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Adapt NetEq delay to received FEC (both RED and codec inband).

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This is achieved by notifing NetEq controller of all received packets
after splitting, which then does deduping so that only useful packets
are counted.

The goal is to reduce underruns when FEC is used.

The behavior is default enabled with a field trial kill-switch.

Bug: webrtc:13322
Change-Id: I2a1a78ead1a58940ef92da0d43413eda5ba1caf3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/337440
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41665}
This commit is contained in:
Jakob Ivarsson 2024-02-05 11:30:21 +01:00 committed by WebRTC LUCI CQ
parent 35fe95802d
commit f6ae657b07
4 changed files with 61 additions and 20 deletions
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3
experiments/field_trials.py
View File

@ -47,6 +47,9 @@ ACTIVE_FIELD_TRIALS: FrozenSet[FieldTrial] = frozenset([
FieldTrial('WebRTC-Audio-GainController2',
'webrtc:7494',
date(2024, 4, 1)),
FieldTrial('WebRTC-Audio-NetEqFecDelayAdaptation',
'webrtc:13322',
date(2024, 4, 1)),
FieldTrial('WebRTC-Audio-OpusSetSignalVoiceWithDtx',
'webrtc:4559',
date(2024, 4, 1)),

65
modules/audio_coding/neteq/neteq_impl.cc
View File

@ -20,6 +20,7 @@
#include <vector>
#include "api/audio_codecs/audio_decoder.h"
#include "api/neteq/neteq_controller.h"
#include "api/neteq/tick_timer.h"
#include "common_audio/signal_processing/include/signal_processing_library.h"
#include "modules/audio_coding/codecs/cng/webrtc_cng.h"
@ -50,6 +51,7 @@
#include "rtc_base/strings/audio_format_to_string.h"
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
namespace {
@ -174,6 +176,8 @@ NetEqImpl::NetEqImpl(const NetEq::Config& config,
accelerate_factory_(std::move(deps.accelerate_factory)),
preemptive_expand_factory_(std::move(deps.preemptive_expand_factory)),
stats_(std::move(deps.stats)),
enable_fec_delay_adaptation_(
!field_trial::IsDisabled("WebRTC-Audio-NetEqFecDelayAdaptation")),
controller_(std::move(deps.neteq_controller)),
last_mode_(Mode::kNormal),
decoded_buffer_length_(kMaxFrameSize),
@ -695,6 +699,7 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
packet_buffer_->Flush();
buffer_flush_occured = true;
}
NetEqController::PacketArrivedInfo info = ToPacketArrivedInfo(packet);
int return_val = packet_buffer_->InsertPacket(std::move(packet));
if (return_val == PacketBuffer::kFlushed) {
buffer_flush_occured = true;
@ -702,6 +707,15 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
// An error occurred.
return kOtherError;
}
if (enable_fec_delay_adaptation_) {
info.buffer_flush = buffer_flush_occured;
const bool should_update_stats = !new_codec_ && !buffer_flush_occured;
auto relative_delay =
controller_->PacketArrived(fs_hz_, should_update_stats, info);
if (relative_delay) {
stats_->RelativePacketArrivalDelay(relative_delay.value());
}
}
}
if (buffer_flush_occured) {
@ -752,24 +766,26 @@ int NetEqImpl::InsertPacketInternal(const RTPHeader& rtp_header,
}
}
const DecoderDatabase::DecoderInfo* dec_info =
decoder_database_->GetDecoderInfo(main_payload_type);
RTC_DCHECK(dec_info); // Already checked that the payload type is known.
if (!enable_fec_delay_adaptation_) {
const DecoderDatabase::DecoderInfo* dec_info =
decoder_database_->GetDecoderInfo(main_payload_type);
RTC_DCHECK(dec_info); // Already checked that the payload type is known.
NetEqController::PacketArrivedInfo info;
info.is_cng_or_dtmf = dec_info->IsComfortNoise() || dec_info->IsDtmf();
info.packet_length_samples =
number_of_primary_packets * decoder_frame_length_;
info.main_timestamp = main_timestamp;
info.main_sequence_number = main_sequence_number;
info.is_dtx = is_dtx;
info.buffer_flush = buffer_flush_occured;
NetEqController::PacketArrivedInfo info;
info.is_cng_or_dtmf = dec_info->IsComfortNoise() || dec_info->IsDtmf();
info.packet_length_samples =
number_of_primary_packets * decoder_frame_length_;
info.main_timestamp = main_timestamp;
info.main_sequence_number = main_sequence_number;
info.is_dtx = is_dtx;
info.buffer_flush = buffer_flush_occured;
const bool should_update_stats = !new_codec_;
auto relative_delay =
controller_->PacketArrived(fs_hz_, should_update_stats, info);
if (relative_delay) {
stats_->RelativePacketArrivalDelay(relative_delay.value());
const bool should_update_stats = !new_codec_;
auto relative_delay =
controller_->PacketArrived(fs_hz_, should_update_stats, info);
if (relative_delay) {
stats_->RelativePacketArrivalDelay(relative_delay.value());
}
}
return 0;
}
@ -2150,4 +2166,21 @@ NetEqImpl::OutputType NetEqImpl::LastOutputType() {
return OutputType::kNormalSpeech;
}
}
NetEqController::PacketArrivedInfo NetEqImpl::ToPacketArrivedInfo(
const Packet& packet) const {
const DecoderDatabase::DecoderInfo* dec_info =
decoder_database_->GetDecoderInfo(packet.payload_type);
NetEqController::PacketArrivedInfo info;
info.is_cng_or_dtmf =
dec_info && (dec_info->IsComfortNoise() || dec_info->IsDtmf());
info.packet_length_samples =
packet.frame ? packet.frame->Duration() : decoder_frame_length_;
info.main_timestamp = packet.timestamp;
info.main_sequence_number = packet.sequence_number;
info.is_dtx = packet.frame && packet.frame->IsDtxPacket();
return info;
}
} // namespace webrtc

