| /* | |
| * 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 |