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 1060 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 701 matching lines @@
   }
   playout_ = playout;
   return true;
 }
 
 void WebRtcVoiceMediaChannel::SetSend(bool send) {
   TRACE_EVENT0("webrtc", "WebRtcVoiceMediaChannel::SetSend");
   if (send_ == send) {
     return;
   }
+  send_ = send;
 
-  // Apply channel specific options when channel is enabled for sending.
-  if (send) {
+  // Apply channel specific options.
+  if (send_) {
     engine()->ApplyOptions(options_);
   }
 
+  SetupRecording();

[juberti2, 2016/03/31 18:57:38]

Won't this happen anyway as a result of SetSend?

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

Well, it used to, but in the patch set you comment on here, AudioSendStream had
been changed to call directly onto voe::Channel (instead of going through
VoEBase), meaning only the channel state  would be altered and the ADM
untouched, hence the explicit call to SetupRecording() here.

+
   // Change the settings on each send channel.
   for (auto& kv : send_streams_) {
     kv.second->SetSend(send);
   }
-
-  send_ = send;
 }
 
 bool WebRtcVoiceMediaChannel::SetAudioSend(uint32_t ssrc,
                                            bool enable,
                                            const AudioOptions* options,
                                            AudioSource* source) {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   // TODO(solenberg): The state change should be fully rolled back if any one of
   //                  these calls fail.
   if (!SetLocalSource(ssrc, source)) {
@@ skipping 77 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 << ".";
     }
   }
 
+  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 568 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() {
+  if (send_ && !engine()->adm()->Recording()) {
+    // InitRecording() is called as soon as we know we will be sending, as it
+    // may take time on some platforms (e.g. Android).
+    if (engine()->adm()->InitRecording() != 0) {
+      LOG(LS_WARNING) << "Failed to initialize recording";

[juberti2, 2016/03/31 18:57:38]

Shouldn't we propagate this error upwards instead of silently failing?

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

We used to propagate it up, but eventually, in pc/channel.cc, the return value
would be ignored. See:
https://codereview.webrtc.org/1741933002/diff/280001/webrtc/pc/channel.cc
(line 1457)
The decision was made to simplify the logic and maintain current semantics.

+      return;
+    }
+    // StartRecording() is only called once we actually have a stream to send.
+    if (!send_streams_.empty() && engine()->adm()->StartRecording() != 0) {
+      LOG(LS_WARNING) << "Failed to start recording";
+      return;
+    }
+  }
+}
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE