Update to the neteq_rtpplay utility to support RtcEventLog input files.

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

Index: webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
new file mode 100644
index 0000000000000000000000000000000000000000..d144b516a28cf9e8030cb1a0e9e29882e359a981
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h
@@ -0,0 +1,70 @@
+/*
+ *  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_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+#define WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_
+
+#include <string>
+
+#include "webrtc/base/constructormagic.h"
+#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
+
+namespace webrtc {
+
+class RtpHeaderParser;
+
+namespace rtclog {
+class EventStream;
+}  // namespace rtclog
+
+namespace test {
+
+class Packet;
+
+class RtcEventLogSource : public PacketSource {
+ public:
+  // Creates an RtcEventLogSource reading from |file_name|. If the file cannot
+  // be opened, or has the wrong format, NULL will be returned.
+  static RtcEventLogSource* Create(const std::string& file_name);
+
+  virtual ~RtcEventLogSource();
+
+  // Registers an RTP header extension and binds it to |id|.
+  virtual bool RegisterRtpHeaderExtension(RTPExtensionType type, uint8_t id);
+
+  // Returns a pointer to the next packet. Returns NULL if end of file was
+  // reached.
+  Packet* NextPacket() override;
+
+  // Returns the timestamp of the next audio output event, in milliseconds. The
+  // maximum value of int64_t is returned if there are no more audio output
+  // events available.
+  int64_t NextAudioOutputEventMs();
+
+ private:
+  RtcEventLogSource();
+
+  bool OpenFile(const std::string& file_name);
+
+  int rtp_packet_index_ = 0;
+  int audio_output_index_ = 0;
+
+  rtc::scoped_ptr<rtclog::EventStream> event_log_;
+  rtc::scoped_ptr<RtpHeaderParser> parser_;
+
+  DISALLOW_COPY_AND_ASSIGN(RtcEventLogSource);
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_TOOLS_RTC_EVENT_LOG_SOURCE_H_