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7687ad58b2
This is a reland of 80c4cca491
Original change's description:
> NetEq: Deprecate playout modes Fax, Off and Streaming
>
> 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}
Bug: webrtc:9421
Change-Id: Ice351b635788167f2971b26470f73a5e5fa1a240
Reviewed-on: https://webrtc-review.googlesource.com/86543
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23799}
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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