Add stub draft of audio generator to APM

This provides the empty shell of an AudioGenerator class.
It is intended to be used for debugging purposes and can be inserted
into the APM much like an AecDump. It allows for playing out diagnostic
audio unaffected by codecs and network jitter, while still capturing
API interaction like in a normal call.

NOTRY=True

Bug: webrtc:8882
Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7
Reviewed-on: https://webrtc-review.googlesource.com/53300
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22282}

modules/audio_processing/BUILD.gn

@@ -140,6 +140,7 @@ rtc_static_library("audio_processing") {
     ":aec_dump_interface",
     ":apm_logging",
     ":audio_frame_view",
+    ":audio_generator_interface",
     ":audio_processing_c",
     ":audio_processing_statistics",
     "..:module_api",
@@ -235,6 +236,46 @@ rtc_source_set("aec_dump_interface") {
   ]
 }
 
+rtc_source_set("audio_generator_interface") {
+  visibility = [ "*" ]
+  sources = [
+    "include/audio_generator.h",
+  ]
+  deps = [
+    ":audio_frame_view",
+    "../../rtc_base:rtc_base_approved",
+    "../../system_wrappers:system_wrappers",
+  ]
+}
+
+rtc_source_set("audio_generator_factory") {
+  visibility = [ "*" ]
+  sources = [
+    "include/audio_generator_factory.cc",
+    "include/audio_generator_factory.h",
+  ]
+  deps = [
+    ":audio_generator_interface",
+    ":file_audio_generator",
+    "../../common_audio:common_audio",
+    "../../rtc_base:rtc_base_approved",
+    "../../system_wrappers:system_wrappers",
+  ]
+}
+
+rtc_source_set("file_audio_generator") {
+  sources = [
+    "audio_generator/file_audio_generator.cc",
+    "audio_generator/file_audio_generator.h",
+  ]
+  deps = [
+    ":audio_generator_interface",
+    "../../common_audio:common_audio",
+    "../../rtc_base:rtc_base_approved",
+    "../../system_wrappers:system_wrappers",
+  ]
+}
+
 rtc_source_set("audio_processing_c") {
   visibility = [ ":*" ]  # Only targets in this file can depend on this.
   sources = [
@@ -501,6 +542,7 @@ if (rtc_include_tests) {
       ":audio_frame_view",
       ":audio_processing",
       ":audioproc_test_utils",
+      ":file_audio_generator_unittests",
       ":mocks",
       "..:module_api",
       "../..:typedefs",
@@ -617,6 +659,22 @@ if (rtc_include_tests) {
     }
   }
 
+  rtc_source_set("file_audio_generator_unittests") {
+    testonly = true
+
+    sources = [
+      "audio_generator/file_audio_generator_unittest.cc",
+    ]
+
+    deps = [
+      ":audio_generator_factory",
+      ":audio_processing",
+      ":file_audio_generator",
+      "../../rtc_base:rtc_base_approved",
+      "../../test:test_support",
+    ]
+  }
+
   rtc_source_set("analog_mic_simulation") {
     sources = [
       "test/fake_recording_device.cc",

modules/audio_processing/audio_generator/file_audio_generator.cc

@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2018 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 "modules/audio_processing/audio_generator/file_audio_generator.h"
+
+namespace webrtc {
+
+FileAudioGenerator::FileAudioGenerator(
+    std::unique_ptr<WavReader> input_audio_file) {
+  // TODO(bugs.webrtc.org/8882) Stub.
+  // Read audio from file into internal buffer.
+}
+
+FileAudioGenerator::~FileAudioGenerator() = default;
+
+void FileAudioGenerator::FillFrame(AudioFrameView<float> audio) {
+  // TODO(bugs.webrtc.org/8882) Stub.
+  // Fill |audio| from internal buffer.
+}
+
+size_t FileAudioGenerator::NumChannels() {
+  return num_channels_;
+}
+
+size_t FileAudioGenerator::SampleRateHz() {
+  return sample_rate_hz_;
+}
+
+}  // namespace webrtc

