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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

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

@ -101,6 +101,7 @@ 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),
@ -109,8 +110,7 @@ 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()),
no_time_stretching_(config.for_test_no_time_stretching) {
tick_timer_.get()) {
RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
int fs = config.sample_rate_hz;
if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
@ -358,6 +358,23 @@ 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());
@ -920,6 +937,33 @@ 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.
@ -1247,7 +1291,10 @@ int NetEqImpl::GetDecision(Operations* operation,
// Get packets from buffer.
int extracted_samples = 0;
if (packet) {
if (packet && *operation != kAlternativePlc &&
*operation != kAlternativePlcIncreaseTimestamp &&
*operation != kAudioRepetition &&
*operation != kAudioRepetitionIncreaseTimestamp) {
sync_buffer_->IncreaseEndTimestamp(packet->timestamp - end_timestamp);
if (decision_logic_->CngOff()) {
// Adjustment of timestamp only corresponds to an actual packet loss
@ -1836,6 +1883,29 @@ 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 {
@ -2060,8 +2130,8 @@ NetEqImpl::OutputType NetEqImpl::LastOutputType() {
void NetEqImpl::CreateDecisionLogic() {
decision_logic_.reset(DecisionLogic::Create(
fs_hz_, output_size_samples_, no_time_stretching_,
decoder_database_.get(), *packet_buffer_.get(), delay_manager_.get(),
buffer_level_filter_.get(), tick_timer_.get()));
fs_hz_, output_size_samples_, playout_mode_, decoder_database_.get(),
*packet_buffer_.get(), delay_manager_.get(), buffer_level_filter_.get(),
tick_timer_.get()));
}
} // namespace webrtc