WIP: Move the creation of AudioCodecFactory into PeerConnectionFactory.

webrtc/modules/audio_coding/neteq/include/neteq.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
 #define WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_
 
 #include <string.h>  // Provide access to size_t.
 
 #include <string>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/neteq/audio_decoder_impl.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 // Forward declarations.
 class AudioFrame;
 struct WebRtcRTPHeader;
+class AudioDecoderFactory;
 
 struct NetEqNetworkStatistics {
   uint16_t current_buffer_size_ms;  // Current jitter buffer size in ms.
   uint16_t preferred_buffer_size_ms;  // Target buffer size in ms.
   uint16_t jitter_peaks_found;  // 1 if adding extra delay due to peaky
                                 // jitter; 0 otherwise.
   uint16_t packet_loss_rate;  // Loss rate (network + late) in Q14.
   uint16_t packet_discard_rate;  // Late loss rate in Q14.
   uint16_t expand_rate;  // Fraction (of original stream) of synthesized
                          // audio inserted through expansion (in Q14).
@@ skipping 86 matching lines @@
     kDecodedTooMuch,
     kFrameSplitError,
     kRedundancySplitError,
     kPacketBufferCorruption,
     kSyncPacketNotAccepted
   };
 
   // Creates a new NetEq object, with parameters set in |config|. The |config|
   // object will only have to be valid for the duration of the call to this
   // method.
-  static NetEq* Create(const NetEq::Config& config);
+  static NetEq* Create(
+      const NetEq::Config& config,
+      std::shared_ptr<AudioDecoderFactory> decoder_factory);
 
   virtual ~NetEq() {}
 
   // Inserts a new packet into NetEq. The |receive_timestamp| is an indication
   // of the time when the packet was received, and should be measured with
   // the same tick rate as the RTP timestamp of the current payload.
   // Returns 0 on success, -1 on failure.
   virtual int InsertPacket(const WebRtcRTPHeader& rtp_header,
                            rtc::ArrayView<const uint8_t> payload,
                            uint32_t receive_timestamp) = 0;
@@ skipping 143 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_