modules/audio_processing/audio_generator/file_audio_generator.h

@@ -0,0 +1,48 @@
+/*
+ *  Copyright (c) 2018 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 MODULES_AUDIO_PROCESSING_AUDIO_GENERATOR_FILE_AUDIO_GENERATOR_H_
+#define MODULES_AUDIO_PROCESSING_AUDIO_GENERATOR_FILE_AUDIO_GENERATOR_H_
+
+#include <memory>
+
+#include "common_audio/wav_file.h"
+#include "modules/audio_processing/include/audio_generator.h"
+#include "rtc_base/constructormagic.h"
+
+namespace webrtc {
+
+// Provides looping audio from a file. The file is read in its entirety on
+// construction and then closed. This class wraps a webrtc::WavReader, and is
+// hence unsuitable for non-diagnostic code.
+class FileAudioGenerator : public AudioGenerator {
+ public:
+  // Reads the playout audio from a given WAV file.
+  explicit FileAudioGenerator(std::unique_ptr<WavReader> input_audio_file);
+
+  ~FileAudioGenerator() override;
+
+  // Fill |audio| with audio from a file.
+  void FillFrame(AudioFrameView<float> audio) override;
+
+  size_t NumChannels() override;
+
+  size_t SampleRateHz() override;
+
+ private:
+  size_t num_channels_;
+  size_t sample_rate_hz_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(FileAudioGenerator);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AUDIO_GENERATOR_FILE_AUDIO_GENERATOR_H_

modules/audio_processing/audio_generator/file_audio_generator_unittest.cc

@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2017 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>
+
+#include "modules/audio_processing/include/audio_generator_factory.h"
+#include "modules/audio_processing/include/audio_processing.h"
+#include "test/gtest.h"
+#include "test/testsupport/fileutils.h"
+
+namespace webrtc {
+namespace test {
+
+// TODO(bugs.webrtc.org/8882) Stub.
+// Add unit tests for both file audio and generated audio.
+
+TEST(FileAudioGeneratorTest, CreationDeletion) {
+  const std::string audio_filename =
+      test::ResourcePath("voice_engine/audio_tiny48", "wav");
+  auto audio_generator = AudioGeneratorFactory::Create(audio_filename);
+}
+
+}  // namespace test
+}  // namespace webrtc

modules/audio_processing/audio_processing_impl.cc

@@ -1588,6 +1588,17 @@ void AudioProcessingImpl::DetachAecDump() {
   }
 }
 
+void AudioProcessingImpl::AttachPlayoutAudioGenerator(
+    std::unique_ptr<AudioGenerator> audio_generator) {
+  // TODO(bugs.webrtc.org/8882) Stub.
+  // Reset internal audio generator with audio_generator.
+}
+
+void AudioProcessingImpl::DetachPlayoutAudioGenerator() {
+  // TODO(bugs.webrtc.org/8882) Stub.
+  // Delete audio generator, if one is attached.
+}
+
 AudioProcessing::AudioProcessingStatistics::AudioProcessingStatistics() {
   residual_echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);
   echo_return_loss.Set(-100.0f, -100.0f, -100.0f, -100.0f);

modules/audio_processing/audio_processing_impl.h

@@ -62,6 +62,9 @@ class AudioProcessingImpl : public AudioProcessing {
   void UpdateHistogramsOnCallEnd() override;
   void AttachAecDump(std::unique_ptr<AecDump> aec_dump) override;
   void DetachAecDump() override;
+  void AttachPlayoutAudioGenerator(
+      std::unique_ptr<AudioGenerator> audio_generator) override;
+  void DetachPlayoutAudioGenerator() override;
 
   // Capture-side exclusive methods possibly running APM in a
   // multi-threaded manner. Acquire the capture lock.