4
modules/audio_coding/neteq/neteq_impl.h
View File

@ -342,6 +342,9 @@ class NetEqImpl : public webrtc::NetEq {
NetEqNetworkStatistics CurrentNetworkStatisticsInternal() const
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
NetEqController::PacketArrivedInfo ToPacketArrivedInfo(
const Packet& packet) const RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
Clock* const clock_;
mutable Mutex mutex_;
@ -363,6 +366,7 @@ class NetEqImpl : public webrtc::NetEq {
const std::unique_ptr<PreemptiveExpandFactory> preemptive_expand_factory_
RTC_GUARDED_BY(mutex_);
const std::unique_ptr<StatisticsCalculator> stats_ RTC_GUARDED_BY(mutex_);
const bool enable_fec_delay_adaptation_ RTC_GUARDED_BY(mutex_);
std::unique_ptr<BackgroundNoise> background_noise_ RTC_GUARDED_BY(mutex_);
std::unique_ptr<NetEqController> controller_ RTC_GUARDED_BY(mutex_);

9
modules/audio_coding/neteq/neteq_unittest.cc
View File

@ -76,12 +76,13 @@ TEST_F(NetEqDecodingTest, MAYBE_TestOpusBitExactness) {
webrtc::test::ResourcePath("audio_coding/neteq_opus", "rtp");
const std::string output_checksum =
"2efdbea92c3fb2383c59f89d881efec9f94001d0|"
"a6831b946b59913852ae3e53f99fa8f209bb23cd";
"434bdc4ec08546510ee903d001c8be1a01c44e24|"
"4336be0091e2faad7a194c16ee0a05e727325727|"
"cefd2de4adfa8f6a9b66a3639ad63c2f6779d0cd";
const std::string network_stats_checksum =
"dfaf4399fd60293405290476ccf1c05c807c71a0|"
"076662525572dba753b11578330bd491923f7f5e";
"5f2c8e3dff9cff55dd7a9f4167939de001566d95|"
"80ab17c17da030d4f2dfbf314ac44aacdadd7f0c";
DecodeAndCompare(input_rtp_file, output_checksum, network_stats_checksum,
absl::GetFlag(FLAGS_gen_ref));