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NetEq: Deprecate playout modes Fax, Off and Streaming

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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}
This commit is contained in:
Henrik Lundin
2018-06-21 11:13:07 +02:00
committed by Commit Bot
parent c0260b4f2b
commit 80c4cca491
23 changed files with 479 additions and 782 deletions
Download Patch File Download Diff File Expand all files Collapse all files

82
modules/audio_coding/neteq/neteq_impl.cc
View File

@ -101,7 +101,6 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
reset_decoder_(false),
ssrc_(0),
first_packet_(true),
playout_mode_(config.playout_mode),
enable_fast_accelerate_(config.enable_fast_accelerate),
nack_enabled_(false),
enable_muted_state_(config.enable_muted_state),
@ -110,7 +109,8 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
tick_timer_.get()),
speech_expand_uma_logger_("WebRTC.Audio.SpeechExpandRatePercent",
10, // Report once every 10 s.
tick_timer_.get()) {
tick_timer_.get()),
no_time_stretching_(config.for_test_no_time_stretching) {
RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
int fs = config.sample_rate_hz;
if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
@ -358,23 +358,6 @@ int NetEqImpl::FilteredCurrentDelayMs() const {
return static_cast<int>(delay_samples) / rtc::CheckedDivExact(fs_hz_, 1000);
}
// Deprecated.
// TODO(henrik.lundin) Delete.
void NetEqImpl::SetPlayoutMode(NetEqPlayoutMode mode) {
rtc::CritScope lock(&crit_sect_);
if (mode != playout_mode_) {
playout_mode_ = mode;
CreateDecisionLogic();
}
}
// Deprecated.
// TODO(henrik.lundin) Delete.
NetEqPlayoutMode NetEqImpl::PlayoutMode() const {
rtc::CritScope lock(&crit_sect_);
return playout_mode_;
}
int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
rtc::CritScope lock(&crit_sect_);
assert(decoder_database_.get());
@ -937,33 +920,6 @@ int NetEqImpl::GetAudioInternal(AudioFrame* audio_frame, bool* muted) {
return_value = DoDtmf(dtmf_event, &play_dtmf);
break;
}
case kAlternativePlc: {
// TODO(hlundin): Write test for this.
DoAlternativePlc(false);
break;
}
case kAlternativePlcIncreaseTimestamp: {
// TODO(hlundin): Write test for this.
DoAlternativePlc(true);
break;
}
case kAudioRepetitionIncreaseTimestamp: {
// TODO(hlundin): Write test for this.
sync_buffer_->IncreaseEndTimestamp(
static_cast<uint32_t>(output_size_samples_));
// Skipping break on purpose. Execution should move on into the
// next case.
RTC_FALLTHROUGH();
}
case kAudioRepetition: {
// TODO(hlundin): Write test for this.
// Copy last |output_size_samples_| from |sync_buffer_| to
// |algorithm_buffer|.
algorithm_buffer_->PushBackFromIndex(
*sync_buffer_, sync_buffer_->Size() - output_size_samples_);
expand_->Reset();
break;
}
case kUndefined: {
RTC_LOG(LS_ERROR) << "Invalid operation kUndefined.";
assert(false); // This should not happen.
@ -1291,10 +1247,7 @@ int NetEqImpl::GetDecision(Operations* operation,
// Get packets from buffer.
int extracted_samples = 0;
if (packet && *operation != kAlternativePlc &&
*operation != kAlternativePlcIncreaseTimestamp &&
*operation != kAudioRepetition &&
*operation != kAudioRepetitionIncreaseTimestamp) {
if (packet) {
sync_buffer_->IncreaseEndTimestamp(packet->timestamp - end_timestamp);
if (decision_logic_->CngOff()) {
// Adjustment of timestamp only corresponds to an actual packet loss
@ -1883,29 +1836,6 @@ int NetEqImpl::DoDtmf(const DtmfEvent& dtmf_event, bool* play_dtmf) {
return 0;
}
void NetEqImpl::DoAlternativePlc(bool increase_timestamp) {
AudioDecoder* decoder = decoder_database_->GetActiveDecoder();
size_t length;
if (decoder && decoder->HasDecodePlc()) {
// Use the decoder's packet-loss concealment.
// TODO(hlundin): Will probably need a longer buffer for multi-channel.
int16_t decoded_buffer[kMaxFrameSize];
length = decoder->DecodePlc(1, decoded_buffer);
if (length > 0)
algorithm_buffer_->PushBackInterleaved(decoded_buffer, length);
} else {
// Do simple zero-stuffing.
length = output_size_samples_;
algorithm_buffer_->Zeros(length);
// By not advancing the timestamp, NetEq inserts samples.
stats_.AddZeros(length);
}
if (increase_timestamp) {
sync_buffer_->IncreaseEndTimestamp(static_cast<uint32_t>(length));
}
expand_->Reset();
}
int NetEqImpl::DtmfOverdub(const DtmfEvent& dtmf_event,
size_t num_channels,
int16_t* output) const {
@ -2130,8 +2060,8 @@ NetEqImpl::OutputType NetEqImpl::LastOutputType() {
void NetEqImpl::CreateDecisionLogic() {
decision_logic_.reset(DecisionLogic::Create(
fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
*packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
tick_timer_.get()));
fs_hz_, output_size_samples_, no_time_stretching_,
decoder_database_.get(), *packet_buffer_.get(), delay_manager_.get(),
buffer_level_filter_.get(), tick_timer_.get()));
}
} // namespace webrtc