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Convert channel counts to size_t.

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IIRC, this was originally requested by ajm during review of the other size_t conversions I did over the past year, and I agreed it made sense, but wanted to do it separately since those changes were already gargantuan.

BUG=chromium:81439
TEST=none
R=henrik.lundin@webrtc.org, henrika@webrtc.org, kjellander@webrtc.org, minyue@webrtc.org, perkj@webrtc.org, solenberg@webrtc.org, stefan@webrtc.org, tina.legrand@webrtc.org

Review URL: https://codereview.webrtc.org/1316523002 .

Cr-Commit-Position: refs/heads/master@{#11229}
This commit is contained in:
Peter Kasting
2016-01-12 16:26:35 -08:00
parent 92e677a1f8
commit 6955870806
190 changed files with 892 additions and 868 deletions
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39
webrtc/modules/audio_processing/audio_processing_impl.cc
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@ -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(