Refactor neteq_rtpplay

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

+/*
+ *  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/neteq_packet_source_input.h"
+
+#include <algorithm>
+#include <limits>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/neteq/tools/rtc_event_log_source.h"
+#include "webrtc/modules/audio_coding/neteq/tools/rtp_file_source.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqPacketSourceInput::NetEqPacketSourceInput() : next_output_event_ms_(0) {}
+
+rtc::Optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
+  return packet_
+             ? rtc::Optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
+             : rtc::Optional<int64_t>();
+}
+
+rtc::Optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
+  return packet_ ? rtc::Optional<RTPHeader>(packet_->header())
+                 : rtc::Optional<RTPHeader>();
+}
+
+void NetEqPacketSourceInput::LoadNextPacket() {
+  packet_ = source()->NextPacket();
+}
+
+std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::GetPacket() {
+  if (!packet_) {
+    return std::unique_ptr<NetEqInput::PacketData>();

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

I think the "NetEqInput::" part can be removed (since you use it without it on
line 44).

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

Done.

+  }
+  std::unique_ptr<PacketData> packet_data(new PacketData);
+  packet_->ConvertHeader(&packet_data->header);
+  packet_data->payload.SetData(packet_->payload(),
+                               packet_->payload_length_bytes());
+  packet_data->time_ms = packet_->time_ms();
+
+  LoadNextPacket();
+
+  return packet_data;
+}
+
+NetEqRtpDumpInput::NetEqRtpDumpInput(const std::string& file_name)
+    : source_(RtpFileSource::Create(file_name)) {
+  LoadNextPacket();
+}
+
+rtc::Optional<int64_t> NetEqRtpDumpInput::NextOutputEventTime() const {
+  return next_output_event_ms_;
+}
+
+void NetEqRtpDumpInput::AdvanceOutputEvent() {
+  if (next_output_event_ms_) {
+    *next_output_event_ms_ += kOutputPeriodMs;
+  }
+  if (!NextPacketTime()) {
+    next_output_event_ms_ = rtc::Optional<int64_t>();
+  }
+}
+
+PacketSource* NetEqRtpDumpInput::source() {
+  return source_.get();
+}
+
+NetEqEventLogInput::NetEqEventLogInput(const std::string& file_name)
+    : source_(RtcEventLogSource::Create(file_name)) {
+  LoadNextPacket();
+  AdvanceOutputEvent();
+}
+
+rtc::Optional<int64_t> NetEqEventLogInput::NextOutputEventTime() const {
+  return rtc::Optional<int64_t>(next_output_event_ms_);
+}
+
+void NetEqEventLogInput::AdvanceOutputEvent() {
+  next_output_event_ms_ =
+      rtc::Optional<int64_t>(source_->NextAudioOutputEventMs());
+  if (*next_output_event_ms_ == std::numeric_limits<int64_t>::max()) {
+    next_output_event_ms_ = rtc::Optional<int64_t>();
+  }
+}
+
+PacketSource* NetEqEventLogInput::source() {
+  return source_.get();
+}
+
+}  // namespace test
+}  // namespace webrtc