webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.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_coding/main/acm2/acm_send_test_oldapi.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_coding/main/acm2/acm_send_test_oldapi.cc

diff --git a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc

index 1819d59d96d305fbbd3c0e7d024d74d62fd19ded..74e98d9cd57b02d9957b247e8934c6fab654656f 100644

--- a/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc

+++ b/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc

@@ -31,7 +31,8 @@ AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,

acm_(webrtc::AudioCodingModule::Create(0, &clock_)),

audio_source_(audio_source),

source_rate_hz_(source_rate_hz),

- input_block_size_samples_(source_rate_hz_ * kBlockSizeMs / 1000),

+ input_block_size_samples_(

+ static_cast<size_t>(source_rate_hz_ * kBlockSizeMs / 1000)),

codec_registered_(false),

test_duration_ms_(test_duration_ms),

frame_type_(kAudioFrameSpeech),