Delete all use of tick_util.h.

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #include "webrtc/audio/audio_receive_stream.h"
 
 #include <string>
 #include <utility>
 
 #include "webrtc/audio_sink.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h"
-#include "webrtc/system_wrappers/include/tick_util.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_neteq_stats.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
@@ skipping 129 matching lines @@
   RTPHeader header;
   if (!rtp_header_parser_->Parse(packet, length, &header)) {
     return false;
   }
 
   // Only forward if the parsed header has one of the headers necessary for
   // bandwidth estimation. RTP timestamps has different rates for audio and
   // video and shouldn't be mixed.
   if (remote_bitrate_estimator_ &&
       header.extension.hasTransportSequenceNumber) {
-    int64_t arrival_time_ms = TickTime::MillisecondTimestamp();
+    int64_t arrival_time_ms = rtc::Time64();

[stefan-webrtc, 2016/04/19 09:19:12]

Is there a function with a better name? I'd prefer if it actually said that the
time was in milliseconds.

[nisse-webrtc, 2016/04/19 12:19:25]

There's another pending transition, I think. There's Time32 and Time64, which
both return milliseconds but with different size of the return value. And Time,
which currently is an alias for Time32, with a TODO to change it to Time64.

I could rename Time64 to TimeMs (or add an alias), but I'm not sure that's
appropriate for this cl.

[stefan-webrtc, 2016/04/19 12:40:47]

I'd prefer an int64_t TimeMilliseconds() and maybe also a templated version of
it for the special cases where you want to put the time into for instance the
bitstream. But maybe the int64_t version is enough?

[nisse-webrtc, 2016/04/19 13:48:50]

I think int64_t is enough, no need for templates. I have to check if Time64 is
used anywhere, it's a bit silly with to have to keep that alias.

     if (packet_time.timestamp >= 0)
       arrival_time_ms = (packet_time.timestamp + 500) / 1000;
     size_t payload_size = length - header.headerLength;
     remote_bitrate_estimator_->IncomingPacket(arrival_time_ms, payload_size,
                                               header, false);
   }
   return true;
 }
 
 webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
@@ skipping 56 matching lines @@
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc