1 files changed, 90 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
new file mode 100644
index 0000000000..55a5653238
--- /dev/null
+++ b/third_party/libwebrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
@@ -0,0 +1,90 @@
+/*
+ *  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 "modules/audio_coding/neteq/tools/neteq_packet_source_input.h"
+
+#include <algorithm>
+#include <limits>
+
+#include "absl/strings/string_view.h"
+#include "modules/audio_coding/neteq/tools/rtp_file_source.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace test {
+
+NetEqPacketSourceInput::NetEqPacketSourceInput() : next_output_event_ms_(0) {}
+
+absl::optional<int64_t> NetEqPacketSourceInput::NextPacketTime() const {
+  return packet_
+             ? absl::optional<int64_t>(static_cast<int64_t>(packet_->time_ms()))
+             : absl::nullopt;
+}
+
+absl::optional<RTPHeader> NetEqPacketSourceInput::NextHeader() const {
+  return packet_ ? absl::optional<RTPHeader>(packet_->header()) : absl::nullopt;
+}
+
+void NetEqPacketSourceInput::LoadNextPacket() {
+  packet_ = source()->NextPacket();
+}
+
+std::unique_ptr<NetEqInput::PacketData> NetEqPacketSourceInput::PopPacket() {
+  if (!packet_) {
+    return std::unique_ptr<PacketData>();
+  }
+  std::unique_ptr<PacketData> packet_data(new PacketData);
+  packet_data->header = packet_->header();
+  if (packet_->payload_length_bytes() == 0 &&
+      packet_->virtual_payload_length_bytes() > 0) {
+    // This is a header-only "dummy" packet. Set the payload to all zeros, with
+    // length according to the virtual length.
+    packet_data->payload.SetSize(packet_->virtual_payload_length_bytes());
+    std::fill_n(packet_data->payload.data(), packet_data->payload.size(), 0);
+  } else {
+    packet_data->payload.SetData(packet_->payload(),
+                                 packet_->payload_length_bytes());
+  }
+  packet_data->time_ms = packet_->time_ms();
+
+  LoadNextPacket();
+
+  return packet_data;
+}
+
+NetEqRtpDumpInput::NetEqRtpDumpInput(absl::string_view file_name,
+                                     const RtpHeaderExtensionMap& hdr_ext_map,
+                                     absl::optional<uint32_t> ssrc_filter)
+    : source_(RtpFileSource::Create(file_name, ssrc_filter)) {
+  for (const auto& ext_pair : hdr_ext_map) {
+    source_->RegisterRtpHeaderExtension(ext_pair.second, ext_pair.first);
+  }
+  LoadNextPacket();
+}
+
+absl::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_ = absl::nullopt;
+  }
+}
+
+PacketSource* NetEqRtpDumpInput::source() {
+  return source_.get();
+}
+
+}  // namespace test
+}  // namespace webrtc