Revert 237311 "Added an "enable-audio-processor" flag and WebRtc..."

trunk/src/content/renderer/media/media_stream_audio_processor_unittest.cc

Index: trunk/src/content/renderer/media/media_stream_audio_processor_unittest.cc
===================================================================
--- trunk/src/content/renderer/media/media_stream_audio_processor_unittest.cc	(revision 237333)
+++ trunk/src/content/renderer/media/media_stream_audio_processor_unittest.cc	(working copy)
@@ -1,166 +0,0 @@
-// Copyright 2013 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file.
-
-#include "base/command_line.h"
-#include "base/file_util.h"
-#include "base/files/file_path.h"
-#include "base/logging.h"
-#include "base/path_service.h"
-#include "base/time/time.h"
-#include "content/public/common/content_switches.h"
-#include "content/renderer/media/media_stream_audio_processor.h"
-#include "content/renderer/media/rtc_media_constraints.h"
-#include "media/audio/audio_parameters.h"
-#include "media/base/audio_bus.h"
-#include "testing/gmock/include/gmock/gmock.h"
-#include "testing/gtest/include/gtest/gtest.h"
-#include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
-
-using ::testing::_;
-using ::testing::AnyNumber;
-using ::testing::AtLeast;
-using ::testing::Return;
-
-namespace content {
-
-namespace {
-
-#if defined(ANDROID)
-const int kAudioProcessingSampleRate = 16000;
-#else
-const int kAudioProcessingSampleRate = 32000;
-#endif
-const int kAudioProcessingNumberOfChannel = 1;
-
-// The number of packers used for testing.
-const int kNumberOfPacketsForTest = 100;
-
-void ReadDataFromSpeechFile(char* data, int length) {
-  base::FilePath file;
-  CHECK(PathService::Get(base::DIR_SOURCE_ROOT, &file));
-  file = file.Append(FILE_PATH_LITERAL("media"))
-             .Append(FILE_PATH_LITERAL("test"))
-             .Append(FILE_PATH_LITERAL("data"))
-             .Append(FILE_PATH_LITERAL("speech_16b_stereo_48kHz.raw"));
-  DCHECK(base::PathExists(file));
-  int64 data_file_size64 = 0;
-  DCHECK(file_util::GetFileSize(file, &data_file_size64));
-  EXPECT_EQ(length, file_util::ReadFile(file, data, length));
-  DCHECK(data_file_size64 > length);
-}
-
-void ApplyFixedAudioConstraints(RTCMediaConstraints* constraints) {
-  // Constant constraint keys which enables default audio constraints on
-  // mediastreams with audio.
-  struct {
-    const char* key;
-    const char* value;
-  } static const kDefaultAudioConstraints[] = {
-    { webrtc::MediaConstraintsInterface::kEchoCancellation,
-      webrtc::MediaConstraintsInterface::kValueTrue },
-  #if defined(OS_CHROMEOS) || defined(OS_MACOSX)
-    // Enable the extended filter mode AEC on platforms with known echo issues.
-    { webrtc::MediaConstraintsInterface::kExperimentalEchoCancellation,
-      webrtc::MediaConstraintsInterface::kValueTrue },
-  #endif
-    { webrtc::MediaConstraintsInterface::kAutoGainControl,
-      webrtc::MediaConstraintsInterface::kValueTrue },
-    { webrtc::MediaConstraintsInterface::kExperimentalAutoGainControl,
-      webrtc::MediaConstraintsInterface::kValueTrue },
-    { webrtc::MediaConstraintsInterface::kNoiseSuppression,
-      webrtc::MediaConstraintsInterface::kValueTrue },
-    { webrtc::MediaConstraintsInterface::kHighpassFilter,
-      webrtc::MediaConstraintsInterface::kValueTrue },
-  };
-
-  for (size_t i = 0; i < ARRAYSIZE_UNSAFE(kDefaultAudioConstraints); ++i) {
-    constraints->AddMandatory(kDefaultAudioConstraints[i].key,
-                              kDefaultAudioConstraints[i].value, false);
-  }
-}
-
-}  // namespace
-
-class MediaStreamAudioProcessorTest : public ::testing::Test {
- public:
-  MediaStreamAudioProcessorTest()
-      : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
-                media::CHANNEL_LAYOUT_STEREO, 48000, 16, 512) {
-    CommandLine::ForCurrentProcess()->AppendSwitch(
-        switches::kEnableAudioTrackProcessing);
-  }
-
- protected:
-  // Helper method to save duplicated code.
-  void ProcessDataAndVerifyFormat(MediaStreamAudioProcessor* audio_processor,
-                                  int expected_output_sample_rate,
-                                  int expected_output_channels,
-                                  int expected_output_buffer_size) {
-    // Read the audio data from a file.
-    const int packet_size =
-        params_.frames_per_buffer() * 2 * params_.channels();
-    const size_t length = packet_size * kNumberOfPacketsForTest;
-    scoped_ptr<char[]> capture_data(new char[length]);
-    ReadDataFromSpeechFile(capture_data.get(), length);
-    const int16* data_ptr = reinterpret_cast<const int16*>(capture_data.get());
-    scoped_ptr<media::AudioBus> data_bus = media::AudioBus::Create(
-        params_.channels(), params_.frames_per_buffer());
-    for (int i = 0; i < kNumberOfPacketsForTest; ++i) {
-      data_bus->FromInterleaved(data_ptr, data_bus->frames(), 2);
-      audio_processor->PushCaptureData(data_bus.get());
-
-      // |audio_processor| does nothing when the audio processing is off in
-      // the processor.
-      audio_processor->PushRenderData(
-          data_ptr,
-          params_.sample_rate(), params_.channels(),
-          params_.frames_per_buffer(), base::TimeDelta::FromMilliseconds(10));
-
-      int16* output = NULL;
-      while(audio_processor->ProcessAndConsumeData(
-          base::TimeDelta::FromMilliseconds(10), 255, false, &output)) {
-        EXPECT_TRUE(output != NULL);
-        EXPECT_EQ(audio_processor->OutputFormat().sample_rate(),
-                  expected_output_sample_rate);
-        EXPECT_EQ(audio_processor->OutputFormat().channels(),
-                  expected_output_channels);
-        EXPECT_EQ(audio_processor->OutputFormat().frames_per_buffer(),
-                  expected_output_buffer_size);
-      }
-
-      data_ptr += params_.frames_per_buffer() * params_.channels();
-    }
-  }
-
-  media::AudioParameters params_;
-};
-
-TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) {
-  // Setup the audio processor with empty constraint.
-  RTCMediaConstraints constraints;
-  MediaStreamAudioProcessor audio_processor(&constraints);
-  audio_processor.SetCaptureFormat(params_);
-  EXPECT_FALSE(audio_processor.has_audio_processing());
-
-  ProcessDataAndVerifyFormat(&audio_processor,
-                             params_.sample_rate(),
-                             params_.channels(),
-                             params_.sample_rate() / 100);
-}
-
-TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) {
-  // Setup the audio processor with default constraint.
-  RTCMediaConstraints constraints;
-  ApplyFixedAudioConstraints(&constraints);
-  MediaStreamAudioProcessor audio_processor(&constraints);
-  audio_processor.SetCaptureFormat(params_);
-  EXPECT_TRUE(audio_processor.has_audio_processing());
-
-  ProcessDataAndVerifyFormat(&audio_processor,
-                             kAudioProcessingSampleRate,
-                             kAudioProcessingNumberOfChannel,
-                             kAudioProcessingSampleRate / 100);
-}
-
-}  // namespace content