RtcVideoEncoder: use the current timestamp for encoded frames.

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 435 matching lines @@
   }
   DCHECK_EQ(GetStatus(), WEBRTC_VIDEO_CODEC_UNINITIALIZED);
   SetStatus(WEBRTC_VIDEO_CODEC_OK);
   SignalAsyncWaiter(WEBRTC_VIDEO_CODEC_OK);
 }
 
 void RTCVideoEncoder::Impl::BitstreamBufferReady(int32_t bitstream_buffer_id,
     size_t payload_size,
     bool key_frame,
     base::TimeDelta timestamp) {
-  DVLOG(3) << "Impl::BitstreamBufferReady(): "
-              "bitstream_buffer_id=" << bitstream_buffer_id
-           << ", payload_size=" << payload_size
-           << ", key_frame=" << key_frame;
+  DVLOG(3) << "Impl::BitstreamBufferReady(): bitstream_buffer_id="
+           << bitstream_buffer_id << ", payload_size=" << payload_size
+           << ", key_frame=" << key_frame
+           << ", timestamp ms=" << timestamp.InMilliseconds();
   DCHECK(thread_checker_.CalledOnValidThread());
 
   if (bitstream_buffer_id < 0 ||
       bitstream_buffer_id >= static_cast<int>(output_buffers_.size())) {
     LogAndNotifyError(FROM_HERE, "invalid bitstream_buffer_id",
                       media::VideoEncodeAccelerator::kPlatformFailureError);
     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).
-  // This is based on how input timestamps are calculated in
-  // webrtc/video/video_capture_input.cc.
-  const uint32_t rtp_timestamp =
-      static_cast<uint32_t>(timestamp.InMilliseconds()) * 90;
+  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);
 
   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_ = timestamp.InMilliseconds();
+  image.capture_time_ms_ = capture_time_ms;
   image._frameType =
       (key_frame ? webrtc::kVideoFrameKey : webrtc::kVideoFrameDelta);
   image._completeFrame = true;
 
   ReturnEncodedImage(image, bitstream_buffer_id, picture_id_);
   // Picture ID must wrap after reaching the maximum.
   picture_id_ = (picture_id_ + 1) & 0x7FFF;
 }
 
 void RTCVideoEncoder::Impl::NotifyError(
@@ skipping 363 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