Refactor neteq_rtpplay

webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h

Index: webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
diff --git a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
new file mode 100644
index 0000000000000000000000000000000000000000..2c51998261c3998baa85345f6fafcab173d865a6
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h
@@ -0,0 +1,69 @@
+/*
+ *  Copyright (c) 2016 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_FAKE_DECODE_FROM_FILE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_
+
+#include <memory>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
+#include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
+
+namespace webrtc {
+namespace test {
+
+// Provides an AudioDecoder implementation that delivers audio data from a file.
+// The "encoded" input should contain information about what RTP timestamp the
+// encoding represents, and how many samples the decoder should produce for that
+// encoding. A helper method PrepareEncoded is provided to prepare such
+// encodings. If packets are missing, as determined from the timestamps, the
+// file reading will skip forward to match the loss.
+class FakeDecodeFromFile : public AudioDecoder {
+ public:
+  FakeDecodeFromFile(std::unique_ptr<InputAudioFile> input,
+                     int sample_rate_hz,
+                     bool stereo)
+      : input_(std::move(input)),
+        sample_rate_hz_(sample_rate_hz),
+        stereo_(stereo) {}
+
+  ~FakeDecodeFromFile() = default;
+
+  void Reset() override {}
+
+  int SampleRateHz() const override { return sample_rate_hz_; }
+
+  size_t Channels() const override { return stereo_ ? 2 : 1; }
+
+  int DecodeInternal(const uint8_t* encoded,
+                     size_t encoded_len,
+                     int sample_rate_hz,
+                     int16_t* decoded,
+                     SpeechType* speech_type) override;
+
+  // Helper method. Writes |timestamp| and |samples| to |encoded| in a format
+  // that the FakeDecpdeFromFile decoder will understand. |encoded| must be at
+  // least 8 bytes long.
+  static void PrepareEncoded(uint32_t timestamp,
+                             size_t samples,
+                             rtc::ArrayView<uint8_t> encoded);
+
+ private:
+  std::unique_ptr<InputAudioFile> input_;
+  rtc::Optional<uint32_t> next_timestamp_from_input_;
+  const int sample_rate_hz_;
+  const bool stereo_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_FAKE_DECODE_FROM_FILE_H_