b661c658da
This reverts commit 6f5b0f920af08d66e6b77ee4f91ade5797145368. Reason for revert: Breaks internal project. Original change's description: > Remove rtc::Optional alias and api:optional target > > Update left-overs where old target still was used. > > Bug: webrtc:9078 > Change-Id: I2162c928091fc4ff1dea33a3f03adbe47207d206 > Reviewed-on: https://webrtc-review.googlesource.com/84740 > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#23913} TBR=danilchap@webrtc.org,kwiberg@webrtc.org Change-Id: I95f5ec33520b823c3d0c9cb83d945d6a15355367 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9078 Reviewed-on: https://webrtc-review.googlesource.com/88140 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23921}
78 lines
2.3 KiB
C++
78 lines
2.3 KiB
C++
/*
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* Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_coding/neteq/tools/neteq_input.h"
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#include <sstream>
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namespace webrtc {
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namespace test {
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std::string NetEqInput::PacketData::ToString() const {
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std::stringstream ss;
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ss << "{"
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<< "time_ms: " << static_cast<int64_t>(time_ms) << ", "
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<< "header: {"
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<< "pt: " << static_cast<int>(header.payloadType) << ", "
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<< "sn: " << header.sequenceNumber << ", "
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<< "ts: " << header.timestamp << ", "
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<< "ssrc: " << header.ssrc << "}, "
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<< "payload bytes: " << payload.size() << "}";
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return ss.str();
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}
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TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input,
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int64_t duration_ms)
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: input_(std::move(input)),
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start_time_ms_(input_->NextEventTime()),
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duration_ms_(duration_ms) {}
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rtc::Optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
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return ended_ ? rtc::Optional<int64_t>() : input_->NextPacketTime();
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}
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rtc::Optional<int64_t> TimeLimitedNetEqInput::NextOutputEventTime() const {
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return ended_ ? rtc::Optional<int64_t>() : input_->NextOutputEventTime();
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}
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std::unique_ptr<NetEqInput::PacketData> TimeLimitedNetEqInput::PopPacket() {
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if (ended_) {
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return std::unique_ptr<PacketData>();
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}
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auto packet = input_->PopPacket();
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MaybeSetEnded();
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return packet;
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}
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void TimeLimitedNetEqInput::AdvanceOutputEvent() {
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if (!ended_) {
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input_->AdvanceOutputEvent();
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MaybeSetEnded();
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}
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}
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bool TimeLimitedNetEqInput::ended() const {
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return ended_ || input_->ended();
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}
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rtc::Optional<RTPHeader> TimeLimitedNetEqInput::NextHeader() const {
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return ended_ ? rtc::Optional<RTPHeader>() : input_->NextHeader();
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}
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void TimeLimitedNetEqInput::MaybeSetEnded() {
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if (NextEventTime() && start_time_ms_ &&
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*NextEventTime() - *start_time_ms_ > duration_ms_) {
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ended_ = true;
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}
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}
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} // namespace test
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} // namespace webrtc
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