webrtc/modules/audio_coding/main/acm2/acm_send_test.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: webrtc/modules/audio_coding/main/acm2/acm_send_test.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: webrtc/modules/audio_coding/main/acm2/acm_send_test.h

diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h b/webrtc/modules/audio_coding/main/acm2/acm_send_test.h

index 4c4db5bd13b32b181748dd1e5c591ed5475306ae..09fe9e665567ea9a74346e5c517129623c4af72e 100644

--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test.h

+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test.h

@@ -63,7 +63,7 @@ class AcmSendTest : public AudioPacketizationCallback, public PacketSource {

rtc::scoped_ptr<AudioCoding> acm_;

InputAudioFile* audio_source_;

int source_rate_hz_;

- const int input_block_size_samples_;

+ const size_t input_block_size_samples_;

AudioFrame input_frame_;

bool codec_registered_;

int test_duration_ms_;

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