Early initialize recording on the ADM from WebRtcVoiceMediaChannel.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 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.
  */
 
@@ skipping 1093 matching lines @@
     return;
   }
   LOG_RTCERR0(StopRtcEventLog);
 }
 
 int WebRtcVoiceEngine::CreateVoEChannel() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return voe_wrapper_->base()->CreateChannel(voe_config_);
 }
 
+webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
+  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
+  RTC_DCHECK(adm_);
+  return adm_;
+}
+
 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     : public AudioSource::Sink {
  public:
   WebRtcAudioSendStream(int ch, webrtc::AudioTransport* voe_audio_transport,
                         uint32_t ssrc, const std::string& c_name,
                         const std::vector<webrtc::RtpExtension>& extensions,
                         webrtc::Call* call)
       : voe_audio_transport_(voe_audio_transport),
         call_(call),
         config_(nullptr) {
@@ skipping 702 matching lines @@
   playout_ = playout;
   return true;
 }
 
 void WebRtcVoiceMediaChannel::SetSend(bool send) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSend");
   if (send_ == send) {
     return;
   }
 
-  // Apply channel specific options when channel is enabled for sending.
+  // Apply channel specific options, and initialize ADM for recording, when
+  // channel is enabled for sending.
   if (send) {
     engine()->ApplyOptions(options_);
+    SetupRecording();
   }
 
   // Change the settings on each send channel.
   for (auto& kv : send_streams_) {
     kv.second->SetSend(send);
   }
 
   send_ = send;
 }
 
@@ skipping 85 matching lines @@
       if (engine()->voe()->rtp()->SetLocalSSRC(recv_channel, ssrc) != 0) {
         LOG_RTCERR2(SetLocalSSRC, recv_channel, ssrc);
         return false;
       }
       engine()->voe()->base()->AssociateSendChannel(recv_channel, channel);
       LOG(LS_INFO) << "VoiceEngine channel #" << recv_channel
                    << " is associated with channel #" << channel << ".";
     }
   }
 
+  // Initialize ADM for recording.
+  if (send_) {
+    SetupRecording();
+  }
   send_streams_[ssrc]->SetSend(send_);
   return true;
 }
 
 bool WebRtcVoiceMediaChannel::RemoveSendStream(uint32_t ssrc) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::RemoveSendStream");
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "RemoveSendStream: " << ssrc;
 
   auto it = send_streams_.find(ssrc);
@@ skipping 561 matching lines @@
     if (engine()->voe()->base()->StartPlayout(channel) == -1) {
       LOG_RTCERR1(StartPlayout, channel);
       return false;
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
+
+void WebRtcVoiceMediaChannel::SetupRecording() {

[Taylor Brandstetter, 2016/03/29 01:05:37]

Maybe add a comment explaining that this is done as a performance optimization,
in case people wonder later why InitRecording is called in two places.

[the sun, 2016/03/31 10:49:47]

Done.

+  if (engine()->adm()->InitRecording() != 0) {
+    LOG(LS_WARNING) << "Foo";
+    return;
+  }
+  if (!send_streams_.empty()) {
+    if (engine()->adm()->StartRecording() != 0) {

[Taylor Brandstetter, 2016/03/29 01:05:37]

It's really only "InitRecording" that was slow on Android. I'm not sure about
the side-effects of calling "StartRecording" early on various platforms, so
maybe we should leave that out for now?

[the sun, 2016/03/31 10:49:47]

Note that StartRecording() is only called once we have a stream to send &&
sending is enabled. Added a comment.

[Taylor Brandstetter, 2016/03/31 18:11:25]

But after my recent change, a stream won't call "Start" until it's enabled for
sending *and* has a source (look in "UpdateSendState"). If an application does
an initial offer/answer just to get ICE connected/etc., sending may be enabled
here even when the application isn't ready to start sending media.

[the sun, 2016/04/01 09:07:46]

Ah, thanks for explaining. I decided to leave the old call to
VoEBase::StartSend() in AudioSendStream and only make the speculative call to
InitRecording() from WVoMC::SetSend().

+      LOG(LS_WARNING) << "Bar";
+    }
+  }
+}
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE