Index: webrtc/modules/audio_processing/audio_processing_impl.cc |
diff --git a/webrtc/modules/audio_processing/audio_processing_impl.cc b/webrtc/modules/audio_processing/audio_processing_impl.cc |
index fea57856df1e048ebb7b655d760026caaaa38ae2..744309c7745b44aafecb7737b4b06845a45f6f31 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc |
@@ -410,16 +410,13 @@ int AudioProcessingImpl::InitializeLocked() { |
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. |
@@ -429,7 +426,7 @@ int AudioProcessingImpl::InitializeLocked(const ProcessingConfig& config) { |
} |
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; |
} |
@@ -527,22 +524,22 @@ int AudioProcessingImpl::proc_split_sample_rate_hz() const { |
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(); |
} |
@@ -631,7 +628,8 @@ int AudioProcessingImpl::ProcessStream(const float* const* src, |
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 |
@@ -645,7 +643,8 @@ int AudioProcessingImpl::ProcessStream(const float* const* src, |
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)); |
@@ -879,7 +878,7 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked( |
return kNullPointerError; |
} |
- if (reverse_input_config.num_channels() <= 0) { |
+ if (reverse_input_config.num_channels() == 0) { |
return kBadNumberChannelsError; |
} |
@@ -898,7 +897,7 @@ int AudioProcessingImpl::AnalyzeReverseStreamLocked( |
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(), |
@@ -1455,12 +1454,12 @@ int AudioProcessingImpl::WriteInitMessage() { |
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( |