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Adding a delay line to NetEq's output

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This change adds an optional delay to NetEq's output. Note, this is not
equivalent to increasing the jitter buffer with the same extra length.

Bug: b/156734419
Change-Id: I8b70b6b3bffcfd3da296ccf29853864baa03d6bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175110
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31343}
This commit is contained in:
Henrik Lundin
2020-05-25 11:26:15 +02:00
committed by Commit Bot
parent 848ea9f0d3
commit c49e9c253f
12 changed files with 306 additions and 14 deletions
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58
modules/audio_coding/neteq/neteq_impl.cc
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@ -140,7 +140,10 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
10, // Report once every 10 s.
tick_timer_.get()),
no_time_stretching_(config.for_test_no_time_stretching),
enable_rtx_handling_(config.enable_rtx_handling) {
enable_rtx_handling_(config.enable_rtx_handling),
output_delay_chain_(
rtc::CheckedDivExact(config.extra_output_delay_ms, 10)),
output_delay_chain_ms_(config.extra_output_delay_ms) {
RTC_LOG(LS_INFO) << "NetEq config: " << config.ToString();
int fs = config.sample_rate_hz;
if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000) {
@ -255,6 +258,25 @@ int NetEqImpl::GetAudio(AudioFrame* audio_frame,
last_output_sample_rate_hz_ == 32000 ||
last_output_sample_rate_hz_ == 48000)
<< "Unexpected sample rate " << last_output_sample_rate_hz_;
if (!output_delay_chain_.empty()) {
if (output_delay_chain_empty_) {
for (auto& f : output_delay_chain_) {
f.CopyFrom(*audio_frame);
}
output_delay_chain_empty_ = false;
delayed_last_output_sample_rate_hz_ = last_output_sample_rate_hz_;
} else {
RTC_DCHECK_GE(output_delay_chain_ix_, 0);
RTC_DCHECK_LT(output_delay_chain_ix_, output_delay_chain_.size());
swap(output_delay_chain_[output_delay_chain_ix_], *audio_frame);
*muted = audio_frame->muted();
output_delay_chain_ix_ =
(output_delay_chain_ix_ + 1) % output_delay_chain_.size();
delayed_last_output_sample_rate_hz_ = audio_frame->sample_rate_hz();
}
}
return kOK;
}
@ -297,7 +319,8 @@ bool NetEqImpl::SetMinimumDelay(int delay_ms) {
rtc::CritScope lock(&crit_sect_);
if (delay_ms >= 0 && delay_ms <= 10000) {
assert(controller_.get());
return controller_->SetMinimumDelay(delay_ms);
return controller_->SetMinimumDelay(
std::max(delay_ms - output_delay_chain_ms_, 0));
}
return false;
}
@ -306,7 +329,8 @@ bool NetEqImpl::SetMaximumDelay(int delay_ms) {
rtc::CritScope lock(&crit_sect_);
if (delay_ms >= 0 && delay_ms <= 10000) {
assert(controller_.get());
return controller_->SetMaximumDelay(delay_ms);
return controller_->SetMaximumDelay(
std::max(delay_ms - output_delay_chain_ms_, 0));
}
return false;
}
@ -327,7 +351,7 @@ int NetEqImpl::GetBaseMinimumDelayMs() const {
int NetEqImpl::TargetDelayMs() const {
rtc::CritScope lock(&crit_sect_);
RTC_DCHECK(controller_.get());
return controller_->TargetLevelMs();
return controller_->TargetLevelMs() + output_delay_chain_ms_;
}
int NetEqImpl::FilteredCurrentDelayMs() const {
@ -337,7 +361,8 @@ int NetEqImpl::FilteredCurrentDelayMs() const {
const int delay_samples =
controller_->GetFilteredBufferLevel() + sync_buffer_->FutureLength();
// The division below will truncate. The return value is in ms.
return delay_samples / rtc::CheckedDivExact(fs_hz_, 1000);
return delay_samples / rtc::CheckedDivExact(fs_hz_, 1000) +
output_delay_chain_ms_;
}
int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
@ -351,6 +376,13 @@ int NetEqImpl::NetworkStatistics(NetEqNetworkStatistics* stats) {
stats->jitter_peaks_found = controller_->PeakFound();
stats_->GetNetworkStatistics(fs_hz_, total_samples_in_buffers,
decoder_frame_length_, stats);
// Compensate for output delay chain.
stats->current_buffer_size_ms += output_delay_chain_ms_;
stats->preferred_buffer_size_ms += output_delay_chain_ms_;
stats->mean_waiting_time_ms += output_delay_chain_ms_;
stats->median_waiting_time_ms += output_delay_chain_ms_;
stats->min_waiting_time_ms += output_delay_chain_ms_;
stats->max_waiting_time_ms += output_delay_chain_ms_;
return 0;
}
@ -394,12 +426,19 @@ absl::optional<uint32_t> NetEqImpl::GetPlayoutTimestamp() const {
// which is indicated by returning an empty value.
return absl::nullopt;
}
return timestamp_scaler_->ToExternal(playout_timestamp_);
size_t sum_samples_in_output_delay_chain = 0;
for (const auto& audio_frame : output_delay_chain_) {
sum_samples_in_output_delay_chain += audio_frame.samples_per_channel();
}
return timestamp_scaler_->ToExternal(
playout_timestamp_ -
static_cast<uint32_t>(sum_samples_in_output_delay_chain));
}
int NetEqImpl::last_output_sample_rate_hz() const {
rtc::CritScope lock(&crit_sect_);
return last_output_sample_rate_hz_;
return delayed_last_output_sample_rate_hz_.value_or(
last_output_sample_rate_hz_);
}
absl::optional<NetEq::DecoderFormat> NetEqImpl::GetDecoderFormat(
@ -1988,8 +2027,9 @@ int NetEqImpl::ExtractPackets(size_t required_samples,
extracted_samples = packet->timestamp - first_timestamp + packet_duration;
RTC_DCHECK(controller_);
stats_->JitterBufferDelay(packet_duration, waiting_time_ms,
controller_->TargetLevelMs());
stats_->JitterBufferDelay(
packet_duration, waiting_time_ms + output_delay_chain_ms_,
controller_->TargetLevelMs() + output_delay_chain_ms_);
packet_list->push_back(std::move(*packet)); // Store packet in list.
packet = absl::nullopt; // Ensure it's never used after the move.