--- a/media/webrtc/trunk/gtest/moz.build
+++ b/media/webrtc/trunk/gtest/moz.build
@@ -152,17 +152,16 @@ if CONFIG['OS_TARGET'] in ['Darwin', 'Li
         '../webrtc/modules/audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc',
         '../webrtc/modules/audio_coding/codecs/isac/fix/source/transform_unittest.cc',
         '../webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc',
     # Requires external files
     #    '../webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc',
     # Requires external files
     #    '../webrtc/modules/audio_coding/codecs/isac/main/source/isac_unittest.cc',
     # Requires external files
-    #    '../webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc',
         '../webrtc/modules/audio_coding/neteq/audio_multi_vector_unittest.cc',
         '../webrtc/modules/audio_coding/neteq/audio_vector_unittest.cc',
         '../webrtc/modules/audio_coding/neteq/background_noise_unittest.cc',
         '../webrtc/modules/audio_coding/neteq/buffer_level_filter_unittest.cc',
         '../webrtc/modules/audio_coding/neteq/comfort_noise_unittest.cc',
         '../webrtc/modules/audio_coding/neteq/decision_logic_unittest.cc',
         '../webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc',
         '../webrtc/modules/audio_coding/neteq/delay_manager_unittest.cc',

--- a/media/webrtc/trunk/webrtc/modules/audio_coding/BUILD.gn
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/BUILD.gn
@@ -747,30 +747,20 @@ source_set("webrtc_opus") {
     public_deps = [
       rtc_opus_dir,
     ]
   } else if (build_with_mozilla) {
     include_dirs = [ getenv("DIST") + "/include/opus" ]
   }
 }
 
-config("neteq_config") {
-  include_dirs = [
-    # Need Opus header files for the audio classifier.
-    "//third_party/opus/src/celt",
-    "//third_party/opus/src/src",
-  ]
-}
-
 source_set("neteq") {
   sources = [
     "neteq/accelerate.cc",
     "neteq/accelerate.h",
-    "neteq/audio_classifier.cc",
-    "neteq/audio_classifier.h",
     "neteq/audio_decoder_impl.cc",
     "neteq/audio_decoder_impl.h",
     "neteq/audio_multi_vector.cc",
     "neteq/audio_multi_vector.h",
     "neteq/audio_vector.cc",
     "neteq/audio_vector.h",
     "neteq/background_noise.cc",
     "neteq/background_noise.h",

deleted file mode 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.cc
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- *  Copyright (c) 2014 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 "webrtc/modules/audio_coding/neteq/audio_classifier.h"
-
-#include <assert.h>
-#include <string.h>
-
-namespace webrtc {
-
-static const int kDefaultSampleRateHz = 48000;
-static const int kDefaultFrameRateHz = 50;
-static const int kDefaultFrameSizeSamples =
-    kDefaultSampleRateHz / kDefaultFrameRateHz;
-static const float kDefaultThreshold = 0.5f;
-
-AudioClassifier::AudioClassifier()
-    : analysis_info_(),
-      is_music_(false),
-      music_probability_(0),
-      // This actually assigns the pointer to a static constant struct
-      // rather than creates a struct and |celt_mode_| does not need
-      // to be deleted.
-      celt_mode_(opus_custom_mode_create(kDefaultSampleRateHz,
-                                         kDefaultFrameSizeSamples,
-                                         NULL)) {
-  assert(celt_mode_);
-  tonality_analysis_init(&analysis_state_);
-}
-
-AudioClassifier::~AudioClassifier() {}
-
-bool AudioClassifier::Analysis(const int16_t* input,
-                               int input_length,
-                               int channels) {
-  // Must be 20 ms frames at 48 kHz sampling.
-  assert((input_length / channels) == kDefaultFrameSizeSamples);
-
-  // Only mono or stereo are allowed.
-  assert(channels == 1 || channels == 2);
-
-  // Call Opus' classifier, defined in
-  // "third_party/opus/src/src/analysis.h", with lsb_depth = 16.
-  // Also uses a down-mixing function downmix_int, defined in
-  // "third_party/opus/src/src/opus_private.h", with
-  // constants c1 = 0, and c2 = -2.
-  run_analysis(&analysis_state_,
-               celt_mode_,
-               input,
-               kDefaultFrameSizeSamples,
-               kDefaultFrameSizeSamples,
-               0,
-               -2,
-               channels,
-               kDefaultSampleRateHz,
-               16,
-               downmix_int,
-               &analysis_info_);
-  music_probability_ = analysis_info_.music_prob;
-  is_music_ = music_probability_ > kDefaultThreshold;
-  return is_music_;
-}
-
-bool AudioClassifier::is_music() const {
-  return is_music_;
-}
-
-}  // namespace webrtc

