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}

modules/audio_coding/neteq/tools/neteq_input.h

@@ -78,6 +78,28 @@ class NetEqInput {
   virtual absl::optional<RTPHeader> NextHeader() const = 0;
 };
 
+// Wrapper class to impose a time limit on a NetEqInput object, typically
+// another time limit than what the object itself provides. For example, an
+// input taken from a file can be cut shorter by wrapping it in this class.
+class TimeLimitedNetEqInput : public NetEqInput {
+ public:
+  TimeLimitedNetEqInput(std::unique_ptr<NetEqInput> input, int64_t duration_ms);
+  rtc::Optional<int64_t> NextPacketTime() const override;
+  rtc::Optional<int64_t> NextOutputEventTime() const override;
+  std::unique_ptr<PacketData> PopPacket() override;
+  void AdvanceOutputEvent() override;
+  bool ended() const override;
+  rtc::Optional<RTPHeader> NextHeader() const override;
+
+ private:
+  void MaybeSetEnded();
+
+  std::unique_ptr<NetEqInput> input_;
+  const rtc::Optional<int64_t> start_time_ms_;
+  const int64_t duration_ms_;
+  bool ended_ = false;
+};
+
 }  // namespace test
 }  // namespace webrtc
 #endif  // MODULES_AUDIO_CODING_NETEQ_TOOLS_NETEQ_INPUT_H_