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Split out counting unique rtp timestamps from packet_buffer

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Bug: None
Change-Id: Ia6fd05f284e8304cf56ab9ddf944fb222a4c9573
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158676
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29656}
This commit is contained in:
Danil Chapovalov
2019-10-30 14:12:24 +01:00
committed by Commit Bot
parent a0adf3d440
commit 09860e0bc3
9 changed files with 149 additions and 91 deletions
Download Patch File Download Diff File Expand all files Collapse all files

20
modules/video_coding/packet_buffer.cc
View File

@ -40,7 +40,6 @@ PacketBuffer::PacketBuffer(Clock* clock,
first_packet_received_(false),
is_cleared_to_first_seq_num_(false),
buffer_(start_buffer_size),
unique_frames_seen_(0),
sps_pps_idr_is_h264_keyframe_(
field_trial::IsEnabled("WebRTC-SpsPpsIdrIsH264Keyframe")) {
RTC_DCHECK_LE(start_buffer_size, max_buffer_size);
@ -56,7 +55,6 @@ PacketBuffer::~PacketBuffer() {
PacketBuffer::InsertResult PacketBuffer::InsertPacket(VCMPacket* packet) {
PacketBuffer::InsertResult result;
rtc::CritScope lock(&crit_);
OnTimestampReceived(packet->timestamp);
uint16_t seq_num = packet->seqNum;
size_t index = seq_num % buffer_.size();
@ -208,11 +206,6 @@ absl::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 (buffer_.size() == max_size_) {
RTC_LOG(LS_WARNING) << "PacketBuffer is already at max size (" << max_size_
@ -486,18 +479,5 @@ 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