Moving/renaming webrtc/common.h.

webrtc/voice_engine/channel.cc

Index: webrtc/voice_engine/channel.cc
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index d6daffddf83ee0f751d59665e85b15d89ff2622b..19395e9b2997a7b76b48e9b64657ebdf59d2a710 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -21,7 +21,6 @@
 #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"
@@ -751,13 +750,12 @@ 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"
@@ -816,8 +814,7 @@ void Channel::RecordFileEnded(int32_t id) {
 
 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()),
@@ -883,28 +880,19 @@ Channel::Channel(int32_t channelId,
       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();