deleted file mode 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- *  Copyright (c) 2014 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.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_CLASSIFIER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_CLASSIFIER_H_
-
-extern "C" {
-#include "celt.h"
-#include "analysis.h"
-#include "opus_private.h"
-}
-
-#include "webrtc/base/scoped_ptr.h"
-#include "webrtc/typedefs.h"
-
-namespace webrtc {
-
-// This class provides a speech/music classification and is a wrapper over the
-// Opus classifier. It currently only supports 48 kHz mono or stereo with a
-// frame size of 20 ms.
-
-class AudioClassifier {
- public:
-  AudioClassifier();
-  virtual ~AudioClassifier();
-
-  // Classifies one frame of audio data in input,
-  // input_length   : must be channels * 960;
-  // channels       : must be 1 (mono) or 2 (stereo).
-  bool Analysis(const int16_t* input, int input_length, int channels);
-
-  // Gets the current classification : true = music, false = speech.
-  virtual bool is_music() const;
-
-  // Gets the current music probability.
-  float music_probability() const { return music_probability_; }
-
- private:
-  AnalysisInfo analysis_info_;
-  bool is_music_;
-  float music_probability_;
-  const CELTMode* celt_mode_;
-  TonalityAnalysisState analysis_state_;
-};
-
-}  // namespace webrtc
-
-#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_AUDIO_CLASSIFIER_H_

deleted file mode 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/audio_classifier_unittest.cc
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- *  Copyright (c) 2014 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 "webrtc/modules/audio_coding/neteq/audio_classifier.h"
-
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <string>
-
-#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/test/testsupport/fileutils.h"
-
-namespace webrtc {
-
-static const size_t kFrameSize = 960;
-
-TEST(AudioClassifierTest, AllZeroInput) {
-  int16_t in_mono[kFrameSize] = {0};
-
-  // Test all-zero vectors and let the classifier converge from its default
-  // to the expected value.
-  AudioClassifier zero_classifier;
-  for (int i = 0; i < 100; ++i) {
-    zero_classifier.Analysis(in_mono, kFrameSize, 1);
-  }
-  EXPECT_TRUE(zero_classifier.is_music());
-}
-
-void RunAnalysisTest(const std::string& audio_filename,
-                     const std::string& data_filename,
-                     size_t channels) {
-  AudioClassifier classifier;
-  rtc::scoped_ptr<int16_t[]> in(new int16_t[channels * kFrameSize]);
-  bool is_music_ref;
-
-  FILE* audio_file = fopen(audio_filename.c_str(), "rb");
-  ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename
-                                  << std::endl;
-  FILE* data_file = fopen(data_filename.c_str(), "rb");
-  ASSERT_TRUE(audio_file != NULL) << "Failed to open file " << audio_filename
-                                  << std::endl;
-  while (fread(in.get(), sizeof(int16_t), channels * kFrameSize, audio_file) ==
-         channels * kFrameSize) {
-    bool is_music =
-        classifier.Analysis(in.get(), channels * kFrameSize, channels);
-    EXPECT_EQ(is_music, classifier.is_music());
-    ASSERT_EQ(1u, fread(&is_music_ref, sizeof(is_music_ref), 1, data_file));
-    EXPECT_EQ(is_music_ref, is_music);
-  }
-  fclose(audio_file);
-  fclose(data_file);
-}
-
-TEST(AudioClassifierTest, DoAnalysisMono) {
-  RunAnalysisTest(test::ResourcePath("short_mixed_mono_48", "pcm"),
-                  test::ResourcePath("short_mixed_mono_48", "dat"),
-                  1);
-}
-
-TEST(AudioClassifierTest, DoAnalysisStereo) {
-  RunAnalysisTest(test::ResourcePath("short_mixed_stereo_48", "pcm"),
-                  test::ResourcePath("short_mixed_stereo_48", "dat"),
-                  2);
-}
-
-}  // namespace webrtc

--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq.gypi
@@ -91,18 +91,16 @@
             ],
           },
         }],
       ],
       'sources': [
         'include/neteq.h',
         'accelerate.cc',
         'accelerate.h',
-        'audio_classifier.cc',
-        'audio_classifier.h',
         'audio_decoder_impl.cc',
         'audio_decoder_impl.h',
         'audio_multi_vector.cc',
         'audio_multi_vector.h',
         'audio_vector.cc',
         'audio_vector.h',
         'background_noise.cc',
         'background_noise.h',

--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
+++ b/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/neteq_tests.gypi
@@ -173,28 +173,16 @@
         'neteq_test_tools',
       ],
       'sources': [
         'test/rtp_to_text.cc',
       ],
     },
 
