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Unified Diff: webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.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/codecs/tools/audio_codec_speed_test.cc
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
index c7cafdff9ba8084f3e00eb0565c7e599c8ea7956..3395721f8ba63f65bbc911743fbf772b58d682ae 100644
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
+++ b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.cc
@@ -65,7 +65,8 @@ void AudioCodecSpeedTest::SetUp() {
memcpy(&in_data_[loop_length_samples_], &in_data_[0],
input_length_sample_ * channels_ * sizeof(int16_t));
- max_bytes_ = input_length_sample_ * channels_ * sizeof(int16_t);
+ max_bytes_ =
+ static_cast<size_t>(input_length_sample_ * channels_ * sizeof(int16_t));
out_data_.reset(new int16_t[output_length_sample_ * channels_]);
bit_stream_.reset(new uint8_t[max_bytes_]);

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