Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(368)

Unified Diff: webrtc/audio/audio_send_stream.cc

Issue 2638083002: Attach TransportFeedbackPacketLossTracker to ANA (PLR only) (Closed)
Patch Set: CR response Created 3 years, 9 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View side-by-side diff with in-line comments
Download patch
Index: webrtc/audio/audio_send_stream.cc
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc
index 6364202aa65ccc573acbb76e29034e813d5e32a3..a146ceb8e977bc0d8f302dd665d2e46dff669f43 100644
--- a/webrtc/audio/audio_send_stream.cc
+++ b/webrtc/audio/audio_send_stream.cc
@@ -40,6 +40,11 @@ bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
} // namespace
namespace internal {
+// TODO(elad.alon): Subsequent CL will make these values experiment-dependent.
+constexpr size_t kPacketLossTrackerMaxWindowSizeMs = 15000;
+constexpr size_t kPlrMinNumAckedPackets = 50;
the sun 2017/03/22 12:06:42 In this context, the abbreviations PLR and RPLR wi
elad.alon_webrtc.org 2017/03/22 14:34:47 Done.
+constexpr size_t kRplrMinNumAckedPairs = 40;
+
AudioSendStream::AudioSendStream(
const webrtc::AudioSendStream::Config& config,
const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
@@ -49,11 +54,15 @@ AudioSendStream::AudioSendStream(
BitrateAllocator* bitrate_allocator,
RtcEventLog* event_log,
RtcpRttStats* rtcp_rtt_stats)
- : worker_queue_(worker_queue),
+ : clock_(Clock::GetRealTimeClock()),
the sun 2017/03/22 12:06:42 Remove.
elad.alon_webrtc.org 2017/03/22 14:34:46 Done.
+ worker_queue_(worker_queue),
config_(config),
audio_state_(audio_state),
bitrate_allocator_(bitrate_allocator),
- send_side_cc_(send_side_cc) {
+ send_side_cc_(send_side_cc),
+ packet_loss_tracker_(kPacketLossTrackerMaxWindowSizeMs,
+ kPlrMinNumAckedPackets,
+ kRplrMinNumAckedPairs) {
LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
RTC_DCHECK_NE(config_.voe_channel_id, -1);
RTC_DCHECK(audio_state_.get());
@@ -72,6 +81,7 @@ AudioSendStream::AudioSendStream(
config_.rtp.nack.rtp_history_ms / 20);
channel_proxy_->RegisterExternalTransport(config.send_transport);
+ send_side_cc_->RegisterTransportFeedbackAdapterObserver(this);
for (const auto& extension : config.rtp.extensions) {
if (extension.uri == RtpExtension::kAudioLevelUri) {
@@ -96,6 +106,7 @@ AudioSendStream::AudioSendStream(
AudioSendStream::~AudioSendStream() {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
+ send_side_cc_->DeRegisterTransportFeedbackAdapterObserver(this);
channel_proxy_->DeRegisterExternalTransport();
channel_proxy_->ResetCongestionControlObjects();
channel_proxy_->SetRtcEventLog(nullptr);
@@ -247,6 +258,31 @@ uint32_t AudioSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
return 0;
}
+void AudioSendStream::OnPacketAdded(uint32_t ssrc, uint16_t seq_num) {
the sun 2017/03/22 12:06:42 Add thread checkers to document on which thread(s)
elad.alon_webrtc.org 2017/03/22 14:34:46 Done.
+ // Only packets that belong to this stream are of interest.
+ if (ssrc == config_.rtp.ssrc) {
+ rtc::CritScope lock(&packet_loss_tracker_cs_);
+ packet_loss_tracker_.OnPacketAdded(seq_num, clock_->TimeInMilliseconds());
+ // TODO(elad.alon): Take care of the following known issue - this could
the sun 2017/03/22 12:06:42 Which following issues?
elad.alon_webrtc.org 2017/03/22 14:34:46 The one immediately after the hyphen. Minyue has a
minyue-webrtc 2017/03/22 16:27:40 The work "following" makes people think an issue o
elad.alon_webrtc.org 2017/03/22 16:34:19 I think the wording of the latest patchset (PS24)
the sun 2017/03/23 11:11:14 The wording has not changed in the sense that the
elad.alon_webrtc.org 2017/03/23 13:37:30 Minyue, I'm going to take up this suggestion, as i
minyue-webrtc 2017/03/23 13:57:16 You are to make it // TODO(elad.alon): This could
elad.alon_webrtc.org 2017/03/23 18:04:23 OK
+ // potentially reset the window, setting both PLR and RPLR to unknown.
+ }
+}
+
+void AudioSendStream::OnNewTransportFeedbackVector(
+ const std::vector<PacketFeedback>& packet_feedback_vector) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ rtc::CritScope lock(&packet_loss_tracker_cs_);
the sun 2017/03/22 12:06:42 packet_loss_tracker_ and channel_proxy_ are both s
elad.alon_webrtc.org 2017/03/22 14:34:46 PLT internals. OnPacketAdded() and OnNewTransportF
the sun 2017/03/23 11:11:14 This one is interesting and tricky. On one hand I
elad.alon_webrtc.org 2017/03/23 13:37:30 One additional thing to consider is that the CL im
minyue-webrtc 2017/03/23 13:57:16 The "greater reward" isn't big. It is not a big de
+ packet_loss_tracker_.OnNewTransportFeedbackVector(packet_feedback_vector);
+ const auto plr = packet_loss_tracker_.GetPacketLossRate();
+ // TODO(elad.alon): Resolve the following known issue - if PLR goes back
+ // to unknown, no indication is given that the previously sent value is no
+ // longer relevant. This will be taken care of with some refactoring which is
+ // now being done.
+ if (plr) {
+ channel_proxy_->OnTwccBasedUplinkPacketLossRate(*plr);
+ }
+}
+
const webrtc::AudioSendStream::Config& AudioSendStream::config() const {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
return config_;

Powered by Google App Engine
This is Rietveld 408576698