| Index: webrtc/modules/audio_processing/test/audioproc_float.cc
|
| diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
|
| index d2983b2c56d2f6d99e21449f90a0ad70dd949f4f..dac43629cf65d69151f34b83f3b1a4f8e0c1aa79 100644
|
| --- a/webrtc/modules/audio_processing/test/audioproc_float.cc
|
| +++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
|
| @@ -127,13 +127,6 @@
|
| TickTime processing_start_time;
|
| TickInterval accumulated_time;
|
| int num_chunks = 0;
|
| -
|
| - const StreamConfig input_config = {
|
| - in_file.sample_rate(), in_buf.num_channels(),
|
| - };
|
| - const StreamConfig output_config = {
|
| - out_file.sample_rate(), out_buf.num_channels(),
|
| - };
|
| while (in_file.ReadSamples(in_interleaved.size(),
|
| &in_interleaved[0]) == in_interleaved.size()) {
|
| // Have logs display the file time rather than wallclock time.
|
| @@ -146,8 +139,14 @@
|
| if (FLAGS_perf) {
|
| processing_start_time = TickTime::Now();
|
| }
|
| - CHECK_EQ(kNoErr, ap->ProcessStream(in_buf.channels(), input_config,
|
| - output_config, out_buf.channels()));
|
| + CHECK_EQ(kNoErr,
|
| + ap->ProcessStream(in_buf.channels(),
|
| + in_buf.num_frames(),
|
| + in_file.sample_rate(),
|
| + LayoutFromChannels(in_buf.num_channels()),
|
| + out_file.sample_rate(),
|
| + LayoutFromChannels(out_buf.num_channels()),
|
| + out_buf.channels()));
|
| if (FLAGS_perf) {
|
| accumulated_time += TickTime::Now() - processing_start_time;
|
| }
|
|
|