talk/media/webrtc/webrtcvoiceengine.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: talk/media/webrtc/webrtcvoiceengine.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: talk/media/webrtc/webrtcvoiceengine.cc

diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc

index e288e7040fac5b450d307986c37c389b2d279c5b..80208ba0c7617b5a895ead759b4276c995e8f6cc 100644

--- a/talk/media/webrtc/webrtcvoiceengine.cc

+++ b/talk/media/webrtc/webrtcvoiceengine.cc

@@ -1569,7 +1569,7 @@ bool WebRtcVoiceEngine::UnregisterProcessor(

void WebRtcVoiceEngine::Process(int channel,

webrtc::ProcessingTypes type,

int16_t audio10ms[],

- int length,

+ size_t length,

int sampling_freq,

bool is_stereo) {

rtc::CritScope cs(&signal_media_critical_);

@@ -1665,7 +1665,7 @@ class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer

int bits_per_sample,

int sample_rate,

int number_of_channels,

- int number_of_frames) override {

+ size_t number_of_frames) override {

voe_audio_transport_->OnData(channel_,

audio_data,

bits_per_sample,

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