webrtc/modules/audio_device/fine_audio_buffer_unittest.cc - Issue 2715963002: Simplifies FineAudioBuffer by using rtc::Buffer

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Unified Diff: webrtc/modules/audio_device/fine_audio_buffer_unittest.cc

Issue 2715963002: Simplifies FineAudioBuffer by using rtc::Buffer (Closed)

Patch Set: Improved buffer handling after feedback from kwiberg@ Created 3 years, 10 months ago

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Index: webrtc/modules/audio_device/fine_audio_buffer_unittest.cc

diff --git a/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc b/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc

index 18b203521cde1b32bd3fa6f42099238c04ab49b2..535f16816cb4998230d6394d14b2af9c88007ea6 100644

--- a/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc

+++ b/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc

@@ -118,7 +118,7 @@ void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {

sample_rate);

std::unique_ptr<int8_t[]> out_buffer;

- out_buffer.reset(new int8_t[fine_buffer.RequiredPlayoutBufferSizeBytes()]);

+ out_buffer.reset(new int8_t[kFrameSizeBytes]);

std::unique_ptr<int8_t[]> in_buffer;

in_buffer.reset(new int8_t[kFrameSizeBytes]);

for (int i = 0; i < kNumberOfFrames; ++i) {