Convert channel counts to size_t.

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 89 matching lines @@
   // |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,
-               int* num_channels,
+               size_t* num_channels,
                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,
@@ skipping 92 matching lines @@
 
   // 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.
   // Returns 0 on success, otherwise an error code.
   int GetAudioInternal(size_t max_length,
                        int16_t* output,
                        size_t* samples_per_channel,
-                       int* num_channels) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
+                       size_t* num_channels)
+      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,
                   bool* play_dtmf) EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
@@ skipping 163 matching lines @@
   bool enable_fast_accelerate_ GUARDED_BY(crit_sect_);
   rtc::scoped_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_