Reland "NetEq: Deprecate playout modes Fax, Off and Streaming"

This is a reland of 80c4cca4915dbc6094a5bfae749f85f7371eadd1

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}

modules/audio_coding/neteq/tools/neteq_input.cc

@@ -28,5 +28,50 @@ std::string NetEqInput::PacketData::ToString() const {
   return ss.str();
 }
 
+TimeLimitedNetEqInput::TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input,
+                                             int64_t duration_ms)
+    : input_(std::move(input)),
+      start_time_ms_(input_->NextEventTime()),
+      duration_ms_(duration_ms) {}
+
+rtc::Optional<int64_t> TimeLimitedNetEqInput::NextPacketTime() const {
+  return ended_ ? rtc::Optional<int64_t>() : input_->NextPacketTime();
+}
+
+rtc::Optional<int64_t> TimeLimitedNetEqInput::NextOutputEventTime() const {
+  return ended_ ? rtc::Optional<int64_t>() : input_->NextOutputEventTime();
+}
+
+std::unique_ptr<NetEqInput::PacketData> TimeLimitedNetEqInput::PopPacket() {
+  if (ended_) {
+    return std::unique_ptr<PacketData>();
+  }
+  auto packet = input_->PopPacket();
+  MaybeSetEnded();
+  return packet;
+}
+
+void TimeLimitedNetEqInput::AdvanceOutputEvent() {
+  if (!ended_) {
+    input_->AdvanceOutputEvent();
+    MaybeSetEnded();
+  }
+}
+
+bool TimeLimitedNetEqInput::ended() const {
+  return ended_ || input_->ended();
+}
+
+rtc::Optional<RTPHeader> TimeLimitedNetEqInput::NextHeader() const {
+  return ended_ ? rtc::Optional<RTPHeader>() : input_->NextHeader();
+}
+
+void TimeLimitedNetEqInput::MaybeSetEnded() {
+  if (NextEventTime() && start_time_ms_ &&
+      *NextEventTime() - *start_time_ms_ > duration_ms_) {
+    ended_ = true;
+  }
+}
+
 }  // namespace test
 }  // namespace webrtc