Index: webrtc/video/video_quality_test.cc |
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc |
index 84dbb5104e7c970a0634857cdb82b018e56c02a7..5be9f1ba5745769369082a01011549192e1a28d7 100644 |
--- a/webrtc/video/video_quality_test.cc |
+++ b/webrtc/video/video_quality_test.cc |
@@ -43,6 +43,7 @@ static const int kPayloadTypeVP9 = 124; |
class VideoAnalyzer : public PacketReceiver, |
public Transport, |
+ public I420FrameCallback, |
public VideoRenderer, |
public VideoCaptureInput, |
public EncodedFrameObserver { |
@@ -68,6 +69,8 @@ class VideoAnalyzer : public PacketReceiver, |
frames_recorded_(0), |
frames_processed_(0), |
dropped_frames_(0), |
+ dropped_frames_before_first_encode_(0), |
+ dropped_frames_before_rendering_(0), |
last_render_time_(0), |
rtp_timestamp_delta_(0), |
avg_psnr_threshold_(avg_psnr_threshold), |
@@ -136,18 +139,26 @@ class VideoAnalyzer : public PacketReceiver, |
void IncomingCapturedFrame(const VideoFrame& video_frame) override { |
VideoFrame copy = video_frame; |
copy.set_timestamp(copy.ntp_time_ms() * 90); |
- |
{ |
rtc::CritScope lock(&crit_); |
- if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0) |
- first_send_frame_ = copy; |
- |
frames_.push_back(copy); |
} |
input_->IncomingCapturedFrame(video_frame); |
} |
+ void FrameCallback(VideoFrame* video_frame) { |
+ rtc::CritScope lock(&crit_); |
+ if (first_send_frame_.IsZeroSize() && rtp_timestamp_delta_ == 0) { |
+ while (frames_.front().timestamp() != video_frame->timestamp()) { |
+ ++dropped_frames_before_first_encode_; |
+ frames_.pop_front(); |
+ RTC_CHECK(!frames_.empty()); |
+ } |
+ first_send_frame_ = *video_frame; |
+ } |
+ } |
+ |
bool SendRtp(const uint8_t* packet, |
size_t length, |
const PacketOptions& options) override { |
@@ -162,7 +173,7 @@ class VideoAnalyzer : public PacketReceiver, |
rtc::CritScope lock(&crit_); |
int64_t timestamp = wrap_handler_.Unwrap(header.timestamp); |
- if (rtp_timestamp_delta_ == 0) { |
+ if (!first_send_frame_.IsZeroSize()) { |
rtp_timestamp_delta_ = timestamp - first_send_frame_.timestamp(); |
first_send_frame_.Reset(); |
} |
@@ -195,10 +206,19 @@ class VideoAnalyzer : public PacketReceiver, |
rtc::CritScope lock(&crit_); |
- while (frames_.front().timestamp() < send_timestamp) { |
+ while (frames_.front().timestamp() != send_timestamp) { |
+ if (last_rendered_frame_.IsZeroSize()) { |
+ // No previous frame rendered, this one was dropped after sending but |
+ // before rendering. |
+ ++dropped_frames_before_rendering_; |
+ frames_.pop_front(); |
+ RTC_CHECK(!frames_.empty()); |
+ continue; |
+ } |
AddFrameComparison(frames_.front(), last_rendered_frame_, true, |
render_time_ms); |
frames_.pop_front(); |
+ RTC_DCHECK(!frames_.empty()); |
} |
VideoFrame reference_frame = frames_.front(); |
@@ -349,6 +369,7 @@ class VideoAnalyzer : public PacketReceiver, |
bool dropped, |
int64_t render_time_ms) |
EXCLUSIVE_LOCKS_REQUIRED(crit_) { |
+ RTC_DCHECK(!render.IsZeroSize()); |
int64_t reference_timestamp = wrap_handler_.Unwrap(reference.timestamp()); |
int64_t send_time_ms = send_times_[reference_timestamp]; |
send_times_.erase(reference_timestamp); |
@@ -483,6 +504,10 @@ class VideoAnalyzer : public PacketReceiver, |
PrintResult("sender_time", sender_time_, " ms"); |
printf("RESULT dropped_frames: %s = %d frames\n", test_label_.c_str(), |
dropped_frames_); |
+ printf("RESULT dropped_frames_before_first_encode: %s = %d frames\n", |
+ test_label_.c_str(), dropped_frames_before_first_encode_); |
+ printf("RESULT dropped_frames_before_rendering: %s = %d frames\n", |
+ test_label_.c_str(), dropped_frames_before_rendering_); |
stefan-webrtc
2016/03/14 13:14:03
It's a bit weird that we mix PrintResult with prin
pbos-webrtc
2016/03/14 15:12:58
Done.
|
PrintResult("receiver_time", receiver_time_, " ms"); |
PrintResult("total_delay_incl_network", end_to_end_, " ms"); |
PrintResult("time_between_rendered_frames", rendered_delta_, " ms"); |
@@ -606,6 +631,8 @@ class VideoAnalyzer : public PacketReceiver, |
int frames_recorded_; |
int frames_processed_; |
int dropped_frames_; |
+ int dropped_frames_before_first_encode_; |
+ int dropped_frames_before_rendering_; |
int64_t last_render_time_; |
uint32_t rtp_timestamp_delta_; |
@@ -999,6 +1026,7 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) { |
SetupCommon(&analyzer, &recv_transport); |
video_receive_configs_[params_.ss.selected_stream].renderer = &analyzer; |
+ video_send_config_.pre_encode_callback = &analyzer; |
for (auto& config : video_receive_configs_) |
config.pre_decode_callback = &analyzer; |
RTC_DCHECK(!video_send_config_.post_encode_callback); |