Change NetEq::GetAudio to use AudioFrame

webrtc/modules/audio_coding/neteq/neteq_impl.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.
  */
 
@@ skipping 78 matching lines @@
   // losses when Video NACK is enabled but Audio NACK is not. Clients of NetEq
   // 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.
   int InsertSyncPacket(const WebRtcRTPHeader& rtp_header,
                        uint32_t receive_timestamp) override;
 
-  // 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.
-  // The speech type is written to |type|, if |type| is not NULL.
-  // Returns kOK on success, or kFail in case of an error.
-  int GetAudio(size_t max_length,
-               int16_t* output_audio,
-               size_t* samples_per_channel,
-               size_t* num_channels,
-               NetEqOutputType* type) override;
+  int GetAudio(AudioFrame* audio_frame, NetEqOutputType* type) override;
 
   int RegisterPayloadType(NetEqDecoder codec,
                           const std::string& codec_name,
                           uint8_t rtp_payload_type) override;
 
   int RegisterExternalDecoder(AudioDecoder* decoder,
                               NetEqDecoder codec,
                               const std::string& codec_name,
                               uint8_t rtp_payload_type,
                               int sample_rate_hz) override;
@@ skipping 82 matching lines @@
 
   // Inserts a new packet into NetEq. This is used by the InsertPacket method
   // above. Returns 0 on success, otherwise an error code.
   // TODO(hlundin): Merge this with InsertPacket above?
   int InsertPacketInternal(const WebRtcRTPHeader& rtp_header,
                            rtc::ArrayView<const uint8_t> payload,
                            uint32_t receive_timestamp,
                            bool is_sync_packet)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
-  // Delivers 10 ms of audio data. The data is written to |output|, 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.
+  // Delivers 10 ms of audio data. The data is written to |audio_frame|.
   // Returns 0 on success, otherwise an error code.
-  int GetAudioInternal(size_t max_length,
-                       int16_t* output,
-                       size_t* samples_per_channel,
-                       size_t* num_channels)
+  int GetAudioInternal(AudioFrame* audio_frame)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Provides a decision to the GetAudioInternal method. The decision what to
   // do is written to |operation|. Packets to decode are written to
   // |packet_list|, and a DTMF event to play is written to |dtmf_event|. When
   // DTMF should be played, |play_dtmf| is set to true by the method.
   // Returns 0 on success, otherwise an error code.
   int GetDecision(Operations* operation,
                   PacketList* packet_list,
                   DtmfEvent* dtmf_event,
@@ skipping 164 matching lines @@
   bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
   std::unique_ptr<Nack> nack_ GUARDED_BY(crit_sect_);
   bool nack_enabled_ GUARDED_BY(crit_sect_);
 
  private:
   RTC_DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_