content/renderer/media/media_stream_audio_processor

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This source file includes following definitions.
ReadDataFromSpeechFile
ProcessDataAndVerifyFormat
VerifyDefaultComponents
TEST_F
TEST_F
TEST_F
TEST_F
// 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/public/common/media_stream_request.h"
#include "content/renderer/media/media_stream_audio_processor.h"
#include "content/renderer/media/mock_media_constraint_factory.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/WebKit/public/platform/WebMediaConstraints.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(base::GetFileSize(file, &data_file_size64));
  EXPECT_EQ(length, base::ReadFile(file, data, length));
  DCHECK(data_file_size64 > length);
}

}  // namespace

class MediaStreamAudioProcessorTest : public ::testing::Test {
 public:
  MediaStreamAudioProcessorTest()
      : params_(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                media::CHANNEL_LAYOUT_STEREO, 48000, 16, 512) {
  }

 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.
      webrtc::AudioProcessing* ap = audio_processor->audio_processing_.get();
#if defined(OS_ANDROID) || defined(OS_IOS)
      const bool is_aec_enabled = ap && ap->echo_control_mobile()->is_enabled();
      // AEC should be turned off for mobiles.
      DCHECK(!ap || !ap->echo_cancellation()->is_enabled());
#else
      const bool is_aec_enabled = ap && ap->echo_cancellation()->is_enabled();
#endif
      if (is_aec_enabled) {
        audio_processor->OnPlayoutData(data_bus.get(), params_.sample_rate(),
                                       10);
      }

      int16* output = NULL;
      int new_volume = 0;
      while(audio_processor->ProcessAndConsumeData(
          base::TimeDelta::FromMilliseconds(10), 255, false, &new_volume,
          &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();
    }
  }

  void VerifyDefaultComponents(MediaStreamAudioProcessor* audio_processor) {
    webrtc::AudioProcessing* audio_processing =
        audio_processor->audio_processing_.get();
#if defined(OS_ANDROID)
    EXPECT_TRUE(audio_processing->echo_control_mobile()->is_enabled());
    EXPECT_TRUE(audio_processing->echo_control_mobile()->routing_mode() ==
        webrtc::EchoControlMobile::kSpeakerphone);
    EXPECT_FALSE(audio_processing->echo_cancellation()->is_enabled());
#elif !defined(OS_IOS)
    EXPECT_TRUE(audio_processing->echo_cancellation()->is_enabled());
    EXPECT_TRUE(audio_processing->echo_cancellation()->suppression_level() ==
        webrtc::EchoCancellation::kHighSuppression);
    EXPECT_TRUE(audio_processing->echo_cancellation()->are_metrics_enabled());
    EXPECT_TRUE(
        audio_processing->echo_cancellation()->is_delay_logging_enabled());
#endif

    EXPECT_TRUE(audio_processing->noise_suppression()->is_enabled());
    EXPECT_TRUE(audio_processing->noise_suppression()->level() ==
        webrtc::NoiseSuppression::kHigh);
    EXPECT_TRUE(audio_processing->high_pass_filter()->is_enabled());
    EXPECT_TRUE(audio_processing->gain_control()->is_enabled());
#if defined(OS_ANDROID) || defined(OS_IOS)
    EXPECT_TRUE(audio_processing->gain_control()->mode() ==
        webrtc::GainControl::kFixedDigital);
    EXPECT_FALSE(audio_processing->voice_detection()->is_enabled());
#else
    EXPECT_TRUE(audio_processing->gain_control()->mode() ==
        webrtc::GainControl::kAdaptiveAnalog);
    EXPECT_TRUE(audio_processing->voice_detection()->is_enabled());
    EXPECT_TRUE(audio_processing->voice_detection()->likelihood() ==
        webrtc::VoiceDetection::kVeryLowLikelihood);
#endif
  }

  media::AudioParameters params_;
};

TEST_F(MediaStreamAudioProcessorTest, WithoutAudioProcessing) {
  // Setup the audio processor without enabling the flag.
  blink::WebMediaConstraints constraints;
  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
      new WebRtcAudioDeviceImpl());
  scoped_refptr<MediaStreamAudioProcessor> audio_processor(
      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
          constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE,
          webrtc_audio_device.get()));
  EXPECT_FALSE(audio_processor->has_audio_processing());
  audio_processor->OnCaptureFormatChanged(params_);

