Moving/renaming webrtc/common.h.

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 11 matching lines @@
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/helpers.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringencode.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/base/trace_event.h"
-#include "webrtc/common.h"
 #include "webrtc/media/base/audiosource.h"
 #include "webrtc/media/base/mediaconstants.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/engine/payload_type_mapper.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/system_wrappers/include/trace.h"
@@ skipping 488 matching lines @@
   for (const AudioCodec& codec : send_codecs_) {
     LOG(LS_INFO) << ToString(codec);
   }
 
   LOG(LS_INFO) << "Supported recv codecs in order of preference:";
   recv_codecs_ = CollectRecvCodecs();
   for (const AudioCodec& codec : recv_codecs_) {
     LOG(LS_INFO) << ToString(codec);
   }
 
-  voe_config_.Set<webrtc::VoicePacing>(new webrtc::VoicePacing(true));
-
   // Temporarily turn logging level up for the Init() call.
   webrtc::Trace::SetTraceCallback(this);
   webrtc::Trace::set_level_filter(kElevatedTraceFilter);
   LOG(LS_INFO) << webrtc::VoiceEngine::GetVersionString();
   RTC_CHECK_EQ(0, voe_wrapper_->base()->Init(adm_.get(), nullptr,
                                              decoder_factory_));
   webrtc::Trace::set_level_filter(kDefaultTraceFilter);
 
   // No ADM supplied? Get the default one from VoE.
   if (!adm_) {
@@ skipping 242 matching lines @@
     voep->EnableStereoChannelSwapping(*options.stereo_swapping);
     if (voep->IsStereoChannelSwappingEnabled() != *options.stereo_swapping) {
       LOG_RTCERR1(EnableStereoChannelSwapping, *options.stereo_swapping);
       return false;
     }
   }
 
   if (options.audio_jitter_buffer_max_packets) {
     LOG(LS_INFO) << "NetEq capacity is "
                  << *options.audio_jitter_buffer_max_packets;
-    voe_config_.Set<webrtc::NetEqCapacityConfig>(
-        new webrtc::NetEqCapacityConfig(
-            *options.audio_jitter_buffer_max_packets));
+    acm_config_.neteq_config.max_packets_in_buffer =
+        std::max(20, *options.audio_jitter_buffer_max_packets);
   }
-
   if (options.audio_jitter_buffer_fast_accelerate) {
     LOG(LS_INFO) << "NetEq fast mode? "
                  << *options.audio_jitter_buffer_fast_accelerate;
-    voe_config_.Set<webrtc::NetEqFastAccelerate>(
-        new webrtc::NetEqFastAccelerate(
-            *options.audio_jitter_buffer_fast_accelerate));
+    acm_config_.neteq_config.enable_fast_accelerate =
+        *options.audio_jitter_buffer_fast_accelerate;
   }
 
   if (options.typing_detection) {
     LOG(LS_INFO) << "Typing detection is enabled? "
                  << *options.typing_detection;
     if (voep->SetTypingDetectionStatus(*options.typing_detection) == -1) {
       // In case of error, log the info and continue
       LOG_RTCERR1(SetTypingDetectionStatus, *options.typing_detection);
     }
   }
@@ skipping 243 matching lines @@
     if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() !=
         webrtc::AudioProcessing::kNoError) {
       LOG_RTCERR0(StopDebugRecording);
     }
     is_dumping_aec_ = false;
   }
 }
 
 int WebRtcVoiceEngine::CreateVoEChannel() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  return voe_wrapper_->base()->CreateChannel(voe_config_);
+  return voe_wrapper_->base()->CreateChannel(acm_config_);
 }
 
 webrtc::AudioDeviceModule* WebRtcVoiceEngine::adm() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(adm_);
   return adm_;
 }
 
 AudioCodecs WebRtcVoiceEngine::CollectRecvCodecs() const {
   PayloadTypeMapper mapper;
@@ skipping 1573 matching lines @@
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   const auto it = send_streams_.find(ssrc);
   if (it != send_streams_.end()) {
     return it->second->channel();
   }
   return -1;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE