This is to get rid of a bug relating to the return of NULL in calling GetDecoder when there are DTMF packets.

BUG=3140
TEST=trybots
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10929006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5830 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/modules/audio_coding/neteq4/neteq_impl.cc

@@ -512,19 +512,6 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     memcpy(&main_header, &packet_list.front()->header, sizeof(main_header));
   }
 
-  // Check for FEC in packets, and separate payloads into several packets.
-  int ret = payload_splitter_->SplitFec(&packet_list, decoder_database_.get());
-  if (ret != PayloadSplitter::kOK) {
-    LOG_FERR1(LS_WARNING, SplitFec, packet_list.size());
-    PacketBuffer::DeleteAllPackets(&packet_list);
-    switch (ret) {
-      case PayloadSplitter::kUnknownPayloadType:
-        return kUnknownRtpPayloadType;
-      default:
-        return kOtherError;
-    }
-  }
-
   // Check payload types.
   if (decoder_database_->CheckPayloadTypes(packet_list) ==
       DecoderDatabase::kDecoderNotFound) {
@@ -571,6 +558,19 @@ int NetEqImpl::InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
     }
   }
 
+  // Check for FEC in packets, and separate payloads into several packets.
+  int ret = payload_splitter_->SplitFec(&packet_list, decoder_database_.get());
+  if (ret != PayloadSplitter::kOK) {
+    LOG_FERR1(LS_WARNING, SplitFec, packet_list.size());
+    PacketBuffer::DeleteAllPackets(&packet_list);
+    switch (ret) {
+      case PayloadSplitter::kUnknownPayloadType:
+        return kUnknownRtpPayloadType;
+      default:
+        return kOtherError;
+    }
+  }
+
   // Split payloads into smaller chunks. This also verifies that all payloads
   // are of a known payload type. SplitAudio() method is protected against
   // sync-packets.

webrtc/modules/audio_coding/neteq4/payload_splitter.cc

@@ -140,7 +140,10 @@ int PayloadSplitter::SplitFec(PacketList* packet_list,
 
     // Not an FEC packet.
     AudioDecoder* decoder = decoder_database->GetDecoder(payload_type);
-    if (!decoder->PacketHasFec(packet->payload, packet->payload_length)) {
+    // decoder should not return NULL.
+    assert(decoder != NULL);
+    if (!decoder ||
+        !decoder->PacketHasFec(packet->payload, packet->payload_length)) {
       ++it;
       continue;
     }
@@ -152,8 +155,7 @@ int PayloadSplitter::SplitFec(PacketList* packet_list,
 
         new_packet->header = packet->header;
         int duration = decoder->
-            PacketDurationRedundant(packet->payload,
+            PacketDurationRedundant(packet->payload, packet->payload_length);
-                                    packet->payload_length) * 3 / 2;
         new_packet->header.timestamp -= duration;
         new_packet->payload = new uint8_t[packet->payload_length];
         memcpy(new_packet->payload, packet->payload, packet->payload_length);

webrtc/modules/audio_coding/neteq4/payload_splitter_unittest.cc

@@ -743,7 +743,7 @@ TEST(FecPayloadSplitter, MixedPayload) {
   // Check first packet.
   packet = packet_list.front();
   EXPECT_EQ(0, packet->header.payloadType);
-  EXPECT_EQ(kBaseTimestamp - 20 * 48, packet->header.timestamp);
+  EXPECT_EQ(kBaseTimestamp - 20 * 32, packet->header.timestamp);
   EXPECT_EQ(10, packet->payload_length);
   EXPECT_FALSE(packet->primary);
   delete [] packet->payload;

webrtc/modules/audio_coding/neteq4/timestamp_scaler.cc

@@ -55,8 +55,7 @@ uint32_t TimestampScaler::ToInternal(uint32_t external_timestamp,
       // Use timestamp scaling with factor 2/3 (32 kHz sample rate, but RTP
       // timestamps run on 48 kHz).
       // TODO(tlegrand): Remove scaling for kDecoderCNGswb48kHz once ACM has
-      // full 48 kHz support. Change also ought to be made in
+      // full 48 kHz support.
-      // PayloadSplitter::SplitFec().
       numerator_ = 2;
       denominator_ = 3;
     }