1ff41eb784
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}
84 lines
2.9 KiB
C++
84 lines
2.9 KiB
C++
/*
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* Copyright (c) 2016 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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#ifndef MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
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#define MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
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#include <algorithm>
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#include <memory>
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#include <string>
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#include "absl/types/optional.h"
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#include "common_types.h" // NOLINT(build/include)
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#include "modules/audio_coding/neteq/tools/packet.h"
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#include "modules/audio_coding/neteq/tools/packet_source.h"
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#include "rtc_base/buffer.h"
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namespace webrtc {
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namespace test {
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// Interface class for input to the NetEqTest class.
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class NetEqInput {
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public:
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struct PacketData {
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std::string ToString() const;
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RTPHeader header;
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rtc::Buffer payload;
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int64_t time_ms;
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};
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virtual ~NetEqInput() = default;
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// Returns at what time (in ms) NetEq::InsertPacket should be called next, or
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// empty if the source is out of packets.
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virtual absl::optional<int64_t> NextPacketTime() const = 0;
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// Returns at what time (in ms) NetEq::GetAudio should be called next, or
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// empty if no more output events are available.
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virtual absl::optional<int64_t> NextOutputEventTime() const = 0;
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// Returns the time (in ms) for the next event from either NextPacketTime()
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// or NextOutputEventTime(), or empty if both are out of events.
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absl::optional<int64_t> NextEventTime() const {
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const auto a = NextPacketTime();
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const auto b = NextOutputEventTime();
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// Return the minimum of non-empty |a| and |b|, or empty if both are empty.
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if (a) {
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return b ? std::min(*a, *b) : a;
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}
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return b ? b : absl::nullopt;
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}
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// Returns the next packet to be inserted into NetEq. The packet following the
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// returned one is pre-fetched in the NetEqInput object, such that future
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// calls to NextPacketTime() or NextHeader() will return information from that
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// packet.
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virtual std::unique_ptr<PacketData> PopPacket() = 0;
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// Move to the next output event. This will make NextOutputEventTime() return
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// a new value (potentially the same if several output events share the same
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// time).
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virtual void AdvanceOutputEvent() = 0;
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// Returns true if the source has come to an end. An implementation must
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// eventually return true from this method, or the test will end up in an
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// infinite loop.
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virtual bool ended() const = 0;
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// Returns the RTP header for the next packet, i.e., the packet that will be
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// delivered next by PopPacket().
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virtual absl::optional<RTPHeader> NextHeader() const = 0;
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};
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} // namespace test
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} // namespace webrtc
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#endif // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_
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