Refactor neteq_rtpplay

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

Index: webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
new file mode 100644
index 0000000000000000000000000000000000000000..b1509c34c13a93b2906478cb20973fd65cd2aac0
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.cc
@@ -0,0 +1,61 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_coding/neteq/tools/fake_decode_from_file.h"
+
+#include "webrtc/base/checks.h"
+#include "webrtc/base/safe_conversions.h"
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+
+namespace webrtc {
+namespace test {
+
+int FakeDecodeFromFile::DecodeInternal(const uint8_t* encoded,
+                                       size_t encoded_len,
+                                       int /*sample_rate_hz*/,
+                                       int16_t* decoded,
+                                       SpeechType* speech_type) {
+  RTC_CHECK_GE(encoded_len, 8u);
+  uint32_t timestamp_to_decode =
+      ByteReader<uint32_t>::ReadLittleEndian(encoded);
+  uint32_t samples_to_decode =
+      ByteReader<uint32_t>::ReadLittleEndian(&encoded[4]);
+
+  if (next_timestamp_from_input_ &&
+      timestamp_to_decode != *next_timestamp_from_input_) {

[ivoc, 2016/06/14 16:39:56]

Why would this happen?

[hlundin-webrtc, 2016/06/17 10:30:08]

This happens when packets are lost (through simulating packet losses). If the
code here detects a gap in the timestamps, it will skip the corresponding number
of samples from the file. Added a comment to explain.

+    uint32_t jump = timestamp_to_decode - *next_timestamp_from_input_;
+    RTC_CHECK(input_->Seek(jump));
+  }
+
+  RTC_CHECK(input_->Read(static_cast<size_t>(samples_to_decode), decoded));
+  next_timestamp_from_input_ =
+      rtc::Optional<uint32_t>(timestamp_to_decode + samples_to_decode);
+
+  if (stereo_) {
+    InputAudioFile::DuplicateInterleaved(decoded, samples_to_decode, 2,
+                                         decoded);
+    samples_to_decode *= 2;
+  }
+
+  *speech_type = kSpeech;
+  return samples_to_decode;
+}
+
+void FakeDecodeFromFile::PrepareEncoded(uint32_t timestamp,
+                                        size_t samples,
+                                        rtc::ArrayView<uint8_t> encoded) {
+  RTC_CHECK_GE(encoded.size(), 8u);
+  ByteWriter<uint32_t>::WriteLittleEndian(&encoded[0], timestamp);
+  ByteWriter<uint32_t>::WriteLittleEndian(&encoded[4],
+                                          rtc::checked_cast<uint32_t>(samples));
+}
+
+}  // namespace test
+}  // namespace webrtc