| 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 65d7136de1aeeae05fd76b0136daa87fa3bc03b8..c640a74b8c57066b53978376a11a5bb48f903544 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());
|
| }
|
|
|