Change NetEq::GetAudio to use AudioFrame

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/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;
 
 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
@@ skipping 120 matching lines @@
   // might insert sync-packet when they observe that buffer level of NetEq is
   // decreasing below a certain threshold, defined by the application.
   // Sync-packets should have the same payload type as the last audio payload
   // type, i.e. they cannot have DTMF or CNG payload type, nor a codec change
   // can be implied by inserting a sync-packet.
   // Returns kOk on success, kFail on failure.
   virtual int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                                uint32_t receive_timestamp) = 0;
 
   // Instructs NetEq to deliver 10 ms of audio data. The data is written to
-  // |output_audio|, which can hold (at least) |max_length| elements.
-  // The number of channels that were written to the output is provided in
-  // the output variable |num_channels|, and each channel contains
-  // |samples_per_channel| elements. If more than one channel is written,
-  // the samples are interleaved.
+  // |audio_frame|. All data in |audio_frame| is wiped; |data_|, |interleaved_|,
+  // |num_channels_|, and |samples_per_channel_| are updated upon success. If
+  // an error is returned, some fields may not have been updated.
   // The speech type is written to |type|, if |type| is not NULL.
   // Returns kOK on success, or kFail in case of an error.
-  virtual int GetAudio(size_t max_length, int16_t* output_audio,
-                       size_t* samples_per_channel, size_t* num_channels,
-                       NetEqOutputType* type) = 0;
+  virtual int GetAudio(AudioFrame* audio_frame, NetEqOutputType* type) = 0;
 
   // Associates |rtp_payload_type| with |codec| and |codec_name|, and stores the
   // information in the codec database. Returns 0 on success, -1 on failure.
   // The name is only used to provide information back to the caller about the
   // decoders. Hence, the name is arbitrary, and may be empty.
   virtual int RegisterPayloadType(NetEqDecoder codec,
                                   const std::string& codec_name,
                                   uint8_t rtp_payload_type) = 0;
 
   // Provides an externally created decoder object |decoder| to insert in the
@@ skipping 113 matching lines @@
 
  protected:
   NetEq() {}
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEq);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_INCLUDE_NETEQ_H_