Use webrtc::VideoFrame timestamp in RTCVideoEncoder

content/renderer/media/gpu/rtc_video_encoder.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include "content/renderer/media/gpu/rtc_video_encoder.h"
 
 #include <string.h>
 
 #include "base/bind.h"
 #include "base/location.h"
@@ skipping 14 matching lines @@
 #include "media/filters/h264_parser.h"
 #include "media/renderers/gpu_video_accelerator_factories.h"
 #include "media/video/video_encode_accelerator.h"
 #include "third_party/libyuv/include/libyuv.h"
 #include "third_party/webrtc/base/timeutils.h"
 
 namespace content {
 
 namespace {
 
+// Used for timestamp conversions.
+static const int kMsToRtpTimestamp = 90;
+
 // Translate from webrtc::VideoCodecType and webrtc::VideoCodec to
 // media::VideoCodecProfile.
 media::VideoCodecProfile WebRTCVideoCodecToVideoCodecProfile(
     webrtc::VideoCodecType type,
     const webrtc::VideoCodec* codec_settings) {
   DCHECK_EQ(type, codec_settings->codecType);
   switch (type) {
     case webrtc::kVideoCodecVP8:
       return media::VP8PROFILE_ANY;
     case webrtc::kVideoCodecH264: {
@@ skipping 434 matching lines @@
 
   if (!timestamp.is_zero()) {
     capture_time_us = timestamp.InMicroseconds();;
     capture_time_ms = timestamp.InMilliseconds();
   } else {
     // Fallback to the current time if encoder does not provide timestamp.
     capture_time_us = rtc::TimeMicros();
     capture_time_ms = capture_time_us / base::Time::kMicrosecondsPerMillisecond;
   }
   // RTP timestamp can wrap around. Get the lower 32 bits.
-  rtp_timestamp = static_cast<uint32_t>(
-      capture_time_us * 90 / base::Time::kMicrosecondsPerMillisecond);
+  rtp_timestamp =
+      static_cast<uint32_t>(capture_time_us * kMsToRtpTimestamp /
+                            base::Time::kMicrosecondsPerMillisecond);
 
   webrtc::EncodedImage image(
       reinterpret_cast<uint8_t*>(output_buffer->memory()), payload_size,
       output_buffer->mapped_size());
   image._encodedWidth = input_visible_size_.width();
   image._encodedHeight = input_visible_size_.height();
   image._timeStamp = rtp_timestamp;
   image.capture_time_ms_ = capture_time_ms;
   image._frameType =
       (key_frame ? webrtc::kVideoFrameKey : webrtc::kVideoFrameDelta);
@@ skipping 63 matching lines @@
   scoped_refptr<media::VideoFrame> frame;
   if (next_frame->video_frame_buffer()->native_handle()) {
     frame = static_cast<media::VideoFrame*>(
         next_frame->video_frame_buffer()->native_handle());
     requires_copy = RequiresSizeChange(frame);
   } else {
     requires_copy = true;
   }
 
   if (requires_copy) {
-    const base::TimeDelta timestamp =
-        frame ? frame->timestamp()
-              : base::TimeDelta::FromMilliseconds(next_frame->ntp_time_ms());
     base::SharedMemory* input_buffer = input_buffers_[index];
     frame = media::VideoFrame::WrapExternalSharedMemory(
         media::PIXEL_FORMAT_I420, input_frame_coded_size_,
         gfx::Rect(input_visible_size_), input_visible_size_,
         reinterpret_cast<uint8_t*>(input_buffer->memory()),
-        input_buffer->mapped_size(), input_buffer->handle(), 0, timestamp);
+        input_buffer->mapped_size(), input_buffer->handle(), 0,
+        base::TimeDelta());
     if (!frame.get()) {
       LogAndNotifyError(FROM_HERE, "failed to create frame",
                         media::VideoEncodeAccelerator::kPlatformFailureError);
       return;
     }
     // Do a strided copy of the input frame to match the input requirements for
     // the encoder.
     // TODO(sheu): support zero-copy from WebRTC.  http://crbug.com/269312
     if (libyuv::I420Copy(next_frame->video_frame_buffer()->DataY(),
                          next_frame->video_frame_buffer()->StrideY(),
                          next_frame->video_frame_buffer()->DataU(),
                          next_frame->video_frame_buffer()->StrideU(),
                          next_frame->video_frame_buffer()->DataV(),
                          next_frame->video_frame_buffer()->StrideV(),
                          frame->data(media::VideoFrame::kYPlane),
                          frame->stride(media::VideoFrame::kYPlane),
                          frame->data(media::VideoFrame::kUPlane),
                          frame->stride(media::VideoFrame::kUPlane),
                          frame->data(media::VideoFrame::kVPlane),
                          frame->stride(media::VideoFrame::kVPlane),
                          next_frame->width(), next_frame->height())) {
       LogAndNotifyError(FROM_HERE, "Failed to copy buffer",
                         media::VideoEncodeAccelerator::kPlatformFailureError);
       return;
     }
   }
+  // Use the timestamp set from WebRTC and convert it from 90 kHz.
+  frame->set_timestamp(
+      base::TimeDelta::FromMicroseconds(
+          next_frame->timestamp() * base::Time::kMicrosecondsPerMillisecond /

[pbos, 2016/08/10 17:56:52]

Does this have wrapping issues? timestamp() is a 32bit right? so timestamp() *
1000 should wrap, should this be cast to a int64_t before being passed to
FromMicroseconds?

[emircan, 2016/08/10 22:11:52]

Thanks for pointing this out. Actually we have problems even with int64_t for
some numeric cases.

Suppose WebRTC's timestamp() is 30000 (in 90 kHZ). Then, we set Chrome's
frame->timestamp() in microseconds as (30000/90*1000)= 333333.333333. This drops
the decimal bits. Even if we use FromMillisecondsD(), we fail to cover all the
information. As a result, we may not get the same timestamp after converting
back, which results in random jumps in googEncodeMs. 

Instead, I changed it to use base::TimeDelta::FromInternalValue() ctor and keep
it in 90 kHZ. This removes the convertions overall.

+          kMsToRtpTimestamp));
   frame->AddDestructionObserver(media::BindToCurrentLoop(
       base::Bind(&RTCVideoEncoder::Impl::EncodeFrameFinished, this, index)));
   video_encoder_->Encode(frame, next_frame_keyframe);
   input_buffers_free_.pop_back();
   SignalAsyncWaiter(WEBRTC_VIDEO_CODEC_OK);
 }
 
 void RTCVideoEncoder::Impl::EncodeFrameFinished(int index) {
   DVLOG(3) << "Impl::EncodeFrameFinished(): index=" << index;
   DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 255 matching lines @@
   UMA_HISTOGRAM_BOOLEAN("Media.RTCVideoEncoderInitEncodeSuccess",
                         init_retval == WEBRTC_VIDEO_CODEC_OK);
   if (init_retval == WEBRTC_VIDEO_CODEC_OK) {
     UMA_HISTOGRAM_ENUMERATION("Media.RTCVideoEncoderProfile",
                               profile,
                               media::VIDEO_CODEC_PROFILE_MAX + 1);
   }
 }
 
 }  // namespace content