Update a ton of audio code to use size_t more correctly and in general reduce

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 88 matching lines @@
   // 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,
-               int* samples_per_channel,
+               size_t* samples_per_channel,
                int* num_channels,
                NetEqOutputType* type) override;
 
   // Associates |rtp_payload_type| with |codec| and stores the information in
   // the codec database. Returns kOK on success, kFail on failure.
   int RegisterPayloadType(enum NetEqDecoder codec,
                           uint8_t rtp_payload_type) override;
 
   // Provides an externally created decoder object |decoder| to insert in the
   // decoder database. The decoder implements a decoder of type |codec| and
@@ skipping 76 matching lines @@
 
   // Get sequence number and timestamp of the latest RTP.
   // This method is to facilitate NACK.
   int DecodedRtpInfo(int* sequence_number, uint32_t* timestamp) const override;
 
   // This accessor method is only intended for testing purposes.
   const SyncBuffer* sync_buffer_for_test() const;
 
  protected:
   static const int kOutputSizeMs = 10;
-  static const int kMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
+  static const size_t kMaxFrameSize = 2880;  // 60 ms @ 48 kHz.
   // TODO(hlundin): Provide a better value for kSyncBufferSize.
-  static const int kSyncBufferSize = 2 * kMaxFrameSize;
+  static const size_t kSyncBufferSize = 2 * kMaxFrameSize;
 
   // 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,
                            const uint8_t* payload,
                            size_t length_bytes,
                            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.
   // Returns 0 on success, otherwise an error code.
   int GetAudioInternal(size_t max_length,
                        int16_t* output,
-                       int* samples_per_channel,
+                       size_t* samples_per_channel,
                        int* 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,
@@ skipping 72 matching lines @@
 
   // Overdub DTMF on top of |output|.
   int DtmfOverdub(const DtmfEvent& dtmf_event,
                   size_t num_channels,
                   int16_t* output) const EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Extracts packets from |packet_buffer_| to produce at least
   // |required_samples| samples. The packets are inserted into |packet_list|.
   // Returns the number of samples that the packets in the list will produce, or
   // -1 in case of an error.
-  int ExtractPackets(int required_samples, PacketList* packet_list)
+  int ExtractPackets(size_t required_samples, PacketList* packet_list)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Resets various variables and objects to new values based on the sample rate
   // |fs_hz| and |channels| number audio channels.
   void SetSampleRateAndChannels(int fs_hz, size_t channels)
       EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
 
   // Returns the output type for the audio produced by the latest call to
   // GetAudio().
   NetEqOutputType LastOutputType() EXCLUSIVE_LOCKS_REQUIRED(crit_sect_);
@@ skipping 36 matching lines @@
   rtc::scoped_ptr<Normal> normal_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<Merge> merge_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<Accelerate> accelerate_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<PreemptiveExpand> preemptive_expand_ GUARDED_BY(crit_sect_);
   RandomVector random_vector_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<ComfortNoise> comfort_noise_ GUARDED_BY(crit_sect_);
   Rtcp rtcp_ GUARDED_BY(crit_sect_);
   StatisticsCalculator stats_ GUARDED_BY(crit_sect_);
   int fs_hz_ GUARDED_BY(crit_sect_);
   int fs_mult_ GUARDED_BY(crit_sect_);
-  int output_size_samples_ GUARDED_BY(crit_sect_);
-  int decoder_frame_length_ GUARDED_BY(crit_sect_);
+  size_t output_size_samples_ GUARDED_BY(crit_sect_);
+  size_t decoder_frame_length_ GUARDED_BY(crit_sect_);
   Modes last_mode_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<int16_t[]> mute_factor_array_ GUARDED_BY(crit_sect_);
   size_t decoded_buffer_length_ GUARDED_BY(crit_sect_);
   rtc::scoped_ptr<int16_t[]> decoded_buffer_ GUARDED_BY(crit_sect_);
   uint32_t playout_timestamp_ GUARDED_BY(crit_sect_);
   bool new_codec_ GUARDED_BY(crit_sect_);
   uint32_t timestamp_ GUARDED_BY(crit_sect_);
   bool reset_decoder_ GUARDED_BY(crit_sect_);
   uint8_t current_rtp_payload_type_ GUARDED_BY(crit_sect_);
   uint8_t current_cng_rtp_payload_type_ GUARDED_BY(crit_sect_);
@@ skipping 14 matching lines @@
   // module is designed to compensate for this.
   int decoded_packet_sequence_number_ GUARDED_BY(crit_sect_);
   uint32_t decoded_packet_timestamp_ GUARDED_BY(crit_sect_);
 
  private:
   DISALLOW_COPY_AND_ASSIGN(NetEqImpl);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_AUDIO_CODING_NETEQ_NETEQ_IMPL_H_