     {
-      'target_name': 'audio_classifier_test',
-      'type': 'executable',
-      'dependencies': [
-        'neteq',
-        'webrtc_opus',
-      ],
-      'sources': [
-        'test/audio_classifier_test.cc',
-      ],
-    },
-
-    {
       'target_name': 'neteq_test_support',
       'type': 'static_library',
       'dependencies': [
         '<(DEPTH)/testing/gtest.gyp:gtest',
         '<(DEPTH)/third_party/gflags/gflags.gyp:gflags',
         'neteq',
         'neteq_unittest_tools',
         'pcm16b',

deleted file mode 100644
--- a/media/webrtc/trunk/webrtc/modules/audio_coding/neteq/test/audio_classifier_test.cc
+++ /dev/null
@@ -1,105 +0,0 @@
-/*
- *  Copyright (c) 2014 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 "webrtc/modules/audio_coding/neteq/audio_classifier.h"
-
-#include <math.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include <string>
-#include <iostream>
-
-#include "webrtc/base/scoped_ptr.h"
-
-int main(int argc, char* argv[]) {
-  if (argc != 5) {
-    std::cout << "Usage: " << argv[0] <<
-        " channels output_type <input file name> <output file name> "
-        << std::endl << std::endl;
-    std::cout << "Where channels can be 1 (mono) or 2 (interleaved stereo),";
-    std::cout << " outputs can be 1 (classification (boolean)) or 2";
-    std::cout << " (classification and music probability (float)),"
-        << std::endl;
-    std::cout << "and the sampling frequency is assumed to be 48 kHz."
-        << std::endl;
-    return -1;
-  }
-
-  const int kFrameSizeSamples = 960;
-  int channels = atoi(argv[1]);
-  if (channels < 1 || channels > 2) {
-    std::cout << "Disallowed number of channels  " << channels << std::endl;
-    return -1;
-  }
-
-  int outputs = atoi(argv[2]);
-  if (outputs < 1 || outputs > 2) {
-    std::cout << "Disallowed number of outputs  " << outputs << std::endl;
-    return -1;
-  }
-
-  const int data_size = channels * kFrameSizeSamples;
-  rtc::scoped_ptr<int16_t[]> in(new int16_t[data_size]);
-
-  std::string input_filename = argv[3];
-  std::string output_filename = argv[4];
-
-  std::cout << "Input file: " << input_filename << std::endl;
-  std::cout << "Output file: " << output_filename << std::endl;
-
-  FILE* in_file = fopen(input_filename.c_str(), "rb");
-  if (!in_file) {
-    std::cout << "Cannot open input file " << input_filename << std::endl;
-    return -1;
-  }
-
-  FILE* out_file = fopen(output_filename.c_str(), "wb");
-  if (!out_file) {
-    std::cout << "Cannot open output file " << output_filename << std::endl;
-    return -1;
-  }
-
-  webrtc::AudioClassifier classifier;
-  int frame_counter = 0;
-  int music_counter = 0;
-  while (fread(in.get(), sizeof(*in.get()),
-               data_size, in_file) == (size_t) data_size) {
-    bool is_music = classifier.Analysis(in.get(), data_size, channels);
-    if (!fwrite(&is_music, sizeof(is_music), 1, out_file)) {
-       std::cout << "Error writing." << std::endl;
-       return -1;
-    }
-    if (is_music) {
-      music_counter++;
-    }
-    std::cout << "frame " << frame_counter << " decision " << is_music;
-    if (outputs == 2) {
-      float music_prob = classifier.music_probability();
-      if (!fwrite(&music_prob, sizeof(music_prob), 1, out_file)) {
-        std::cout << "Error writing." << std::endl;
-        return -1;
-      }
-      std::cout << " music prob " << music_prob;
-    }
-    std::cout << std::endl;
-    frame_counter++;
-  }
-  std::cout << frame_counter << " frames processed." << std::endl;
-  if (frame_counter > 0) {
-    float music_percentage = music_counter / static_cast<float>(frame_counter);
-    std::cout <<  music_percentage <<  " percent music." << std::endl;
-  }
-
-  fclose(in_file);
-  fclose(out_file);
-  return 0;
-}

--- a/media/webrtc/trunk/webrtc/modules/modules.gyp
+++ b/media/webrtc/trunk/webrtc/modules/modules.gyp
@@ -169,17 +169,16 @@
                 'audio_coding/codecs/isac/fix/source/lpc_masking_model_unittest.cc',
                 'audio_coding/codecs/isac/fix/source/transform_unittest.cc',
                 'audio_coding/codecs/isac/main/source/audio_encoder_isac_unittest.cc',
                 'audio_coding/codecs/isac/main/source/isac_unittest.cc',
                 'audio_coding/codecs/isac/unittest.cc',
                 'audio_coding/codecs/opus/audio_encoder_opus_unittest.cc',
                 'audio_coding/codecs/opus/opus_unittest.cc',
                 'audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc',
-                'audio_coding/neteq/audio_classifier_unittest.cc',
                 'audio_coding/neteq/audio_multi_vector_unittest.cc',
                 'audio_coding/neteq/audio_vector_unittest.cc',
                 'audio_coding/neteq/background_noise_unittest.cc',
                 'audio_coding/neteq/buffer_level_filter_unittest.cc',
                 'audio_coding/neteq/comfort_noise_unittest.cc',
                 'audio_coding/neteq/decision_logic_unittest.cc',
                 'audio_coding/neteq/decoder_database_unittest.cc',
                 'audio_coding/neteq/delay_manager_unittest.cc',