Convert channel counts to size_t.

webrtc/voice_engine/channel.cc

 /*
  *  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 401 matching lines @@
 void Channel::OnIncomingCSRCChanged(uint32_t CSRC, bool added) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::OnIncomingCSRCChanged(CSRC=%d, added=%d)", CSRC,
                added);
 }
 
 int32_t Channel::OnInitializeDecoder(
     int8_t payloadType,
     const char payloadName[RTP_PAYLOAD_NAME_SIZE],
     int frequency,
-    uint8_t channels,
+    size_t channels,
     uint32_t rate) {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnInitializeDecoder(payloadType=%d, "
-                 "payloadName=%s, frequency=%u, channels=%u, rate=%u)",
+                 "payloadName=%s, frequency=%u, channels=%" PRIuS ", rate=%u)",
                  payloadType, payloadName, frequency, channels, rate);
 
     CodecInst receiveCodec = {0};
     CodecInst dummyCodec = {0};
 
     receiveCodec.pltype = payloadType;
     receiveCodec.plfreq = frequency;
     receiveCodec.channels = channels;
     receiveCodec.rate = rate;
     strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
@@ skipping 15 matching lines @@
     return 0;
 }
 
 int32_t
 Channel::OnReceivedPayloadData(const uint8_t* payloadData,
                                size_t payloadSize,
                                const WebRtcRTPHeader* rtpHeader)
 {
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::OnReceivedPayloadData(payloadSize=%" PRIuS ","
-                 " payloadType=%u, audioChannel=%u)",
+                 " payloadType=%u, audioChannel=%" PRIuS ")",
                  payloadSize,
                  rtpHeader->header.payloadType,
                  rtpHeader->type.Audio.channel);
 
     if (!channel_state_.Get().playing)
     {
         // Avoid inserting into NetEQ when we are not playing. Count the
         // packet as discarded.
         WEBRTC_TRACE(kTraceStream, kTraceVoice,
                      VoEId(_instanceId, _channelId),
@@ skipping 599 matching lines @@
         if ((audio_coding_->Codec(idx, &codec) == -1) ||
             (rtp_receiver_->RegisterReceivePayload(
                 codec.plname,
                 codec.pltype,
                 codec.plfreq,
                 codec.channels,
                 (codec.rate < 0) ? 0 : codec.rate) == -1))
         {
             WEBRTC_TRACE(kTraceWarning, kTraceVoice,
                          VoEId(_instanceId,_channelId),
-                         "Channel::Init() unable to register %s (%d/%d/%d/%d) "
-                         "to RTP/RTCP receiver",
+                         "Channel::Init() unable to register %s "
+                         "(%d/%d/%" PRIuS "/%d) to RTP/RTCP receiver",
                          codec.plname, codec.pltype, codec.plfreq,
                          codec.channels, codec.rate);
         }
         else
         {
             WEBRTC_TRACE(kTraceInfo, kTraceVoice,
                          VoEId(_instanceId,_channelId),
-                         "Channel::Init() %s (%d/%d/%d/%d) has been added to "
-                         "the RTP/RTCP receiver",
+                         "Channel::Init() %s (%d/%d/%" PRIuS "/%d) has been "
+                         "added to the RTP/RTCP receiver",
                          codec.plname, codec.pltype, codec.plfreq,
                          codec.channels, codec.rate);
         }
 
         // Ensure that PCMU is used as default codec on the sending side
         if (!STR_CASE_CMP(codec.plname, "PCMU") && (codec.channels == 1))
         {
             SetSendCodec(codec);
         }
 
@@ skipping 481 matching lines @@
 }
 
 int32_t
 Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency)
 {
     WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::SetSendCNPayloadType()");
 
     CodecInst codec;
     int32_t samplingFreqHz(-1);
-    const int kMono = 1;
+    const size_t kMono = 1;
     if (frequency == kFreq32000Hz)
         samplingFreqHz = 32000;
     else if (frequency == kFreq16000Hz)
         samplingFreqHz = 16000;
 
     if (audio_coding_->Codec("CN", &codec, samplingFreqHz, kMono) == -1)
     {
         _engineStatisticsPtr->SetLastError(
             VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
             "SetSendCNPayloadType() failed to retrieve default CN codec "
@@ skipping 1738 matching lines @@
     WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
                  "Channel::Demultiplex()");
     _audioFrame.CopyFrom(audioFrame);
     _audioFrame.id_ = _channelId;
     return 0;
 }
 
 void Channel::Demultiplex(const int16_t* audio_data,
                           int sample_rate,
                           size_t number_of_frames,
-                          int number_of_channels) {
+                          size_t number_of_channels) {
   CodecInst codec;
   GetSendCodec(codec);
 
   // Never upsample or upmix the capture signal here. This should be done at the
   // end of the send chain.
   _audioFrame.sample_rate_hz_ = std::min(codec.plfreq, sample_rate);
   _audioFrame.num_channels_ = std::min(number_of_channels, codec.channels);
   RemixAndResample(audio_data, number_of_frames, number_of_channels,
                    sample_rate, &input_resampler_, &_audioFrame);
 }
@@ skipping 466 matching lines @@
                        VoEId(_instanceId, _channelId),
                        "Channel::EncodeAndSend() inserting Dtmf failed");
             return -1;
         }
 
         // Replace mixed audio with DTMF tone.
         for (size_t sample = 0;
             sample < _audioFrame.samples_per_channel_;
             sample++)
         {
-            for (int channel = 0;
+            for (size_t channel = 0;
                 channel < _audioFrame.num_channels_;
                 channel++)
             {
                 const size_t index =
                     sample * _audioFrame.num_channels_ + channel;
                 _audioFrame.data_[index] = toneBuffer[sample];
             }
         }
 
         assert(_audioFrame.samples_per_channel_ == toneSamples);
@@ skipping 113 matching lines @@
                 codec.plname,
                 codec.pltype,
                 codec.plfreq,
                 codec.channels,
                 (codec.rate < 0) ? 0 : codec.rate) == -1))
         {
             WEBRTC_TRACE(kTraceWarning,
                          kTraceVoice,
                          VoEId(_instanceId, _channelId),
                          "Channel::RegisterReceiveCodecsToRTPModule() unable"
-                         " to register %s (%d/%d/%d/%d) to RTP/RTCP receiver",
+                         " to register %s (%d/%d/%" PRIuS "/%d) to RTP/RTCP "
+                         "receiver",
                          codec.plname, codec.pltype, codec.plfreq,
                          codec.channels, codec.rate);
         }
         else
         {
             WEBRTC_TRACE(kTraceInfo,
                          kTraceVoice,
                          VoEId(_instanceId, _channelId),
                          "Channel::RegisterReceiveCodecsToRTPModule() %s "
-                         "(%d/%d/%d/%d) has been added to the RTP/RTCP "
+                         "(%d/%d/%" PRIuS "/%d) has been added to the RTP/RTCP "
                          "receiver",
                          codec.plname, codec.pltype, codec.plfreq,
                          codec.channels, codec.rate);
         }
     }
 }
 
 // Assuming this method is called with valid payload type.
 int Channel::SetRedPayloadType(int red_payload_type) {
   CodecInst codec;
@@ skipping 108 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt)
       != 0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc