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Reset RtpFrameReferenceFinder on long pause

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Bug: webrtc:11074
Change-Id: I4c9a8761e9039d32885ccf9ac0eebdffdf67f48d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159240
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29747}
This commit is contained in:
Ilya Nikolaevskiy 2019-11-08 16:20:45 +01:00 committed by Commit Bot
parent a7a2ab4bb0
commit 7a4db6eb0e
3 changed files with 35 additions and 14 deletions
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37
video/rtp_video_stream_receiver.cc
View File

@ -40,6 +40,7 @@
#include "rtc_base/logging.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/system/fallthrough.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/field_trial.h"
#include "system_wrappers/include/metrics.h"
#include "video/receive_statistics_proxy.h"
@ -52,6 +53,10 @@ namespace {
constexpr int kPacketBufferStartSize = 512;
constexpr int kPacketBufferMaxSize = 2048;
// Maximum time between frames before resetting reference_finder to avoid RTP
// fields wraparounds to affect FrameBuffer.
constexpr TimeDelta kInactiveStreamThreshold = TimeDelta::Seconds<5>();
int PacketBufferMaxSize() {
// The group here must be a positive power of 2, in which case that is used as
// size. All other values shall result in the default value being used.
@ -214,7 +219,9 @@ RtpVideoStreamReceiver::RtpVideoStreamReceiver(
rtcp_feedback_buffer_(this, nack_sender, this),
packet_buffer_(clock_, kPacketBufferStartSize, PacketBufferMaxSize()),
has_received_frame_(false),
frames_decryptable_(false) {
frames_decryptable_(false),
last_completed_picture_id_(0),
last_assembled_frame_time_(Timestamp::MinusInfinity()) {
constexpr bool remb_candidate = true;
if (packet_router_)
packet_router_->AddReceiveRtpModule(rtp_rtcp_.get(), remb_candidate);
@ -597,6 +604,15 @@ void RtpVideoStreamReceiver::OnAssembledFrame(
RTC_DCHECK_RUN_ON(&network_tc_);
RTC_DCHECK(frame);
bool recreate_reference_finder_requested = false;
Timestamp now = clock_->CurrentTime();
if (last_assembled_frame_time_.IsFinite() &&
now - last_assembled_frame_time_ > kInactiveStreamThreshold) {
recreate_reference_finder_requested = true;
}
last_assembled_frame_time_ = now;
absl::optional<RtpGenericFrameDescriptor> descriptor =
frame->GetGenericFrameDescriptor();
@ -629,14 +645,7 @@ void RtpVideoStreamReceiver::OnAssembledFrame(
if (frame->codec_type() != current_codec_) {
if (frame_is_newer) {
// When we reset the |reference_finder_| we don't want new picture ids
// to overlap with old picture ids. To ensure that doesn't happen we
// start from the |last_completed_picture_id_| and add an offset in case
// of reordering.
reference_finder_ =
std::make_unique<video_coding::RtpFrameReferenceFinder>(
this, last_completed_picture_id_ +
std::numeric_limits<uint16_t>::max());
recreate_reference_finder_requested = true;
current_codec_ = frame->codec_type();
} else {
// Old frame from before the codec switch, discard it.
@ -652,6 +661,16 @@ void RtpVideoStreamReceiver::OnAssembledFrame(
last_assembled_frame_rtp_timestamp_ = frame->Timestamp();
}
if (recreate_reference_finder_requested) {
// When we reset the |reference_finder_| we don't want new picture ids
// to overlap with old picture ids. To ensure that doesn't happen we
// start from the |last_completed_picture_id_| and add an offset in case
// of reordering.
reference_finder_ = std::make_unique<video_coding::RtpFrameReferenceFinder>(
this,
last_completed_picture_id_ + std::numeric_limits<uint16_t>::max());
}
if (buffered_frame_decryptor_ == nullptr) {
reference_finder_->ManageFrame(std::move(frame));
} else {

3
video/rtp_video_stream_receiver.h
View File

@ -319,7 +319,8 @@ class RtpVideoStreamReceiver : public LossNotificationSender,
std::atomic<bool> frames_decryptable_;
absl::optional<ColorSpace> last_color_space_;
int64_t last_completed_picture_id_ = 0;
int64_t last_completed_picture_id_;
Timestamp last_assembled_frame_time_ RTC_GUARDED_BY(network_tc_);
};
} // namespace webrtc

9
video/video_receive_stream.cc
View File

@ -159,10 +159,9 @@ class EncodedFrameForMediaTransport : public video_coding::EncodedFrame {
int64_t RenderTime() const override { return 0; }
};
// TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
// Maximum time between frames before resetting the FrameBuffer to avoid RTP
// timestamps wraparound to affect FrameBuffer.
constexpr int kInactiveStreamThresholdMs = 600000; // 10 minutes.
// timestamps and picture IDs wraparounds to affect FrameBuffer.
constexpr int kInactiveStreamThresholdMs = 5000; // 5 seconds.
} // namespace
@ -546,7 +545,9 @@ void VideoReceiveStream::RequestKeyFrame() {
void VideoReceiveStream::OnCompleteFrame(
std::unique_ptr<video_coding::EncodedFrame> frame) {
RTC_DCHECK_RUN_ON(&network_sequence_checker_);
// TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
// Resetting of stream state if there was long enough pause in the stream.
// This is done to avoid undetected wraparounds in RTP fields.
int64_t time_now_ms = rtc::TimeMillis();
if (last_complete_frame_time_ms_ > 0 &&
time_now_ms - last_complete_frame_time_ms_ > kInactiveStreamThresholdMs) {