talk/media/webrtc/fakewebrtcvoiceengine.h - Issue 1230503003: Update a ton of audio code to use size_t more correctly and in general reduce

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Unified Diff: talk/media/webrtc/fakewebrtcvoiceengine.h

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: talk/media/webrtc/fakewebrtcvoiceengine.h

diff --git a/talk/media/webrtc/fakewebrtcvoiceengine.h b/talk/media/webrtc/fakewebrtcvoiceengine.h

index 3ac2f3bff3808657594d85323534001cefb96d2a..cea9e4026b9cb5a071bf981cfa5a1ea17b6ab33c 100644

--- a/talk/media/webrtc/fakewebrtcvoiceengine.h

+++ b/talk/media/webrtc/fakewebrtcvoiceengine.h

@@ -132,7 +132,7 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {

WEBRTC_STUB(ProcessStream, (webrtc::AudioFrame* frame));

WEBRTC_STUB(ProcessStream, (

const float* const* src,

- int samples_per_channel,

+ size_t samples_per_channel,

int input_sample_rate_hz,

webrtc::AudioProcessing::ChannelLayout input_layout,

int output_sample_rate_hz,

@@ -147,7 +147,7 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {

WEBRTC_STUB(ProcessReverseStream, (webrtc::AudioFrame * frame));

WEBRTC_STUB(AnalyzeReverseStream, (

const float* const* data,

- int samples_per_channel,

+ size_t samples_per_channel,

int sample_rate_hz,

webrtc::AudioProcessing::ChannelLayout layout));

WEBRTC_STUB(ProcessReverseStream,

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