Drop dependency on webrtc's tick_utils.h.

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"
 #include "base/logging.h"
 #include "base/macros.h"
 #include "base/memory/scoped_vector.h"
 #include "base/metrics/histogram.h"
 #include "base/numerics/safe_conversions.h"
 #include "base/rand_util.h"
 #include "base/single_thread_task_runner.h"
 #include "base/synchronization/waitable_event.h"
 #include "base/thread_task_runner_handle.h"
 #include "media/base/bind_to_current_loop.h"
 #include "media/base/bitstream_buffer.h"
 #include "media/base/video_frame.h"
 #include "media/base/video_util.h"
 #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/system_wrappers/include/tick_util.h"
+#include "third_party/webrtc/base/timeutils.h"
 
 namespace content {
 
 namespace {
 
 // Translate from webrtc::VideoCodecType and webrtc::VideoCodec to
 // media::VideoCodecProfile.
 media::VideoCodecProfile WebRTCVideoCodecToVideoCodecProfile(
     webrtc::VideoCodecType type,
     const webrtc::VideoCodec* codec_settings) {
@@ skipping 376 matching lines @@
   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_--;
 
   // Use webrtc timestamps to ensure correct RTP sender behavior.
   // TODO(hshi): obtain timestamp from the capturer, see crbug.com/350106.
-  const int64_t capture_time_us = webrtc::TickTime::MicrosecondTimestamp();
+  const int64_t capture_time_us = rtc::TimeMicros();

[perkj_webrtc, 2016/04/14 13:46:23]

IS the todo above easy to fix? This should come from the input frame and not be
stamped here. This is a bug and leads to wrong timestamp at reconstruction of
the frame on the receiving side.

[nisse-webrtc, 2016/04/19 11:12:41]

No idea really, but I don't think this cl should depend on that. pbos and
jansson have commented on the bug recently.

 
   // Derive the capture time (in ms) and RTP timestamp (in 90KHz ticks).
   const int64_t capture_time_ms = capture_time_us / 1000;
   const uint32_t rtp_timestamp =
       static_cast<uint32_t>(capture_time_us * 90 / 1000);
 
   std::unique_ptr<webrtc::EncodedImage> image(new webrtc::EncodedImage(
       reinterpret_cast<uint8_t*>(output_buffer->memory()), payload_size,
       output_buffer->mapped_size()));
   image->_encodedWidth = input_visible_size_.width();
@@ skipping 371 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