Adding a new constraint to set NetEq buffer capacity from peerconnection

This change makes it possible to set a custom value for the maximum
capacity of the packet buffer in NetEq (the audio jitter buffer). The
default value is 50 packets, but any value can be set with the new
functionality.

R=jmarusic@webrtc.org, mflodman@webrtc.org, pthatcher@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50869004

Cr-Commit-Position: refs/heads/master@{#9159}

webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc

@@ -10,6 +10,7 @@
 
 #include "webrtc/modules/audio_coding/main/interface/audio_coding_module.h"
 
+#include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/main/acm2/acm_codec_database.h"
 #include "webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h"
@@ -20,13 +21,20 @@ namespace webrtc {
 
 // Create module
 AudioCodingModule* AudioCodingModule::Create(int id) {
-  return Create(id, Clock::GetRealTimeClock());
+  Config config;
+  config.id = id;
+  config.clock = Clock::GetRealTimeClock();
+  return Create(config);
 }
 
 AudioCodingModule* AudioCodingModule::Create(int id, Clock* clock) {
-  AudioCodingModule::Config config;
+  Config config;
   config.id = id;
   config.clock = clock;
+  return Create(config);
+}
+
+AudioCodingModule* AudioCodingModule::Create(const Config& config) {
   return new acm2::AudioCodingModuleImpl(config);
 }
 

webrtc/modules/audio_coding/main/interface/audio_coding_module.h

@@ -99,6 +99,7 @@ class AudioCodingModule {
   //
   static AudioCodingModule* Create(int id);
   static AudioCodingModule* Create(int id, Clock* clock);
+  static AudioCodingModule* Create(const Config& config);
   virtual ~AudioCodingModule() {};
 
   ///////////////////////////////////////////////////////////////////////////