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Store first_frame as const& instead of *

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Bug: webrtc:13343
Change-Id: Id6d73539fa3034be9e7d4e6a27ca5b615ad204da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236842
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35291}
This commit is contained in:
Evan Shrubsole
2021-11-01 10:28:31 +01:00
committed by WebRTC LUCI CQ
parent baf1512a4c
commit 5b9b9aa38b
1 changed files with 7 additions and 7 deletions
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14
modules/video_coding/frame_buffer2.cc
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@ -249,14 +249,14 @@ EncodedFrame* FrameBuffer::GetNextFrame() {
RTC_DCHECK(!frames_to_decode_.empty());
bool superframe_delayed_by_retransmission = false;
size_t superframe_size = 0;
EncodedFrame* first_frame = frames_to_decode_[0]->second.frame.get();
int64_t render_time_ms = first_frame->RenderTime();
int64_t receive_time_ms = first_frame->ReceivedTime();
const EncodedFrame& first_frame = *frames_to_decode_[0]->second.frame;
int64_t render_time_ms = first_frame.RenderTime();
int64_t receive_time_ms = first_frame.ReceivedTime();
// Gracefully handle bad RTP timestamps and render time issues.
if (HasBadRenderTiming(*first_frame, now_ms)) {
if (HasBadRenderTiming(first_frame, now_ms)) {
jitter_estimator_.Reset();
timing_->Reset();
render_time_ms = timing_->RenderTimeMs(first_frame->Timestamp(), now_ms);
render_time_ms = timing_->RenderTimeMs(first_frame.Timestamp(), now_ms);
}
for (FrameMap::iterator& frame_it : frames_to_decode_) {
@ -292,8 +292,8 @@ EncodedFrame* FrameBuffer::GetNextFrame() {
if (!superframe_delayed_by_retransmission) {
int64_t frame_delay;
if (inter_frame_delay_.CalculateDelay(first_frame->Timestamp(),
&frame_delay, receive_time_ms)) {
if (inter_frame_delay_.CalculateDelay(first_frame.Timestamp(), &frame_delay,
receive_time_ms)) {
jitter_estimator_.UpdateEstimate(frame_delay, superframe_size);
}