Index: webrtc/common_audio/audio_ring_buffer_unittest.cc |
diff --git a/webrtc/common_audio/audio_ring_buffer_unittest.cc b/webrtc/common_audio/audio_ring_buffer_unittest.cc |
index a83c87599c414c127e95881e88873d4100ff237b..a7a6a9442bca8d0f83641b7652b71a58a83e6950 100644 |
--- a/webrtc/common_audio/audio_ring_buffer_unittest.cc |
+++ b/webrtc/common_audio/audio_ring_buffer_unittest.cc |
@@ -34,27 +34,27 @@ void ReadAndWriteTest(const ChannelBuffer<float>& input, |
while (input_pos + buf.WriteFramesAvailable() < total_frames) { |
// Write until the buffer is as full as possible. |
while (buf.WriteFramesAvailable() >= num_write_chunk_frames) { |
- buf.Write(input.Slice(slice.get(), static_cast<int>(input_pos)), |
- num_channels, num_write_chunk_frames); |
+ buf.Write(input.Slice(slice.get(), input_pos), num_channels, |
+ num_write_chunk_frames); |
input_pos += num_write_chunk_frames; |
} |
// Read until the buffer is as empty as possible. |
while (buf.ReadFramesAvailable() >= num_read_chunk_frames) { |
EXPECT_LT(output_pos, total_frames); |
- buf.Read(output->Slice(slice.get(), static_cast<int>(output_pos)), |
- num_channels, num_read_chunk_frames); |
+ buf.Read(output->Slice(slice.get(), output_pos), num_channels, |
+ num_read_chunk_frames); |
output_pos += num_read_chunk_frames; |
} |
} |
// Write and read the last bit. |
if (input_pos < total_frames) { |
- buf.Write(input.Slice(slice.get(), static_cast<int>(input_pos)), |
- num_channels, total_frames - input_pos); |
+ buf.Write(input.Slice(slice.get(), input_pos), num_channels, |
+ total_frames - input_pos); |
} |
if (buf.ReadFramesAvailable()) { |
- buf.Read(output->Slice(slice.get(), static_cast<int>(output_pos)), |
- num_channels, buf.ReadFramesAvailable()); |
+ buf.Read(output->Slice(slice.get(), output_pos), num_channels, |
+ buf.ReadFramesAvailable()); |
} |
EXPECT_EQ(0u, buf.ReadFramesAvailable()); |
} |