talk/media/webrtc/fakewebrtcvoiceengine.h - Issue 1224093003: Update audio code to use size_t more correctly, talk/ portion.

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

Issue 1224093003: Update audio code to use size_t more correctly, talk/ portion. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master

Patch Set: Created 5 years, 5 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 419170b24dc471bdb2c1a8c2ec9ee7fe6fde6a5c..6c1df3225cfe00883bb238c038d0fbaaab5b4be2 100644

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

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

@@ -130,7 +130,7 @@ 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,

@@ -139,7 +139,7 @@ class FakeAudioProcessing : public webrtc::AudioProcessing {

WEBRTC_STUB(AnalyzeReverseStream, (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(set_stream_delay_ms, (int delay));

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