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Remove decoded timestamp extrapolation from NackTracker.

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UpdateLastDecodedPacket is anyway only called when a new packet is
decoded.

Bug: webrtc:10178
Change-Id: I8cfcc5791e71079034a2d0806c44b3b071ac2ffb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/299180
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39703}
This commit is contained in:
Jakob Ivarsson 2023-03-27 16:39:26 +02:00 committed by WebRTC LUCI CQ
parent d7510fe1e4
commit d6c4b1641d
3 changed files with 14 additions and 53 deletions
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47
modules/audio_coding/neteq/nack_tracker.cc
View File

@ -145,43 +145,22 @@ uint32_t NackTracker::EstimateTimestamp(uint16_t sequence_num,
return sequence_num_diff * samples_per_packet + timestamp_last_received_rtp_;
}
void NackTracker::UpdateEstimatedPlayoutTimeBy10ms() {
while (!nack_list_.empty() &&
nack_list_.begin()->second.time_to_play_ms <= 10)
nack_list_.erase(nack_list_.begin());
for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end(); ++it)
it->second.time_to_play_ms -= 10;
}
void NackTracker::UpdateLastDecodedPacket(uint16_t sequence_number,
uint32_t timestamp) {
if (IsNewerSequenceNumber(sequence_number, sequence_num_last_decoded_rtp_) ||
!any_rtp_decoded_) {
sequence_num_last_decoded_rtp_ = sequence_number;
timestamp_last_decoded_rtp_ = timestamp;
// Packets in the list with sequence numbers less than the
// sequence number of the decoded RTP should be removed from the lists.
// They will be discarded by the jitter buffer if they arrive.
nack_list_.erase(nack_list_.begin(),
nack_list_.upper_bound(sequence_num_last_decoded_rtp_));
// Update estimated time-to-play.
for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end();
++it)
it->second.time_to_play_ms = TimeToPlay(it->second.estimated_timestamp);
} else {
RTC_DCHECK_EQ(sequence_number, sequence_num_last_decoded_rtp_);
// Same sequence number as before. 10 ms is elapsed, update estimations for
// time-to-play.
UpdateEstimatedPlayoutTimeBy10ms();
// Update timestamp for better estimate of time-to-play, for packets which
// are added to NACK list later on.
timestamp_last_decoded_rtp_ += sample_rate_khz_ * 10;
}
any_rtp_decoded_ = true;
sequence_num_last_decoded_rtp_ = sequence_number;
timestamp_last_decoded_rtp_ = timestamp;
// Packets in the list with sequence numbers less than the
// sequence number of the decoded RTP should be removed from the lists.
// They will be discarded by the jitter buffer if they arrive.
nack_list_.erase(nack_list_.begin(),
nack_list_.upper_bound(sequence_num_last_decoded_rtp_));
// Update estimated time-to-play.
for (NackList::iterator it = nack_list_.begin(); it != nack_list_.end();
++it) {
it->second.time_to_play_ms = TimeToPlay(it->second.estimated_timestamp);
}
}
NackTracker::NackList NackTracker::GetNackList() const {

7
modules/audio_coding/neteq/nack_tracker.h
View File

@ -72,8 +72,7 @@ class NackTracker {
// After Reset() is called sampling rate has to be set.
void UpdateSampleRate(int sample_rate_hz);
// Update the sequence number and the timestamp of the last decoded RTP. This
// API should be called every time 10 ms audio is pulled from NetEq.
// Update the sequence number and the timestamp of the last decoded RTP.
void UpdateLastDecodedPacket(uint16_t sequence_number, uint32_t timestamp);
// Update the sequence number and the timestamp of the last received RTP. This
@ -149,10 +148,6 @@ class NackTracker {
// computed correctly.
NackList GetNackList() const;
// This function subtracts 10 ms of time-to-play for all packets in NACK list.
// This is called when 10 ms elapsed with no new RTP packet decoded.
void UpdateEstimatedPlayoutTimeBy10ms();
// Returns a valid number of samples per packet given the current received
// sequence number and timestamp or nullopt of none could be computed.
absl::optional<int> GetSamplesPerPacket(

13
modules/audio_coding/neteq/nack_tracker_unittest.cc
View File

@ -238,19 +238,6 @@ TEST(NackTrackerTest, EstimateTimestampAndTimeToPlay) {
EXPECT_EQ((index + 2) * kPacketSizeMs, it->second.time_to_play_ms);
++it;
}
// Pretend 10 ms is passed, and we had pulled audio from NetEq, it still
// reports the same sequence number as decoded, time-to-play should be
// updated by 10 ms.
nack.UpdateLastDecodedPacket(first_seq_num, first_timestamp);
nack_list = nack.GetNackList();
it = nack_list.begin();
while (it != nack_list.end()) {
seq_num = it->first - seq_num_offset;
int index = seq_num - kLostPackets[0];
EXPECT_EQ((index + 2) * kPacketSizeMs - 10, it->second.time_to_play_ms);
++it;
}
}
}