modules/audio_processing/include/audio_generator.h

@@ -0,0 +1,36 @@
+/*
+ *  Copyright (c) 2018 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 MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_H_
+
+#include "modules/audio_processing/include/audio_frame_view.h"
+
+namespace webrtc {
+// This class is used as input sink for the APM, for diagnostic purposes.
+// Generates an infinite audio signal, [-1, 1] floating point values, in frames
+// of fixed channel count and sample rate.
+class AudioGenerator {
+ public:
+  virtual ~AudioGenerator() {}
+
+  // Fill |audio| with the next samples of the audio signal.
+  virtual void FillFrame(AudioFrameView<float> audio) = 0;
+
+  // Return the number of channels output by the AudioGenerator.
+  virtual size_t NumChannels() = 0;
+
+  // Return the sample rate output by the AudioGenerator.
+  virtual size_t SampleRateHz() = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_H_

modules/audio_processing/include/audio_generator_factory.cc

@@ -0,0 +1,25 @@
+/*
+ *  Copyright (c) 2018 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 "modules/audio_processing/include/audio_generator_factory.h"
+
+#include "common_audio/wav_file.h"
+#include "modules/audio_processing/audio_generator/file_audio_generator.h"
+#include "rtc_base/ptr_util.h"
+
+namespace webrtc {
+
+std::unique_ptr<AudioGenerator> AudioGeneratorFactory::Create(
+    const std::string& file_name) {
+  std::unique_ptr<WavReader> input_audio_file(new WavReader(file_name));
+  return rtc::MakeUnique<FileAudioGenerator>(std::move(input_audio_file));
+}
+
+}  // namespace webrtc

modules/audio_processing/include/audio_generator_factory.h

@@ -0,0 +1,31 @@
+/*
+ *  Copyright (c) 2017 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 MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_FACTORY_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_FACTORY_H_
+
+#include <memory>
+#include <string>
+#include <utility>
+
+#include "modules/audio_processing/include/audio_generator.h"
+
+namespace webrtc {
+
+class AudioGeneratorFactory {
+ public:
+  // Creates an AudioGenerator that reads the playout audio from a given 16-bit
+  // int-encoded WAV file.
+  static std::unique_ptr<AudioGenerator> Create(const std::string& file_name);
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_GENERATOR_FACTORY_H_

modules/audio_processing/include/audio_processing.h

@@ -26,6 +26,7 @@
 #include "api/audio/echo_control.h"
 #include "api/optional.h"
 #include "modules/audio_processing/beamformer/array_util.h"
+#include "modules/audio_processing/include/audio_generator.h"
 #include "modules/audio_processing/include/audio_processing_statistics.h"
 #include "modules/audio_processing/include/config.h"
 #include "rtc_base/arraysize.h"
@@ -480,6 +481,16 @@ class AudioProcessing : public rtc::RefCountInterface {
   // all pending logging tasks are completed.
   virtual void DetachAecDump() = 0;
 
+  // Attaches provided webrtc::AudioGenerator for modifying playout audio.
+  // Calling this method when another AudioGenerator is attached replaces the
+  // active AudioGenerator with a new one.
+  virtual void AttachPlayoutAudioGenerator(
+      std::unique_ptr<AudioGenerator> audio_generator) = 0;
+
+  // If no AudioGenerator is attached, this has no effect. If an AecDump is
+  // attached, its destructor is called.
+  virtual void DetachPlayoutAudioGenerator() = 0;
+
   // Use to send UMA histograms at end of a call. Note that all histogram
   // specific member variables are reset.
   virtual void UpdateHistogramsOnCallEnd() = 0;

modules/audio_processing/include/mock_audio_processing.h

@@ -198,6 +198,10 @@ class MockAudioProcessing : public testing::NiceMock<AudioProcessing> {
   virtual void AttachAecDump(std::unique_ptr<AecDump> aec_dump) {}
   MOCK_METHOD0(DetachAecDump, void());
 
+  virtual void AttachPlayoutAudioGenerator(
+      std::unique_ptr<AudioGenerator> audio_generator) {}
+  MOCK_METHOD0(DetachPlayoutAudioGenerator, void());
+
   MOCK_METHOD0(UpdateHistogramsOnCallEnd, void());
   MOCK_CONST_METHOD0(GetStatistics, AudioProcessingStatistics());
   MOCK_CONST_METHOD1(GetStatistics, AudioProcessingStats(bool));