Remove base::SupportsWeakPtr from {Audio,Video}Sender

media/cast/sender/audio_sender.cc

 // Copyright 2014 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 "media/cast/sender/audio_sender.h"
 
 #include <utility>
 
 #include "base/bind.h"
 #include "base/logging.h"
@@ skipping 12 matching lines @@
     : FrameSender(cast_environment,
                   transport_sender,
                   audio_config,
                   NewFixedCongestionControl(audio_config.max_bitrate)),
       samples_in_encoder_(0),
       weak_factory_(this) {
   if (!audio_config.use_external_encoder) {
     audio_encoder_.reset(new AudioEncoder(
         cast_environment, audio_config.channels, audio_config.rtp_timebase,
         audio_config.max_bitrate, audio_config.codec,
-        base::Bind(&AudioSender::OnEncodedAudioFrame,
-                   weak_factory_.GetWeakPtr(), audio_config.max_bitrate)));
+        base::Bind(&AudioSender::OnEncodedAudioFrame, AsWeakPtr(),
+                   audio_config.max_bitrate)));
   }
 
   // AudioEncoder provides no operational status changes during normal use.
   // Post a task now with its initialization result status to allow the client
   // to start sending frames.
   cast_environment_->PostTask(
       CastEnvironment::MAIN,
       FROM_HERE,
       base::Bind(status_change_cb,
                  audio_encoder_ ? audio_encoder_->InitializationResult() :
@@ skipping 20 matching lines @@
   const base::TimeDelta next_frame_duration =
       RtpTimeDelta::FromTicks(audio_bus->frames()).ToTimeDelta(rtp_timebase());
   if (ShouldDropNextFrame(next_frame_duration))
     return;
 
   samples_in_encoder_ += audio_bus->frames();
 
   audio_encoder_->InsertAudio(std::move(audio_bus), recorded_time);
 }
 
+base::WeakPtr<AudioSender> AudioSender::AsWeakPtr() {
+  return weak_factory_.GetWeakPtr();
+}
+
 int AudioSender::GetNumberOfFramesInEncoder() const {
   // Note: It's possible for a partial frame to be in the encoder, but returning
   // the floor() is good enough for the "design limit" check in FrameSender.
   return samples_in_encoder_ / audio_encoder_->GetSamplesPerFrame();
 }
 
 base::TimeDelta AudioSender::GetInFlightMediaDuration() const {
   const int samples_in_flight = samples_in_encoder_ +
       GetUnacknowledgedFrameCount() * audio_encoder_->GetSamplesPerFrame();
   return RtpTimeDelta::FromTicks(samples_in_flight).ToTimeDelta(rtp_timebase());
 }
 
 void AudioSender::OnEncodedAudioFrame(
     int encoder_bitrate,
     std::unique_ptr<SenderEncodedFrame> encoded_frame,
     int samples_skipped) {
   DCHECK(cast_environment_->CurrentlyOn(CastEnvironment::MAIN));
 
   samples_in_encoder_ -= audio_encoder_->GetSamplesPerFrame() + samples_skipped;
   DCHECK_GE(samples_in_encoder_, 0);
 
   SendEncodedFrame(encoder_bitrate, std::move(encoded_frame));
 }
 
 }  // namespace cast
 }  // namespace media