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/*
* Copyright (c) 2017 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_input.h"
#include <cstdint>
#include <memory>
#include <optional>
#include <string>
#include <utility>
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/strings/string_builder.h"
namespace webrtc {
namespace test {
std::string NetEqInput::ToString(const RtpPacketReceived& packet) {
StringBuilder ss;
ss << "{"
"time_ms: "
<< packet.arrival_time().ms()
<< ", "
"header: {"
"pt: "
<< static_cast<int>(packet.PayloadType())
<< ", "
"sn: "
<< packet.SequenceNumber()
<< ", "
"ts: "
<< packet.Timestamp()
<< ", "
"ssrc: "
<< packet.Ssrc()
<< "}, "
"payload bytes: "
<< packet.payload_size() << "}";
return ss.Release();
}
TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input,
int64_t duration_ms)
: input_(std::move(input)),
start_time_ms_(input_->NextEventTime()),
duration_ms_(duration_ms) {}
TimeLimitedNetEqInput::~TimeLimitedNetEqInput() = default;
std::optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
return ended_ ? std::nullopt : input_->NextPacketTime();
}
std::optional<int64_t> TimeLimitedNetEqInput::NextOutputEventTime() const {
return ended_ ? std::nullopt : input_->NextOutputEventTime();
}
std::optional<NetEqInput::SetMinimumDelayInfo>
TimeLimitedNetEqInput::NextSetMinimumDelayInfo() const {
return ended_ ? std::nullopt : input_->NextSetMinimumDelayInfo();
}
std::unique_ptr<RtpPacketReceived> TimeLimitedNetEqInput::PopPacket() {
if (ended_) {
return nullptr;
}
auto packet = input_->PopPacket();
MaybeSetEnded();
return packet;
}
void TimeLimitedNetEqInput::AdvanceOutputEvent() {
if (!ended_) {
input_->AdvanceOutputEvent();
MaybeSetEnded();
}
}
void TimeLimitedNetEqInput::AdvanceSetMinimumDelay() {
if (!ended_) {
input_->AdvanceSetMinimumDelay();
MaybeSetEnded();
}
}
bool TimeLimitedNetEqInput::ended() const {
return ended_ || input_->ended();
}
const RtpPacketReceived* TimeLimitedNetEqInput::NextPacket() const {
return ended_ ? nullptr : input_->NextPacket();
}
void TimeLimitedNetEqInput::MaybeSetEnded() {
if (NextEventTime() && start_time_ms_ &&
*NextEventTime() - *start_time_ms_ > duration_ms_) {
ended_ = true;
}
}
} // namespace test
} // namespace webrtc
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