Update to the neteq_rtpplay utility to support RtcEventLog input files.

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

Index: webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
new file mode 100644
index 0000000000000000000000000000000000000000..c2bcccaaa3ffc0f044a545c37b6d2691503340fb
--- /dev/null
+++ b/webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.cc
@@ -0,0 +1,127 @@
+/*
+ *  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.
+ */
+
+#include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
+
+#include <assert.h>
+#include <string.h>
+#include <iostream>
+#include <limits>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h"
+#include "webrtc/video/rtc_event_log.h"
+
+// Files generated at build-time by the protobuf compiler.
+#ifdef WEBRTC_ANDROID_PLATFORM_BUILD
+#include "external/webrtc/webrtc/video/rtc_event_log.pb.h"
+#else
+#include "webrtc/video/rtc_event_log.pb.h"
+#endif
+
+namespace webrtc {
+namespace test {
+
+namespace {
+
+const rtclog::RtpPacket* GetRtpPacket(const rtclog::Event& event) {
+  if (!event.has_type() || event.type() != rtclog::Event::RTP_EVENT)
+    return nullptr;
+  if (!event.has_timestamp_us() || !event.has_rtp_packet())
+    return nullptr;
+  const rtclog::RtpPacket& rtp_packet = event.rtp_packet();
+  if (!rtp_packet.has_type() || rtp_packet.type() != rtclog::AUDIO ||
+      !rtp_packet.has_incoming() || !rtp_packet.incoming() ||
+      !rtp_packet.has_packet_length() || rtp_packet.packet_length() == 0 ||
+      !rtp_packet.has_header() || rtp_packet.header().size() == 0 ||
+      rtp_packet.packet_length() < rtp_packet.header().size())
+    return nullptr;
+  return &rtp_packet;
+}
+
+const rtclog::DebugEvent* GetAudioOutputEvent(const rtclog::Event& event) {
+  if (!event.has_type() || event.type() != rtclog::Event::DEBUG_EVENT)
+    return nullptr;
+  if (!event.has_timestamp_us() || !event.has_debug_event())
+    return nullptr;
+  const rtclog::DebugEvent& debug_event = event.debug_event();
+  if (!debug_event.has_type() ||
+      debug_event.type() != rtclog::DebugEvent::AUDIO_PLAYOUT)
+    return nullptr;
+  return &debug_event;
+}
+
+}  // namespace
+
+RtcEventLogSource* RtcEventLogSource::Create(const std::string& file_name) {
+  RtcEventLogSource* source = new RtcEventLogSource();
+  CHECK(source->OpenFile(file_name));
+  return source;
+}
+
+RtcEventLogSource::~RtcEventLogSource() {}
+
+bool RtcEventLogSource::RegisterRtpHeaderExtension(RTPExtensionType type,
+                                                   uint8_t id) {
+  CHECK(parser_.get());
+  return parser_->RegisterRtpHeaderExtension(type, id);
+}
+
+Packet* RtcEventLogSource::NextPacket() {
+  while (rtp_packet_index_ < event_log_->stream_size()) {
+    const rtclog::Event& event = event_log_->stream(rtp_packet_index_);
+    const rtclog::RtpPacket* rtp_packet = GetRtpPacket(event);
+    rtp_packet_index_++;
+    if (rtp_packet) {
+      uint8_t* packet_header = new uint8_t[rtp_packet->header().size()];
+      memcpy(packet_header, rtp_packet->header().data(),
+             rtp_packet->header().size());
+      Packet* packet = new Packet(packet_header, rtp_packet->header().size(),
+                                  rtp_packet->packet_length(),
+                                  event.timestamp_us() / 1000, *parser_.get());
+      if (packet->valid_header()) {
+        // Check if the packet should not be filtered out.
+        if (!filter_.test(packet->header().payloadType) &&
+            !(use_ssrc_filter_ && packet->header().ssrc != ssrc_))
+          return packet;
+      } else {
+        std::cout << "Warning: Packet with index " << (rtp_packet_index_ - 1)
+                  << " has an invalid header and will be ignored." << std::endl;
+      }
+      // The packet has either an invalid header or needs to be filtered out, so
+      // it can be deleted.
+      delete packet;
+    }
+  }
+  return nullptr;
+}
+
+int64_t RtcEventLogSource::NextAudioOutputEventMs() {
+  while (audio_output_index_ < event_log_->stream_size()) {
+    const rtclog::Event& event = event_log_->stream(audio_output_index_);
+    const rtclog::DebugEvent* debug_event = GetAudioOutputEvent(event);
+    audio_output_index_++;
+    if (debug_event)
+      return event.timestamp_us() / 1000;
+  }
+  return std::numeric_limits<int64_t>::max();
+}
+
+RtcEventLogSource::RtcEventLogSource()
+    : PacketSource(), parser_(RtpHeaderParser::Create()) {}
+
+bool RtcEventLogSource::OpenFile(const std::string& file_name) {
+  event_log_.reset(new rtclog::EventStream());
+  return RtcEventLog::ParseRtcEventLog(file_name, event_log_.get());
+}
+
+}  // namespace test
+}  // namespace webrtc