Roll chromium_revision a8e5140..c6076f2 (372922:372974) incl. update to Opus v.1.1.2

webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <string>
@@ skipping 188 matching lines @@
 
   // We run |kRunTimeMs| milliseconds of pure silence.
   const int kRunTimeMs = 2000;
 
   // We check that, after a |kCheckTimeMs| milliseconds (given that the CNG in
   // Opus needs time to adapt), the absolute values of DTX decoded signal are
   // bounded by |kOutputValueBound|.
   const int kCheckTimeMs = 1500;
 
 #if defined(OPUS_FIXED_POINT)
-  const uint16_t kOutputValueBound = 20;
+  const uint16_t kOutputValueBound = 30;
 #else
-  const uint16_t kOutputValueBound = 2;
+  const uint16_t kOutputValueBound = 8;
 #endif
 
   int time = 0;
   while (time < kRunTimeMs) {
     // DTX mode is maintained for maximum |max_dtx_frames| frames.
     int i = 0;
     for (; i < max_dtx_frames; ++i) {
       time += block_length_ms;
       EXPECT_EQ(samples,
                 static_cast<size_t>(EncodeDecode(
@@ skipping 443 matching lines @@
   EXPECT_EQ(0, WebRtcOpus_EncoderFree(opus_encoder_));
   EXPECT_EQ(0, WebRtcOpus_DecoderFree(opus_decoder_));
 }
 
 INSTANTIATE_TEST_CASE_P(VariousMode,
                         OpusTest,
                         Combine(Values(1, 2), Values(0, 1)));
 
 
 }  // namespace webrtc