Adding debug dump tests.

webrtc/modules/audio_processing/test/debug_dump_test.h

Index: webrtc/modules/audio_processing/test/debug_dump_test.h
diff --git a/webrtc/modules/audio_processing/test/debug_dump_test.h b/webrtc/modules/audio_processing/test/debug_dump_test.h
new file mode 100644
index 0000000000000000000000000000000000000000..a3b2ca13fd512090a6a5f5572ea731407e4cbf36
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/debug_dump_test.h
@@ -0,0 +1,142 @@
+/*
+ *  Copyright (c) 2015 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_DEBUG_DUMP_TEST_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_TEST_H_
+
+#include <stddef.h>  // size_t
+#include <string>
+#include <vector>
+
+#include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/modules/audio_coding/neteq/tools/resample_input_audio_file.h"
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+
+#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
+#include "webrtc/audio_processing/debug.pb.h"
+#endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
+
+namespace webrtc {
+namespace test {
+
+class DebugDumpGenerator {

[Andrew MacDonald, 2015/10/20 01:22:09]

Do you foresee anything else ever using this? If not, you could put the
declaration direction in the cc file.

[minyue-webrtc, 2015/10/23 08:44:46]

I thought that cc is too large and therefore put this h. You are right. I'd get
them into one cc file.

+ public:
+  DebugDumpGenerator(std::string input_file_name,
+                     int input_file_rate_hz,
+                     size_t input_channels,

[Andrew MacDonald, 2015/10/20 01:22:09]

Since these values are ints in the protobuf generated classes, I think it's
better to use ints here. Bonus: no casting.

[minyue-webrtc, 2015/10/23 08:44:46]

Done.

+                     std::string reverse_file_name,
+                     int reverse_file_rate_hz,
+                     size_t reverse_channels,
+                     const Config& config,
+                     std::string dump_file_name);
+
+  // Changes the sample rate of the input audio to the APM.
+  void SetInputRate(int rate_hz);
+
+  // Sets if converts stereo input signal to mono by discarding other channels.
+  void ForceInputMono(bool mono);
+
+  // Changes the sample rate of the reverse audio to the APM.
+  void SetReverseRate(int rate_hz);
+
+  // Sets if converts stereo reverse signal to mono by discarding other
+  // channels.
+  void ForceReverseMono(bool mono);
+
+  // Sets the required sample rate of the APM output.
+  void set_output_rate_hz(int rate_hz) {
+    output_rate_hz_ = rate_hz;
+  }
+
+  // Sets the required channels of the APM output.
+  void set_output_channels(int channels) {
+    output_channels_ = channels;
+  }
+
+  void StartRecording();
+  void Process(size_t num_blocks);
+  void StopRecording();
+  AudioProcessing* apm() const { return apm_.get(); }
+
+ private:
+  void ReadAndDeinterleave(ResampleInputAudioFile* audio, size_t channels,
+                           size_t frames_per_channel, bool force_mono,
+                           float* const* buffer);
+
+  // APM input/output settings.
+  int input_rate_hz_;

[Andrew MacDonald, 2015/10/20 01:22:09]

Can any of these be const?

[minyue-webrtc, 2015/10/23 08:44:46]

No. We want to change sample rate in the middle, to verify that the dumping cope
well with sample rate changes.

+  bool input_mono_;
+  int reverse_rate_hz_;
+  bool reverse_mono_;
+  int output_rate_hz_;
+  size_t output_channels_;
+
+  // Input file format.
+  rtc::scoped_ptr<ResampleInputAudioFile> input_audio_;

[Andrew MacDonald, 2015/10/20 01:22:09]

You initialize the ResampleInputAudioFiles in the ctor, so you shouldn't need to
use scoped_ptrs.

[minyue-webrtc, 2015/10/23 08:44:46]

Done.

+  size_t input_channels_;

[Andrew MacDonald, 2015/10/20 01:22:09]

const?

[minyue-webrtc, 2015/10/23 08:44:46]

Yes, this can be const

+
+  // Reverse file format.
+  rtc::scoped_ptr<ResampleInputAudioFile> reverse_audio_;
+  size_t reverse_channels_;
+
+  // Buffer for APM input/output.
+  rtc::scoped_ptr<ChannelBuffer<float>> input_;
+  rtc::scoped_ptr<ChannelBuffer<float>> reverse_;
+  rtc::scoped_ptr<ChannelBuffer<float>> output_;
+
+  rtc::scoped_ptr<AudioProcessing> apm_;
+
+  std::string dump_file_name_;

[Andrew MacDonald, 2015/10/20 01:22:09]

const

[minyue-webrtc, 2015/10/23 08:44:46]

Done.

+
+  // Buffer for reading audio files.
+  std::vector<int16_t> signal_;
+};
+
+class DebugDumpTest : public ::testing::Test {

[Andrew MacDonald, 2015/10/20 01:22:09]

This should definitely go in the cc file.

[minyue-webrtc, 2015/10/23 08:44:46]

Done.

+ public:
+  DebugDumpTest();
+
+  // VerifyDebugDump replays a debug dump using APM and verifies that the result
+  // is bit-exact identical to the output channel in the dump. This is only
+  // guaranteed if the debug dump is started on the first frame.
+  void VerifyDebugDump(std::string dump_file_name);
+
+ private:
+  // Following functions are facilities for replaying debug dumps.
+  void OnInitEvent(const audioproc::Init& msg);
+  void OnStreamEvent(const audioproc::Stream& msg);
+  void OnReverseStreamEvent(const audioproc::ReverseStream& msg);
+  void OnConfigEvent(const audioproc::Config& msg);
+  void MaybeRecreateApm(const audioproc::Config& msg);
+  void ConfigurateApm(const audioproc::Config& msg);
+
+  int input_rate_hz_;
+  size_t input_channels_;
+
+  int output_rate_hz_;
+  size_t output_channels_;
+
+  int reverse_rate_hz_;
+  size_t reverse_channels_;
+
+  // Buffer for APM input/output.
+  rtc::scoped_ptr<ChannelBuffer<float>> input_;
+  rtc::scoped_ptr<ChannelBuffer<float>> reverse_;
+  rtc::scoped_ptr<ChannelBuffer<float>> output_;
+
+  rtc::scoped_ptr<AudioProcessing> apm_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_DEBUG_DUMP_TEST_H_