Adding test::AudioSink interface and derived classes

The AudioSink interface is supposed to be used by tests that produce
audio output. Two implementation classes are also provided:

OutputAudioFile: Writes the audio to a pcm file.
AudioChecksum: Calculates the MD5 checksum of the audio.

These will both be used in future changes.

R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17729004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@6490 4adac7df-926f-26a2-2b94-8c16560cd09d

4 files changed, 159 insertions, 0 deletions

diff --git a/modules/audio_coding/neteq/neteq.gypi b/modules/audio_coding/neteq/neteq.gypi
index ccdc9f5d..21ccee41 100644
--- a/modules/audio_coding/neteq/neteq.gypi
+++ b/modules/audio_coding/neteq/neteq.gypi
@@ -182,10 +182,13 @@
             'tools',
           ],
           'sources': [
+            'tools/audio_checksum.h',
             'tools/audio_loop.cc',
             'tools/audio_loop.h',
+            'tools/audio_sink.h',
             'tools/input_audio_file.cc',
             'tools/input_audio_file.h',
+            'tools/output_audio_file.h',
             'tools/packet.cc',
             'tools/packet.h',
             'tools/packet_source.h',
diff --git a/modules/audio_coding/neteq/tools/audio_checksum.h b/modules/audio_coding/neteq/tools/audio_checksum.h
new file mode 100644
index 00000000..ac568265
--- /dev/null
+++ b/modules/audio_coding/neteq/tools/audio_checksum.h
@@ -0,0 +1,60 @@
+/*
+ *  Copyright (c) 2014 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_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
+
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/md5digest.h"
+#include "webrtc/base/stringencode.h"
+#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
+#include "webrtc/system_wrappers/interface/compile_assert.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+class AudioChecksum : public AudioSink {
+ public:
+  AudioChecksum() : finished_(false) {}
+
+  virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
+    if (finished_)
+      return false;
+
+#ifndef WEBRTC_ARCH_LITTLE_ENDIAN
+#error "Big-endian gives a different checksum"
+#endif
+    checksum_.Update(audio, num_samples * sizeof(*audio));
+    return true;
+  }
+
+  // Finalizes the computations, and returns the checksum.
+  std::string Finish() {
+    if (!finished_) {
+      finished_ = true;
+      checksum_.Finish(checksum_result_, rtc::Md5Digest::kSize);
+    }
+    return rtc::hex_encode(checksum_result_, rtc::Md5Digest::kSize);
+  }
+
+ private:
+  rtc::Md5Digest checksum_;
+  char checksum_result_[rtc::Md5Digest::kSize];
+  bool finished_;
+
+  DISALLOW_COPY_AND_ASSIGN(AudioChecksum);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_CHECKSUM_H_
diff --git a/modules/audio_coding/neteq/tools/audio_sink.h b/modules/audio_coding/neteq/tools/audio_sink.h
new file mode 100644
index 00000000..6e159e65
--- /dev/null
+++ b/modules/audio_coding/neteq/tools/audio_sink.h
@@ -0,0 +1,46 @@
+/*
+ *  Copyright (c) 2014 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_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/interface/module_common_types.h"
+#include "webrtc/typedefs.h"
+
+namespace webrtc {
+namespace test {
+
+// Interface class for an object receiving raw output audio from test
+// applications.
+class AudioSink {
+ public:
+  AudioSink();
+  virtual ~AudioSink() {}
+
+  // Writes |num_samples| from |audio| to the AudioSink. Returns true if
+  // successful, otherwise false.
+  virtual bool WriteArray(const int16_t* audio, size_t num_samples) = 0;
+
+  // Writes |audio_frame| to the AudioSink. Returns true if successful,
+  // otherwise false.
+  bool WriteAudioFrame(const AudioFrame& audio_frame) {
+    return WriteArray(
+        audio_frame.data_,
+        audio_frame.samples_per_channel_ * audio_frame.num_channels_);
+  }
+
+ private:
+  DISALLOW_COPY_AND_ASSIGN(AudioSink);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_AUDIO_SINK_H_
diff --git a/modules/audio_coding/neteq/tools/output_audio_file.h b/modules/audio_coding/neteq/tools/output_audio_file.h
new file mode 100644
index 00000000..1d612807
--- /dev/null
+++ b/modules/audio_coding/neteq/tools/output_audio_file.h
@@ -0,0 +1,50 @@
+/*
+ *  Copyright (c) 2014 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_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_
+
+#include <assert.h>
+#include <stdio.h>
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
+
+namespace webrtc {
+namespace test {
+
+class OutputAudioFile : public AudioSink {
+ public:
+  // Creates an OutputAudioFile, opening a file named |file_name| for writing.
+  // The file format is 16-bit signed host-endian PCM.
+  explicit OutputAudioFile(const std::string& file_name) {
+    out_file_ = fopen(file_name.c_str(), "wb");
+  }
+
+  virtual ~OutputAudioFile() {
+    if (out_file_)
+      fclose(out_file_);
+  }
+
+  virtual bool WriteArray(const int16_t* audio, size_t num_samples) OVERRIDE {
+    assert(out_file_);
+    return fwrite(audio, sizeof(*audio), num_samples, out_file_) == num_samples;
+  }
+
+ private:
+  FILE* out_file_;
+
+  DISALLOW_COPY_AND_ASSIGN(OutputAudioFile);
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_OUTPUT_AUDIO_FILE_H_