webrtc/modules/audio_processing/audio_processing_impl.cc - Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce

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Unified Diff: webrtc/modules/audio_processing/audio_processing_impl.cc

Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master

Patch Set: Resync Created 5 years, 4 months ago

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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 c9e4ddc10c5f2aa7a330dc283ea5cd114161075f..ff4128b6ed4485df0d5abef47a2ff006da53a6b5 100644

--- a/webrtc/modules/audio_processing/audio_processing_impl.cc

+++ b/webrtc/modules/audio_processing/audio_processing_impl.cc

@@ -510,7 +510,7 @@ bool AudioProcessingImpl::output_will_be_muted() const {

}

int AudioProcessingImpl::ProcessStream(const float* const* src,

- int samples_per_channel,

+ size_t samples_per_channel,

int input_sample_rate_hz,

ChannelLayout input_layout,

int output_sample_rate_hz,

@@ -716,7 +716,7 @@ int AudioProcessingImpl::ProcessStreamLocked() {

}

int AudioProcessingImpl::AnalyzeReverseStream(const float* const* data,

- int samples_per_channel,

+ size_t samples_per_channel,

int rev_sample_rate_hz,

ChannelLayout layout) {

const StreamConfig reverse_config = {