MultiEndCall is responsible for analyzing and validating timing information and audiotracks with which a multi-end call can be simulated.
The class creates one WavReaderInterface object for each unique audiotrack and builds the set of speaker names.
Validating if the audiotrack lengths and the timing information are compatible (and hence valid) is not implemented yet.
MultiEndCall is designed using dependency injection. This allows to use mock objects with which we can quickly simulate different timings and track lengths without needing actual wav files.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2761853002
Cr-Commit-Position: refs/heads/master@{#17421}
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn b/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
index 4375c4e..ca5977f 100644
--- a/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
+++ b/webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn
@@ -33,11 +33,21 @@
sources = [
"config.cc",
"config.h",
+ "multiend_call.cc",
+ "multiend_call.h",
"timing.cc",
"timing.h",
+ "wavreader_abstract_factory.h",
+ "wavreader_adaptor.cc",
+ "wavreader_adaptor.h",
+ "wavreader_factory.cc",
+ "wavreader_factory.h",
+ "wavreader_interface.h",
]
deps = [
+ "//webrtc:webrtc_common",
"//webrtc/base:rtc_base_approved",
+ "//webrtc/common_audio",
]
visibility = [ ":*" ] # Only targets in this file can depend on this.
}
@@ -46,11 +56,15 @@
testonly = true
sources = [
"generator_unittest.cc",
+ "mock_wavreader.h",
+ "mock_wavreader_factory.cc",
+ "mock_wavreader_factory.h",
]
deps = [
":lib",
"//testing/gmock",
"//testing/gtest",
+ "//webrtc:webrtc_common",
"//webrtc/test:test_support",
]
}
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc b/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
index 221ad7c..59454d9 100644
--- a/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
+++ b/webrtc/modules/audio_processing/test/conversational_speech/generator_unittest.cc
@@ -8,11 +8,14 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include <cstdio>
+#include <stdio.h>
#include <memory>
#include "webrtc/modules/audio_processing/test/conversational_speech/config.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h"
#include "webrtc/modules/audio_processing/test/conversational_speech/timing.h"
+#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/testsupport/fileutils.h"
@@ -22,7 +25,10 @@
using conversational_speech::LoadTiming;
using conversational_speech::SaveTiming;
+using conversational_speech::MockWavReaderFactory;
+using conversational_speech::MultiEndCall;
using conversational_speech::Turn;
+using conversational_speech::WavReaderAbstractFactory;
const char* const audiotracks_path = "/path/to/audiotracks";
const char* const timing_filepath = "/path/to/timing_file.txt";
@@ -34,7 +40,7 @@
{"A", "a2", 100},
{"B", "b2", -200},
{"A", "a3", 0},
- {"A", "a4", 0},
+ {"A", "a3", 0},
};
const std::size_t kNumberOfTurns = expected_timing.size();
@@ -50,11 +56,6 @@
EXPECT_EQ(output_path, config.output_path());
}
-TEST(ConversationalSpeechTest, ExpectedTimingSize) {
- // Check the expected timing size.
- EXPECT_EQ(kNumberOfTurns, 6u);
-}
-
TEST(ConversationalSpeechTest, TimingSaveLoad) {
// Save test timing.
const std::string temporary_filepath = webrtc::test::TempFilename(
@@ -75,5 +76,21 @@
}
}
+TEST(ConversationalSpeechTest, MultiEndCallCreate) {
+ auto mock_wavreader_factory = std::unique_ptr<MockWavReaderFactory>(
+ new MockWavReaderFactory());
+
+ // There are 5 unique audio tracks to read.
+ EXPECT_CALL(*mock_wavreader_factory, Create(testing::_)).Times(5);
+
+ // Inject the mock wav reader factory.
+ conversational_speech::MultiEndCall multiend_call(
+ expected_timing, audiotracks_path, std::move(mock_wavreader_factory));
+
+ // Test.
