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

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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}
This commit is contained in:
Henrik Lundin
2018-06-21 12:36:28 +00:00
committed by Commit Bot
parent 07efe436c9
commit 1ff41eb784
23 changed files with 782 additions and 479 deletions
Download Patch File Download Diff File Expand all files Collapse all files

45
modules/audio_coding/neteq/tools/neteq_input.cc
View File

@ -28,50 +28,5 @@ 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