num_channels;
+  NetEqOutputType type;
+  EXPECT_EQ(
+      NetEq::kOK,
+      neteq_->GetAudio(
+          kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
+  ASSERT_EQ(kMaxOutputSize, samples_per_channel);
+  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(kOutputNormal, type);
+
+  // Insert two more packets. The first one is out of order, and is already too
+  // old, the second one is the expected next packet.
+  rtp_header.header.sequenceNumber -= 1;
+  rtp_header.header.timestamp -= kPayloadLengthSamples;
+  payload[0] = 1;
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->InsertPacket(
+                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+  rtp_header.header.sequenceNumber += 2;
+  rtp_header.header.timestamp += 2 * kPayloadLengthSamples;
+  payload[0] = 2;
+  EXPECT_EQ(NetEq::kOK,
+            neteq_->InsertPacket(
+                rtp_header, payload, kPayloadLengthBytes, kReceiveTime));
+
+  // Expect only the second packet to be decoded (the one with "2" as the first
+  // payload byte).
+  EXPECT_CALL(mock_decoder, Decode(Pointee(2), kPayloadLengthBytes, _, _))
+      .WillOnce(DoAll(SetArrayArgument<2>(dummy_output,
+                                          dummy_output + kPayloadLengthSamples),
+                      SetArgPointee<3>(AudioDecoder::kSpeech),
+                      Return(kPayloadLengthSamples)));
+
+  // Pull audio once.
+  EXPECT_EQ(
+      NetEq::kOK,
+      neteq_->GetAudio(
+          kMaxOutputSize, output, &samples_per_channel, &num_channels, &type));
+  ASSERT_EQ(kMaxOutputSize, samples_per_channel);
+  EXPECT_EQ(1, num_channels);
+  EXPECT_EQ(kOutputNormal, type);
+
+  // Now check the packet buffer, and make sure it is empty, since the
+  // out-of-order packet should have been discarded.
+  EXPECT_TRUE(packet_buffer_->Empty());
+
+  EXPECT_CALL(mock_decoder, Die());
+}
+
 }  // namespace webrtc