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Add dropped frames metric on the receive side

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Reported to UMA and logged for at the end of the call.

Bug: webrtc:8355
Change-Id: I4ef31bf9e55feaba9cf28be5cb4fcfae929c7179
Reviewed-on: https://webrtc-review.googlesource.com/53760
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22132}
This commit is contained in:
Ilya Nikolaevskiy
2018-02-21 15:57:09 +01:00
committed by Commit Bot
parent 8f83b42946
commit d397a0d46e
8 changed files with 130 additions and 6 deletions
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21
modules/video_coding/packet_buffer.cc
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@ -48,6 +48,7 @@ PacketBuffer::PacketBuffer(Clock* clock,
data_buffer_(start_buffer_size),
sequence_buffer_(start_buffer_size),
received_frame_callback_(received_frame_callback),
unique_frames_seen_(0),
sps_pps_idr_is_h264_keyframe_(
field_trial::IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) {
RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
@ -65,6 +66,8 @@ bool PacketBuffer::InsertPacket(VCMPacket* packet) {
{
rtc::CritScope lock(&crit_);
OnTimestampReceived(packet->timestamp);
uint16_t seq_num = packet->seqNum;
size_t index = seq_num % size_;
@ -207,6 +210,11 @@ rtc::Optional<int64_t> PacketBuffer::LastReceivedKeyframePacketMs() const {
return last_received_keyframe_packet_ms_;
}
int PacketBuffer::GetUniqueFramesSeen() const {
rtc::CritScope lock(&crit_);
return unique_frames_seen_;
}
bool PacketBuffer::ExpandBufferSize() {
if (size_ == max_size_) {
RTC_LOG(LS_WARNING) << "PacketBuffer is already at max size (" << max_size_
@ -484,5 +492,18 @@ void PacketBuffer::UpdateMissingPackets(uint16_t seq_num) {
}
}
void PacketBuffer::OnTimestampReceived(uint32_t rtp_timestamp) {
const size_t kMaxTimestampsHistory = 1000;
if (rtp_timestamps_history_set_.insert(rtp_timestamp).second) {
rtp_timestamps_history_queue_.push(rtp_timestamp);
++unique_frames_seen_;
if (rtp_timestamps_history_set_.size() > kMaxTimestampsHistory) {
uint32_t discarded_timestamp = rtp_timestamps_history_queue_.front();
rtp_timestamps_history_set_.erase(discarded_timestamp);
rtp_timestamps_history_queue_.pop();
}
}
}
} // namespace video_coding
} // namespace webrtc