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Unified Diff: webrtc/video/video_quality_test.cc

Issue 1757313002: Initialize/configure video encoders asychronously. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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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);

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