+ EXPECT_EQ(2u, multiend_call.speaker_names().size());
+ EXPECT_EQ(5u, multiend_call.audiotrack_readers().size());
+}
+
} // namespace test
} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h b/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
new file mode 100644
index 0000000..83aa938
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader.h
@@ -0,0 +1,52 @@
+/*
+ * 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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_
+
+#include <cstddef>
+#include <string>
+
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+#include "webrtc/test/gmock.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class MockWavReader : public WavReaderInterface {
+ public:
+ MockWavReader(
+ int sample_rate, size_t num_channels, size_t num_samples)
+ : sample_rate_(sample_rate), num_channels_(num_channels),
+ num_samples_(num_samples) {}
+ ~MockWavReader() = default;
+
+ // TOOD(alessiob): use ON_CALL to return random samples.
+ MOCK_METHOD2(ReadFloatSamples, size_t(size_t, float*));
+ MOCK_METHOD2(ReadInt16Samples, size_t(size_t, int16_t*));
+
+ // TOOD(alessiob): use ON_CALL to return properties.
+ MOCK_CONST_METHOD0(sample_rate, int());
+ MOCK_CONST_METHOD0(num_channels, size_t());
+ MOCK_CONST_METHOD0(num_samples, size_t());
+
+ private:
+ const int sample_rate_;
+ const size_t num_channels_;
+ const size_t num_samples_;
+};
+
+} // namespace conversational_speech
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_H_
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc b/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
new file mode 100644
index 0000000..1097639
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.cc
@@ -0,0 +1,23 @@
+/*
+ * 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 "webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+MockWavReaderFactory::MockWavReaderFactory() = default;
+
+MockWavReaderFactory::~MockWavReaderFactory() = default;
+
+} // namespace conversational_speech
+} // namespace test
+} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h b/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h
new file mode 100644
index 0000000..3686d12
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/conversational_speech/mock_wavreader_factory.h
@@ -0,0 +1,42 @@
+/*
+ * 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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_FACTORY_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_FACTORY_H_
+
+#include <memory>
+#include <string>
+
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+#include "webrtc/test/gmock.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class MockWavReaderFactory : public WavReaderAbstractFactory {
+ public:
+ MockWavReaderFactory();
+ // TODO(alessiob): add ctor that gets map string->(sr, #samples, #channels).
+ ~MockWavReaderFactory();
+
+ // TODO(alessiob): use ON_CALL to return MockWavReader with desired params.
+ MOCK_CONST_METHOD1(Create, std::unique_ptr<WavReaderInterface>(
+ const std::string&));
+
+ // TODO(alessiob): add const ref to map (see ctor to add).
+};
+
+} // namespace conversational_speech
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MOCK_WAVREADER_FACTORY_H_
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc b/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
new file mode 100644
index 0000000..f16aa75
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.cc
@@ -0,0 +1,75 @@
+/*
+ * 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 "webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h"
+
+#include <utility>
+
+#include "webrtc/base/pathutils.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+MultiEndCall::MultiEndCall(
+ rtc::ArrayView<const Turn> timing, const std::string& audiotracks_path,
+ std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory)
+ : timing_(timing), audiotracks_path_(audiotracks_path),
+ wavreader_abstract_factory_(std::move(wavreader_abstract_factory)) {
+ FindSpeakerNames();
+ CreateAudioTrackReaders();
+ CheckTiming();
+}
+
+MultiEndCall::~MultiEndCall() = default;
+
+const std::set<std::string>& MultiEndCall::speaker_names() const {
+ return speaker_names_;
+}
+
+const std::map<std::string, std::unique_ptr<WavReaderInterface>>&
+ MultiEndCall::audiotrack_readers() const {
+ return audiotrack_readers_;
+}
+
+void MultiEndCall::FindSpeakerNames() {
+ RTC_DCHECK(speaker_names_.empty());
+ for (const Turn& turn : timing_) {
+ speaker_names_.insert(turn.speaker_name);
+ }
+}
+
+void MultiEndCall::CreateAudioTrackReaders() {
+ RTC_DCHECK(audiotrack_readers_.empty());
+ for (const Turn& turn : timing_) {
+ auto it = audiotrack_readers_.find(turn.audiotrack_file_name);
+ if (it != audiotrack_readers_.end())
+ continue;
+
+ // Instance Pathname to retrieve the full path to the audiotrack file.
