Moving/renaming webrtc/common.h.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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/voice_engine/channel.h"
 
 #include <algorithm>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call/rtc_event_log.h"
-#include "webrtc/common.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_receiver_strategy.h"
 #include "webrtc/modules/utility/include/audio_frame_operations.h"
@@ skipping 709 matching lines @@
     }
   }
 
   return (highestNeeded);
 }
 
 int32_t Channel::CreateChannel(
     Channel*& channel,
     int32_t channelId,
     uint32_t instanceId,
-    const Config& config,
-    const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory) {
+    const AudioCodingModule::Config& acm_config) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                "Channel::CreateChannel(channelId=%d, instanceId=%d)", channelId,
                instanceId);
 
-  channel = new Channel(channelId, instanceId, config, decoder_factory);
+  channel = new Channel(channelId, instanceId, acm_config);
   if (channel == NULL) {
     WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, channelId),
                  "Channel::CreateChannel() unable to allocate memory for"
                  " channel");
     return -1;
   }
   return 0;
 }
 
 void Channel::PlayNotification(int32_t id, uint32_t durationMs) {
@@ skipping 38 matching lines @@
   rtc::CritScope cs(&_fileCritSect);
 
   _outputFileRecording = false;
   WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::RecordFileEnded() => output file recorder module is"
                " shutdown");
 }
 
 Channel::Channel(int32_t channelId,
                  uint32_t instanceId,
-                 const Config& config,
-                 const rtc::scoped_refptr<AudioDecoderFactory>& decoder_factory)
+                 const AudioCodingModule::Config& acm_config)
     : _instanceId(instanceId),
       _channelId(channelId),
       event_log_proxy_(new RtcEventLogProxy()),
       rtp_header_parser_(RtpHeaderParser::Create()),
       rtp_payload_registry_(
           new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(true))),
       rtp_receive_statistics_(
           ReceiveStatistics::Create(Clock::GetRealTimeClock())),
       rtp_receiver_(
           RtpReceiver::CreateAudioReceiver(Clock::GetRealTimeClock(),
@@ skipping 45 matching lines @@
       _lastPayloadType(0),
       _includeAudioLevelIndication(false),
       _outputSpeechType(AudioFrame::kNormalSpeech),
       _RxVadDetection(false),
       _rxAgcIsEnabled(false),
       _rxNsIsEnabled(false),
       restored_packet_in_use_(false),
       rtcp_observer_(new VoERtcpObserver(this)),
       network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
       associate_send_channel_(ChannelOwner(nullptr)),
-      pacing_enabled_(config.Get<VoicePacing>().enabled),
+      pacing_enabled_(true),  // TODO(solenberg): Remove flag.

[stefan-webrtc, 2016/09/02 08:23:17]

Doesn't look too complicated to actually remove the flag in this CL. Great to
always having it enabled now anyway!

In case you forgot why we added it, it was to make sure the old VoE API still
worked since it wasn't possible to add a pacer in that case, which would cause
packets to be dropped (if I recall correctly).

[the sun, 2016/09/02 08:54:40]

Yes, I realized, and that's why, unfortunately, we need to keep the flag. For
now. :/

       feedback_observer_proxy_(new TransportFeedbackProxy()),
       seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
       rtp_packet_sender_proxy_(new RtpPacketSenderProxy()),
       retransmission_rate_limiter_(new RateLimiter(Clock::GetRealTimeClock(),
                                                    kMaxRetransmissionWindowMs)),
-      decoder_factory_(decoder_factory) {
+      decoder_factory_(acm_config.decoder_factory) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
-  AudioCodingModule::Config acm_config;
-  acm_config.id = VoEModuleId(instanceId, channelId);
-  if (config.Get<NetEqCapacityConfig>().enabled) {
-    // Clamping the buffer capacity at 20 packets. While going lower will
-    // probably work, it makes little sense.
-    acm_config.neteq_config.max_packets_in_buffer =
-        std::max(20, config.Get<NetEqCapacityConfig>().capacity);
-  }
-  acm_config.neteq_config.enable_fast_accelerate =
-      config.Get<NetEqFastAccelerate>().enabled;
-  acm_config.neteq_config.enable_muted_state = true;
-  acm_config.decoder_factory = decoder_factory;
-  audio_coding_.reset(AudioCodingModule::Create(acm_config));
+  AudioCodingModule::Config acm_config_copy(acm_config);
+  acm_config_copy.id = VoEModuleId(instanceId, channelId);
+  acm_config_copy.neteq_config.enable_muted_state = true;
+  audio_coding_.reset(AudioCodingModule::Create(acm_config_copy));
 
   _outputAudioLevel.Clear();
 
   RtpRtcp::Configuration configuration;
   configuration.audio = true;
   configuration.outgoing_transport = this;
   configuration.receive_statistics = rtp_receive_statistics_.get();
   configuration.bandwidth_callback = rtcp_observer_.get();
   if (pacing_enabled_) {
     configuration.paced_sender = rtp_packet_sender_proxy_.get();
@@ skipping 2568 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc