Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.

webrtc/audio/audio_send_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_send_stream.h"
 
 #include <string>
 
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/conversion.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/voice_engine/channel_proxy.h"
 #include "webrtc/voice_engine/include/voe_base.h"
-#include "webrtc/voice_engine/include/voe_volume_control.h"
+#include "webrtc/voice_engine/transmit_mixer.h"
 #include "webrtc/voice_engine/voice_engine_impl.h"
 
 namespace webrtc {
 
 namespace {
 
 constexpr char kOpusCodecName[] = "opus";
 
 bool IsCodec(const webrtc::CodecInst& codec, const char* ref_name) {
   return (STR_CASE_CMP(codec.plname, ref_name) == 0);
@@ skipping 147 matching lines @@
         // Convert samples to milliseconds.
         if (codec_inst.plfreq / 1000 > 0) {
           stats.jitter_ms =
               block.interarrival_jitter / (codec_inst.plfreq / 1000);
         }
         break;
       }
     }
   }
 
-  // Local speech level.
-  {
-    ScopedVoEInterface<VoEVolumeControl> volume(voice_engine());
-    unsigned int level = 0;
-    int error = volume->GetSpeechInputLevelFullRange(level);
-    RTC_DCHECK_EQ(0, error);
-    stats.audio_level = static_cast<int32_t>(level);
-  }
+  ScopedVoEInterface<VoEBase> base(voice_engine());
+  RTC_DCHECK(base->transmit_mixer());
+  stats.audio_level = base->transmit_mixer()->AudioLevelFullRange();
+  RTC_DCHECK_LE(0, stats.audio_level);
 
-  ScopedVoEInterface<VoEBase> base(voice_engine());
   RTC_DCHECK(base->audio_processing());
   auto audio_processing_stats = base->audio_processing()->GetStatistics();
   stats.echo_delay_median_ms = audio_processing_stats.delay_median;
   stats.echo_delay_std_ms = audio_processing_stats.delay_standard_deviation;
   stats.echo_return_loss = audio_processing_stats.echo_return_loss.instant();
   stats.echo_return_loss_enhancement =
       audio_processing_stats.echo_return_loss_enhancement.instant();
   stats.residual_echo_likelihood =
       audio_processing_stats.residual_echo_likelihood;
   stats.residual_echo_likelihood_recent_max =
@@ skipping 163 matching lines @@
         LOG(LS_WARNING) << "SetVADStatus() failed.";
         return false;
       }
     }
   }
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc