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80c4cca491
The playout modes other than Normal have not been reachable for a long time, other than through tests. It is time to deprecate them. The only meaningful use was that Fax mode was sometimes set from tests, in order to avoid time-stretching operations (accelerate and pre-emptive expand) from messing with the test results. With this CL, a new config is added instead, which lets the user specify exactly this: don't do time-stretching. As a result of Fax and Off modes being removed, the following code clean-up was done: - Fold DecisionLogicNormal into DecisionLogic. - Remove AudioRepetition and AlternativePlc operations, since they can no longer be reached. Bug: webrtc:9421 Change-Id: I651458e9c1931a99f3b07e242817d303bac119df Reviewed-on: https://webrtc-review.googlesource.com/84123 Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23704}
77 lines
2.3 KiB
C++
77 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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