webrtc/modules/audio_coding/neteq/tools/neteq_packet_source_input.cc
Henrik Lundin 9f2e624024 Break out NetEqEventLogInput to separate source files
Building NetEqEventLogInput requires protobuf support, while building
NetEqRtpDumpInput located in the same file does not. This makes both
classes unusable when protobuf support is lacking. With this CL, the
NetEqEventLogInput is broken out into separate files, to allow usage
of NetEqRtpDumpInput even when protobufs are not supported.

Bug: webrtc:9421
Change-Id: I55efec4ec259713654566cdaa00d2e34c5e9a60f
Reviewed-on: https://webrtc-review.googlesource.com/84587
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23803}
2018-07-02 14:15:29 +00:00

94 lines
2.9 KiB
C++

/*
* 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 "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;
}
void NetEqPacketSourceInput::SelectSsrc(uint32_t ssrc) {
source()->SelectSsrc(ssrc);
if (packet_ && packet_->header().ssrc != ssrc)
LoadNextPacket();
}
NetEqRtpDumpInput::NetEqRtpDumpInput(const std::string& file_name,
const RtpHeaderExtensionMap& hdr_ext_map)
: source_(RtpFileSource::Create(file_name)) {
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