  ProcessDataAndVerifyFormat(audio_processor,
                             params_.sample_rate(),
                             params_.channels(),
                             params_.sample_rate() / 100);
  // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
  // |audio_processor|.
  audio_processor = NULL;
}

TEST_F(MediaStreamAudioProcessorTest, WithAudioProcessing) {
  // Setup the audio processor with the flag enabled.
  CommandLine::ForCurrentProcess()->AppendSwitch(
      switches::kEnableAudioTrackProcessing);
  blink::WebMediaConstraints constraints;
  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
      new WebRtcAudioDeviceImpl());
  scoped_refptr<MediaStreamAudioProcessor> audio_processor(
      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
          constraints, 0, MEDIA_DEVICE_AUDIO_CAPTURE,
          webrtc_audio_device.get()));
  EXPECT_TRUE(audio_processor->has_audio_processing());
  audio_processor->OnCaptureFormatChanged(params_);
  VerifyDefaultComponents(audio_processor);

  ProcessDataAndVerifyFormat(audio_processor,
                             kAudioProcessingSampleRate,
                             kAudioProcessingNumberOfChannel,
                             kAudioProcessingSampleRate / 100);
  // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
  // |audio_processor|.
  audio_processor = NULL;
}

TEST_F(MediaStreamAudioProcessorTest, VerifyTabCaptureWithoutAudioProcessing) {
  // Setup the audio processor with enabling the flag.
  CommandLine::ForCurrentProcess()->AppendSwitch(
      switches::kEnableAudioTrackProcessing);
  blink::WebMediaConstraints constraints;
  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
      new WebRtcAudioDeviceImpl());
  // Create MediaStreamAudioProcessor instance for MEDIA_TAB_AUDIO_CAPTURE type.
  scoped_refptr<MediaStreamAudioProcessor> audio_processor(
      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
          constraints, 0, MEDIA_TAB_AUDIO_CAPTURE,
          webrtc_audio_device.get()));
  EXPECT_FALSE(audio_processor->has_audio_processing());
  audio_processor->OnCaptureFormatChanged(params_);

  ProcessDataAndVerifyFormat(audio_processor,
                             params_.sample_rate(),
                             params_.channels(),
                             params_.sample_rate() / 100);

  // Create MediaStreamAudioProcessor instance for MEDIA_LOOPBACK_AUDIO_CAPTURE.
  audio_processor =
      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
          constraints, 0, MEDIA_LOOPBACK_AUDIO_CAPTURE,
          webrtc_audio_device.get());
  EXPECT_FALSE(audio_processor->has_audio_processing());

  // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
  // |audio_processor|.
  audio_processor = NULL;
}

TEST_F(MediaStreamAudioProcessorTest, TurnOffDefaultConstraints) {
  // Setup the audio processor with enabling the flag.
  CommandLine::ForCurrentProcess()->AppendSwitch(
      switches::kEnableAudioTrackProcessing);

  // Turn off the default constraints and pass it to MediaStreamAudioProcessor.
  MockMediaConstraintFactory constraint_factory;
  constraint_factory.DisableDefaultAudioConstraints();
  scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
      new WebRtcAudioDeviceImpl());
  scoped_refptr<MediaStreamAudioProcessor> audio_processor(
      new talk_base::RefCountedObject<MediaStreamAudioProcessor>(
          constraint_factory.CreateWebMediaConstraints(), 0,
          MEDIA_DEVICE_AUDIO_CAPTURE, webrtc_audio_device.get()));
  EXPECT_FALSE(audio_processor->has_audio_processing());
  audio_processor->OnCaptureFormatChanged(params_);

  ProcessDataAndVerifyFormat(audio_processor,
                             params_.sample_rate(),
                             params_.channels(),
                             params_.sample_rate() / 100);
  // Set |audio_processor| to NULL to make sure |webrtc_audio_device| outlives
  // |audio_processor|.
  audio_processor = NULL;
}

}  // namespace content
/* [<][>][^][v][top][bottom][index][help] */
root/content/renderer/media/media_stream_audio_processor_unittest.cc

DEFINITIONS