+ const rtc::Pathname audiotrack_file_path(
+ audiotracks_path_, turn.audiotrack_file_name);
+
+ // Map the audiotrack file name to a new instance of WavReaderInterface.
+ std::unique_ptr<WavReaderInterface> wavreader =
+ wavreader_abstract_factory_->Create(audiotrack_file_path.pathname());
+ audiotrack_readers_.insert(std::make_pair(
+ turn.audiotrack_file_name, std::move(wavreader)));
+ }
+}
+
+void MultiEndCall::CheckTiming() {
+ // TODO(alessiob): use audiotrack lengths and offset to check whether the
+ // timing is valid.
+}
+
+} // namespace conversational_speech
+} // namespace test
+} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h b/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
new file mode 100644
index 0000000..234cb27
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/conversational_speech/multiend_call.h
@@ -0,0 +1,64 @@
+/*
+ * 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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_
+
+#include <map>
+#include <memory>
+#include <set>
+#include <string>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/timing.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class MultiEndCall {
+ public:
+ MultiEndCall(
+ rtc::ArrayView<const Turn> timing, const std::string& audiotracks_path,
+ std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory);
+ ~MultiEndCall();
+
+ const std::set<std::string>& speaker_names() const;
+ const std::map<std::string, std::unique_ptr<WavReaderInterface>>&
+ audiotrack_readers() const;
+
+ private:
+ // Find unique speaker names.
+ void FindSpeakerNames();
+
+ // Create one WavReader instance for each unique audiotrack.
+ void CreateAudioTrackReaders();
+
+ // Check the speaking turns timing.
+ void CheckTiming();
+
+ rtc::ArrayView<const Turn> timing_;
+ const std::string& audiotracks_path_;
+ std::unique_ptr<WavReaderAbstractFactory> wavreader_abstract_factory_;
+ std::set<std::string> speaker_names_;
+ std::map<std::string, std::unique_ptr<WavReaderInterface>>
+ audiotrack_readers_;
+
+ RTC_DISALLOW_COPY_AND_ASSIGN(MultiEndCall);
+};
+
+} // namespace conversational_speech
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_MULTIEND_CALL_H_
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/timing.cc b/webrtc/modules/audio_processing/test/conversational_speech/timing.cc
index dd2fdc4..0aa44fa 100644
--- a/webrtc/modules/audio_processing/test/conversational_speech/timing.cc
+++ b/webrtc/modules/audio_processing/test/conversational_speech/timing.cc
@@ -13,7 +13,6 @@
#include <fstream>
#include <iostream>
-#include "webrtc/base/array_view.h"
#include "webrtc/base/stringencode.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h
new file mode 100644
index 0000000..b48245b
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h
@@ -0,0 +1,34 @@
+/*
+ * 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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_ABSTRACT_FACTORY_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_ABSTRACT_FACTORY_H_
+
+#include <memory>
+#include <string>
+
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class WavReaderAbstractFactory {
+ public:
+ virtual ~WavReaderAbstractFactory() = default;
+ virtual std::unique_ptr<WavReaderInterface> Create(
+ const std::string& filepath) const = 0;
+};
+
+} // namespace conversational_speech
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_ABSTRACT_FACTORY_H_
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.cc b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.cc
new file mode 100644
index 0000000..b441a04
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.cc
@@ -0,0 +1,58 @@
+/*
+ * 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 "webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.h"
+
+#include "webrtc/base/checks.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+WavReaderAdaptor::WavReaderAdaptor(const std::string& filepath) {
+ // TODO(alessiob): implement.
