Convert channel counts to size_t.

webrtc/modules/audio_processing/audio_processing_impl.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 392 matching lines @@
       return err;
     }
   }
 #endif
 
   return kNoError;
 }
 
 int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) {
   for (const auto& stream : config.streams) {
-    if (stream.num_channels() < 0) {
-      return kBadNumberChannelsError;
-    }
     if (stream.num_channels() > 0 && stream.sample_rate_hz() <= 0) {
       return kBadSampleRateError;
     }
   }
 
-  const int num_in_channels = config.input_stream().num_channels();
-  const int num_out_channels = config.output_stream().num_channels();
+  const size_t num_in_channels = config.input_stream().num_channels();
+  const size_t num_out_channels = config.output_stream().num_channels();
 
   // Need at least one input channel.
   // Need either one output channel or as many outputs as there are inputs.
   if (num_in_channels == 0 ||
       !(num_out_channels == 1 || num_out_channels == num_in_channels)) {
     return kBadNumberChannelsError;
   }
 
   if (capture_nonlocked_.beamformer_enabled &&
-      static_cast<size_t>(num_in_channels) != capture_.array_geometry.size()) {
+      num_in_channels != capture_.array_geometry.size()) {
     return kBadNumberChannelsError;
   }
 
   formats_.api_format = config;
 
   // We process at the closest native rate >= min(input rate, output rate)...
   const int min_proc_rate =
       std::min(formats_.api_format.input_stream().sample_rate_hz(),
                formats_.api_format.output_stream().sample_rate_hz());
   int fwd_proc_rate;
@@ skipping 77 matching lines @@
 int AudioProcessingImpl::proc_sample_rate_hz() const {
   // Used as callback from submodules, hence locking is not allowed.
   return capture_nonlocked_.fwd_proc_format.sample_rate_hz();
 }
 
 int AudioProcessingImpl::proc_split_sample_rate_hz() const {
   // Used as callback from submodules, hence locking is not allowed.
   return capture_nonlocked_.split_rate;
 }
 
-int AudioProcessingImpl::num_reverse_channels() const {
+size_t AudioProcessingImpl::num_reverse_channels() const {
   // Used as callback from submodules, hence locking is not allowed.
   return formats_.rev_proc_format.num_channels();
 }
 
-int AudioProcessingImpl::num_input_channels() const {
+size_t AudioProcessingImpl::num_input_channels() const {
   // Used as callback from submodules, hence locking is not allowed.
   return formats_.api_format.input_stream().num_channels();
 }
 
-int AudioProcessingImpl::num_proc_channels() const {
+size_t AudioProcessingImpl::num_proc_channels() const {
   // Used as callback from submodules, hence locking is not allowed.
   return capture_nonlocked_.beamformer_enabled ? 1 : num_output_channels();
 }
 
-int AudioProcessingImpl::num_output_channels() const {
+size_t AudioProcessingImpl::num_output_channels() const {
   // Used as callback from submodules, hence locking is not allowed.
   return formats_.api_format.output_stream().num_channels();
 }
 
 void AudioProcessingImpl::set_output_will_be_muted(bool muted) {
   rtc::CritScope cs(&crit_capture_);
   capture_.output_will_be_muted = muted;
   if (private_submodules_->agc_manager.get()) {
     private_submodules_->agc_manager->SetCaptureMuted(
         capture_.output_will_be_muted);
@@ skipping 68 matching lines @@
          formats_.api_format.input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->Open()) {
     RETURN_ON_ERR(WriteConfigMessage(false));
 
     debug_dump_.capture.event_msg->set_type(audioproc::Event::STREAM);
     audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t channel_size =
         sizeof(float) * formats_.api_format.input_stream().num_frames();
-    for (int i = 0; i < formats_.api_format.input_stream().num_channels(); ++i)
+    for (size_t i = 0; i < formats_.api_format.input_stream().num_channels();
+         ++i)
       msg->add_input_channel(src[i], channel_size);
   }
 #endif
 
