/* * 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 "rtc_base/strings/string_builder.h" namespace webrtc { namespace test { NetEqInput::PacketData::PacketData() = default; NetEqInput::PacketData::~PacketData() = default; std::string NetEqInput::PacketData::ToString() const { rtc::StringBuilder ss; ss << "{" "time_ms: " << static_cast(time_ms) << ", " "header: {" "pt: " << static_cast(header.payloadType) << ", " "sn: " << header.sequenceNumber << ", " "ts: " << header.timestamp << ", " "ssrc: " << header.ssrc << "}, " "payload bytes: " << payload.size() << "}"; return ss.Release(); } TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr input, int64_t duration_ms) : input_(std::move(input)), start_time_ms_(input_->NextEventTime()), duration_ms_(duration_ms) {} TimeLimitedNetEqInput::~TimeLimitedNetEqInput() = default; absl::optional TimeLimitedNetEqInput::NextPacketTime() const { return ended_ ? absl::nullopt : input_->NextPacketTime(); } absl::optional TimeLimitedNetEqInput::NextOutputEventTime() const { return ended_ ? absl::nullopt : input_->NextOutputEventTime(); } std::unique_ptr TimeLimitedNetEqInput::PopPacket() { if (ended_) { return std::unique_ptr(); } auto packet = input_->PopPacket(); MaybeSetEnded(); return packet; } void TimeLimitedNetEqInput::AdvanceOutputEvent() { if (!ended_) { input_->AdvanceOutputEvent(); MaybeSetEnded(); } } bool TimeLimitedNetEqInput::ended() const { return ended_ || input_->ended(); } absl::optional TimeLimitedNetEqInput::NextHeader() const { return ended_ ? absl::nullopt : input_->NextHeader(); } void TimeLimitedNetEqInput::MaybeSetEnded() { if (NextEventTime() && start_time_ms_ && *NextEventTime() - *start_time_ms_ > duration_ms_) { ended_ = true; } } } // namespace test } // namespace webrtc