+}
+
+WavReaderAdaptor::~WavReaderAdaptor() {}
+
+size_t WavReaderAdaptor::ReadFloatSamples(size_t num_samples, float* samples) {
+ // TODO(alessiob): implement.
+ FATAL();
+ return 0u;
+}
+
+size_t WavReaderAdaptor::ReadInt16Samples(
+ size_t num_samples, int16_t* samples) {
+ // TODO(alessiob): implement.
+ FATAL();
+ return 0u;
+}
+
+int WavReaderAdaptor::sample_rate() const {
+ // TODO(alessiob): implement.
+ FATAL();
+ return 0;
+}
+
+size_t WavReaderAdaptor::num_channels() const {
+ // TODO(alessiob): implement.
+ FATAL();
+ return 0u;
+}
+
+size_t WavReaderAdaptor::num_samples() const {
+ // TODO(alessiob): implement.
+ FATAL();
+ return 0u;
+}
+
+} // namespace conversational_speech
+} // namespace test
+} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.h b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.h
new file mode 100644
index 0000000..5489493
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.h
@@ -0,0 +1,42 @@
+/*
+ * 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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_ADAPTOR_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_ADAPTOR_H_
+
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+
+#include <cstddef>
+#include <string>
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class WavReaderAdaptor : public WavReaderInterface {
+ public:
+ explicit WavReaderAdaptor(const std::string& filepath);
+ ~WavReaderAdaptor() override;
+
+ size_t ReadFloatSamples(size_t num_samples, float* samples) override;
+ size_t ReadInt16Samples(size_t num_samples, int16_t* samples) override;
+
+ int sample_rate() const override;
+ size_t num_channels() const override;
+ size_t num_samples() const override;
+};
+
+} // namespace conversational_speech
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_ADAPTOR_H_
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
new file mode 100644
index 0000000..f9d43f8
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.cc
@@ -0,0 +1,30 @@
+/*
+ * 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 "webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h"
+
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_adaptor.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+WavReaderFactory::WavReaderFactory() = default;
+
+WavReaderFactory::~WavReaderFactory() = default;
+
+std::unique_ptr<WavReaderInterface> WavReaderFactory::Create(
+ const std::string& filepath) const {
+ return std::unique_ptr<WavReaderAdaptor>(new WavReaderAdaptor(filepath));
+}
+
+} // namespace conversational_speech
+} // namespace test
+} // namespace webrtc
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h
new file mode 100644
index 0000000..5c46af0
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_factory.h
@@ -0,0 +1,36 @@
+/*
+ * 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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_FACTORY_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_FACTORY_H_
+
+#include <memory>
+#include <string>
+
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_abstract_factory.h"
+#include "webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class WavReaderFactory : public WavReaderAbstractFactory {
+ public:
+ WavReaderFactory();
+ ~WavReaderFactory() override;
+ std::unique_ptr<WavReaderInterface> Create(const std::string& filepath) const
+ override;
+};
+
+} // namespace conversational_speech
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_FACTORY_H_
diff --git a/webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
new file mode 100644
index 0000000..0c99591
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/conversational_speech/wavreader_interface.h
@@ -0,0 +1,40 @@
+/*
+ * 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 WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_INTERFACE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_INTERFACE_H_
+
+#include <stddef.h>
+
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+namespace conversational_speech {
+
+class WavReaderInterface {
+ public:
+ virtual ~WavReaderInterface() = default;
+
+ // Returns the number of samples read.
+ virtual size_t ReadFloatSamples(size_t num_samples, float* samples) = 0;
+ virtual size_t ReadInt16Samples(size_t num_samples, int16_t* samples) = 0;
+
+ // Getters.
+ virtual int sample_rate() const = 0;
+ virtual size_t num_channels() const = 0;
+ virtual size_t num_samples() const = 0;
+};
+
+} // namespace conversational_speech
+} // namespace test
+} // namespace webrtc
+
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_CONVERSATIONAL_SPEECH_WAVREADER_INTERFACE_H_
