RtcVideoEncoder: use the timestamp from encoder.

content/renderer/media/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/rtc_video_encoder.h"
 
 #include <string.h>
 
 #include "base/bind.h"
 #include "base/location.h"
@@ skipping 455 matching lines @@
     return;
   }
   base::SharedMemory* output_buffer = output_buffers_[bitstream_buffer_id];
   if (payload_size > output_buffer->mapped_size()) {
     LogAndNotifyError(FROM_HERE, "invalid payload_size",
                       media::VideoEncodeAccelerator::kPlatformFailureError);
     return;
   }
   output_buffers_free_count_--;
 
-  // CrOS Nyan provides invalid timestamp. Use the current time for now.
-  // TODO(wuchengli): use the timestamp in BitstreamBufferReady after Nyan is
-  // fixed. http://crbug.com/620565.
-  const int64_t capture_time_us = rtc::TimeMicros();
+  // Derive the capture time (in ms) and RTP timestamp (in 90KHz ticks).
+  int64_t capture_time_ms;
+  uint32_t rtp_timestamp;
 
-  // Derive the capture time (in ms) and RTP timestamp (in 90KHz ticks).
-  const int64_t capture_time_ms =
-      capture_time_us / base::Time::kMicrosecondsPerMillisecond;
-
-  const uint32_t rtp_timestamp = static_cast<uint32_t>(
-      capture_time_us * 90 / base::Time::kMicrosecondsPerMillisecond);
+  if (!timestamp.is_zero()) {
+    capture_time_ms = timestamp.InMilliseconds();
+    rtp_timestamp = static_cast<uint32_t>(timestamp.InMilliseconds()) * 90;

[mcasas, 2016/06/29 14:55:50]

nit: s/timestamp.InMilliseconds()/capture_time_ms/ 
to avoid redundant function call and calculation.

[wuchengli, 2016/07/01 06:19:09]

Now I used the same calculation for both cases because I found we may lose
precision if we use milliseconds * 90. microseconds * 90 / 1000 is better.

+  } else {
+    // Fallback to the current time if encoder does not provide timestamp.
+    const int64_t capture_time_us = rtc::TimeMicros();
+    capture_time_ms = capture_time_us / base::Time::kMicrosecondsPerMillisecond;
+    rtp_timestamp = static_cast<uint32_t>(

[mcasas, 2016/06/29 14:55:50]

nit: We'd probably want |rtp_timestamp| to wrap
around if |capture_time_us| is too large, right?
(here and in l.482) If yes: add a comment for
that ? If no: consider using base::saturatedcast():

https://cs.chromium.org/chromium/src/base/numerics/safe_conversions.h?dr=C&q=...

[wuchengli, 2016/07/01 06:19:09]

Yes. We want RTC timestamp to wrap around. I added the comment.

From book "RTP: Audio and Video for the Internet":
Timestamp wrap-around is a normal part of RTP operation and should be handled by
all applications.

+        capture_time_us * 90 / 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) {
+    base::TimeDelta timestamp =

[mcasas, 2016/06/29 14:55:50]

nit: const

[wuchengli, 2016/07/01 06:19:09]

Done.

+        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,
-        base::TimeDelta::FromMilliseconds(next_frame->ntp_time_ms()));
+        input_buffer->mapped_size(), input_buffer->handle(), 0, timestamp);
     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(),
@@ skipping 281 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