   capture_.capture_audio->CopyFrom(src, formats_.api_format.input_stream());
   RETURN_ON_ERR(ProcessStreamLocked());
   capture_.capture_audio->CopyTo(formats_.api_format.output_stream(), dest);
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->Open()) {
     audioproc::Stream* msg = debug_dump_.capture.event_msg->mutable_stream();
     const size_t channel_size =
         sizeof(float) * formats_.api_format.output_stream().num_frames();
-    for (int i = 0; i < formats_.api_format.output_stream().num_channels(); ++i)
+    for (size_t i = 0; i < formats_.api_format.output_stream().num_channels();
+         ++i)
       msg->add_output_channel(dest[i], channel_size);
     RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                           &crit_debug_, &debug_dump_.capture));
   }
 #endif
 
   return kNoError;
 }
 
 int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
@@ skipping 213 matching lines @@
 }
 
 int AudioProcessingImpl::AnalyzeReverseStreamLocked(
     const float* const* src,
     const StreamConfig& reverse_input_config,
     const StreamConfig& reverse_output_config) {
   if (src == nullptr) {
     return kNullPointerError;
   }
 
-  if (reverse_input_config.num_channels() <= 0) {
+  if (reverse_input_config.num_channels() == 0) {
     return kBadNumberChannelsError;
   }
 
   ProcessingConfig processing_config = formats_.api_format;
   processing_config.reverse_input_stream() = reverse_input_config;
   processing_config.reverse_output_stream() = reverse_output_config;
 
   RETURN_ON_ERR(MaybeInitializeRender(processing_config));
   assert(reverse_input_config.num_frames() ==
          formats_.api_format.reverse_input_stream().num_frames());
 
 #ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
   if (debug_dump_.debug_file->Open()) {
     debug_dump_.render.event_msg->set_type(audioproc::Event::REVERSE_STREAM);
     audioproc::ReverseStream* msg =
         debug_dump_.render.event_msg->mutable_reverse_stream();
     const size_t channel_size =
         sizeof(float) * formats_.api_format.reverse_input_stream().num_frames();
-    for (int i = 0;
+    for (size_t i = 0;
          i < formats_.api_format.reverse_input_stream().num_channels(); ++i)
       msg->add_channel(src[i], channel_size);
     RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                           &crit_debug_, &debug_dump_.render));
   }
 #endif
 
   render_.render_audio->CopyFrom(src,
                                  formats_.api_format.reverse_input_stream());
   return ProcessReverseStreamLocked();
@@ skipping 536 matching lines @@
   debug_state->event_msg->Clear();
 
   return kNoError;
 }
 
 int AudioProcessingImpl::WriteInitMessage() {
   debug_dump_.capture.event_msg->set_type(audioproc::Event::INIT);
   audioproc::Init* msg = debug_dump_.capture.event_msg->mutable_init();
   msg->set_sample_rate(formats_.api_format.input_stream().sample_rate_hz());
 
-  msg->set_num_input_channels(
-      formats_.api_format.input_stream().num_channels());
-  msg->set_num_output_channels(
-      formats_.api_format.output_stream().num_channels());
-  msg->set_num_reverse_channels(
-      formats_.api_format.reverse_input_stream().num_channels());
+  msg->set_num_input_channels(static_cast<google::protobuf::int32>(
+      formats_.api_format.input_stream().num_channels()));
+  msg->set_num_output_channels(static_cast<google::protobuf::int32>(
+      formats_.api_format.output_stream().num_channels()));
+  msg->set_num_reverse_channels(static_cast<google::protobuf::int32>(
+      formats_.api_format.reverse_input_stream().num_channels()));
   msg->set_reverse_sample_rate(
       formats_.api_format.reverse_input_stream().sample_rate_hz());
   msg->set_output_sample_rate(
       formats_.api_format.output_stream().sample_rate_hz());
   // TODO(ekmeyerson): Add reverse output fields to
   // debug_dump_.capture.event_msg.
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
@@ skipping 46 matching lines @@
   debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG);
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc