
This reverts commit 80c4cca4915dbc6094a5bfae749f85f7371eadd1. Reason for revert: Breaks downstream tests. 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} TBR=henrik.lundin@webrtc.org,ivoc@webrtc.org,minyue@webrtc.org Change-Id: I555aae8850fc4ac1ea919bfa72c11b5218066f30 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:9421 Reviewed-on: https://webrtc-review.googlesource.com/84680 Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23706}
33 lines
1015 B
C++
33 lines
1015 B
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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} // namespace test
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} // namespace webrtc
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