Update tests and DTX check for Opus 1.1.3.

webrtc/modules/audio_coding/codecs/opus/opus_interface.c

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 77 matching lines @@
   if (samples > 48 * kWebRtcOpusMaxEncodeFrameSizeMs) {
     return -1;
   }
 
   res = opus_encode(inst->encoder,
                     (const opus_int16*)audio_in,
                     (int)samples,
                     encoded,
                     (opus_int32)length_encoded_buffer);
 
-  if (res == 1) {
+  if (res <= 0) {
+    return -1;
+  }
+
+  if (res <= 2) {
     // Indicates DTX since the packet has nothing but a header. In principle,
     // there is no need to send this packet. However, we do transmit the first
     // occurrence to let the decoder know that the encoder enters DTX mode.
     if (inst->in_dtx_mode) {
       return 0;
     } else {
       inst->in_dtx_mode = 1;
       return 1;
     }
-  } else if (res > 1) {
-    inst->in_dtx_mode = 0;
-    return res;
   }
 
-  return -1;
+  inst->in_dtx_mode = 0;
+  return res;
 }
 
 int16_t WebRtcOpus_SetBitRate(OpusEncInst* inst, int32_t rate) {
   if (inst) {
     return opus_encoder_ctl(inst->encoder, OPUS_SET_BITRATE(rate));
   } else {
     return -1;
   }
 }
 
@@ skipping 328 matching lines @@
     return 0;
   }
 
   for (n = 0; n < channels; n++) {
     if (frame_data[0][0] & (0x80 >> ((n + 1) * (frames + 1) - 1)))
       return 1;
   }
 
   return 0;
 }