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Cleanup obsolete filtering of small packets in delay based estimator.

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Also deletes unused constructor in Results struct.

Bug: webrtc:10932
Change-Id: Id33f57db30df49aa23fb0b5959812cc3834f1eaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196508
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32777}
This commit is contained in:
Christoffer Rodbro
2020-12-04 14:17:49 +01:00
committed by Commit Bot
parent d708677d7a
commit d13178cb55
2 changed files with 7 additions and 68 deletions
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48
modules/congestion_controller/goog_cc/delay_based_bwe.cc
View File

@ -46,23 +46,8 @@ constexpr double kTimestampToMs =
constexpr uint32_t kFixedSsrc = 0;
} // namespace
constexpr char BweIgnoreSmallPacketsSettings::kKey[];
constexpr char BweSeparateAudioPacketsSettings::kKey[];
BweIgnoreSmallPacketsSettings::BweIgnoreSmallPacketsSettings(
const WebRtcKeyValueConfig* key_value_config) {
Parser()->Parse(
key_value_config->Lookup(BweIgnoreSmallPacketsSettings::kKey));
}
std::unique_ptr<StructParametersParser>
BweIgnoreSmallPacketsSettings::Parser() {
return StructParametersParser::Create("smoothing", &smoothing_factor, //
"fraction_large", &fraction_large, //
"large", &large_threshold, //
"small", &small_threshold);
}
BweSeparateAudioPacketsSettings::BweSeparateAudioPacketsSettings(
const WebRtcKeyValueConfig* key_value_config) {
Parser()->Parse(
@ -84,20 +69,12 @@ DelayBasedBwe::Result::Result()
recovered_from_overuse(false),
backoff_in_alr(false) {}
DelayBasedBwe::Result::Result(bool probe, DataRate target_bitrate)
: updated(true),
probe(probe),
target_bitrate(target_bitrate),
recovered_from_overuse(false),
backoff_in_alr(false) {}
DelayBasedBwe::DelayBasedBwe(const WebRtcKeyValueConfig* key_value_config,
RtcEventLog* event_log,
NetworkStatePredictor* network_state_predictor)
: event_log_(event_log),
key_value_config_(key_value_config),
ignore_small_(key_value_config),
fraction_large_packets_(0.5),
separate_audio_(key_value_config),
audio_packets_since_last_video_(0),
last_video_packet_recv_time_(Timestamp::MinusInfinity()),
@ -118,11 +95,9 @@ DelayBasedBwe::DelayBasedBwe(const WebRtcKeyValueConfig* key_value_config,
alr_limited_backoff_enabled_(absl::StartsWith(
key_value_config->Lookup("WebRTC-Bwe-AlrLimitedBackoff"),
"Enabled")) {
RTC_LOG(LS_INFO) << "Initialized DelayBasedBwe with small packet filtering "
<< ignore_small_.Parser()->Encode()
<< ", separate audio overuse detection"
<< separate_audio_.Parser()->Encode()
<< " and alr limited backoff "
RTC_LOG(LS_INFO)
<< "Initialized DelayBasedBwe with separate audio overuse detection"
<< separate_audio_.Parser()->Encode() << " and alr limited backoff "
<< (alr_limited_backoff_enabled_ ? "enabled" : "disabled");
}
@ -193,22 +168,6 @@ void DelayBasedBwe::IncomingPacketFeedback(const PacketResult& packet_feedback,
}
last_seen_packet_ = at_time;
// Ignore "small" packets if many/most packets in the call are "large". The
// packet size may have a significant effect on the propagation delay,
// especially at low bandwidths. Variations in packet size will then show up
// as noise in the delay measurement. By default, we include all packets.
DataSize packet_size = packet_feedback.sent_packet.size;
if (!ignore_small_.small_threshold.IsZero()) {
double is_large =
static_cast<double>(packet_size >= ignore_small_.large_threshold);
fraction_large_packets_ +=
ignore_small_.smoothing_factor * (is_large - fraction_large_packets_);
if (packet_size <= ignore_small_.small_threshold &&
fraction_large_packets_ >= ignore_small_.fraction_large) {
return;
}
}
// As an alternative to ignoring small packets, we can separate audio and
// video packets for overuse detection.
InterArrival* inter_arrival_for_packet = video_inter_arrival_.get();
@ -246,6 +205,7 @@ void DelayBasedBwe::IncomingPacketFeedback(const PacketResult& packet_feedback,
uint32_t timestamp_delta = 0;
int64_t recv_delta_ms = 0;
int size_delta = 0;
DataSize packet_size = packet_feedback.sent_packet.size;
bool calculated_deltas = inter_arrival_for_packet->ComputeDeltas(
timestamp, packet_feedback.receive_time.ms(), at_time.ms(),
packet_size.bytes(), &timestamp_delta, &recv_delta_ms, &size_delta);

25
modules/congestion_controller/goog_cc/delay_based_bwe.h
View File

@ -31,21 +31,6 @@
namespace webrtc {
class RtcEventLog;
struct BweIgnoreSmallPacketsSettings {
static constexpr char kKey[] = "WebRTC-BweIgnoreSmallPacketsFix";
BweIgnoreSmallPacketsSettings() = default;
explicit BweIgnoreSmallPacketsSettings(
const WebRtcKeyValueConfig* key_value_config);
double smoothing_factor = 0.1;
double fraction_large = 1.0;
DataSize large_threshold = DataSize::Zero();
DataSize small_threshold = DataSize::Zero();
std::unique_ptr<StructParametersParser> Parser();
};
struct BweSeparateAudioPacketsSettings {
static constexpr char kKey[] = "WebRTC-Bwe-SeparateAudioPackets";
@ -64,7 +49,6 @@ class DelayBasedBwe {
public:
struct Result {
Result();
Result(bool probe, DataRate target_bitrate);
~Result() = default;
bool updated;
bool probe;
@ -112,19 +96,14 @@ class DelayBasedBwe {
Timestamp at_time);
// Updates the current remote rate estimate and returns true if a valid
// estimate exists.
bool UpdateEstimate(Timestamp now,
bool UpdateEstimate(Timestamp at_time,
absl::optional<DataRate> acked_bitrate,
DataRate* target_bitrate);
DataRate* target_rate);
rtc::RaceChecker network_race_;
RtcEventLog* const event_log_;
const WebRtcKeyValueConfig* const key_value_config_;
// Filtering out small packets. Intention is to base the detection only
// on video packets even if we have TWCC sequence numbers for audio.
BweIgnoreSmallPacketsSettings ignore_small_;
double fraction_large_packets_;
// Alternatively, run two separate overuse detectors for audio and video,
// and fall back to the audio one if we haven't seen a video packet in a
// while.