| 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 a91480d2655233e4ed0c3f239ea5e9eff52c7280..ce78eed1e526a0318f81208555f1f77213f50bf0 100644
|
| --- a/webrtc/modules/audio_processing/test/audioproc_float.cc
|
| +++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
|
| @@ -165,6 +165,7 @@ DEFINE_bool(discard_settings_in_aecdump,
|
| DEFINE_bool(store_intermediate_output,
|
| false,
|
| "Creates new output files after each init");
|
| +DEFINE_string(custom_call_order_file, "", "Custom process API call order file");
|
|
|
| void SetSettingIfSpecified(const std::string value,
|
| rtc::Optional<std::string>* parameter) {
|
| @@ -254,6 +255,8 @@ SimulationSettings CreateSettings() {
|
| SetSettingIfSpecified(FLAGS_stream_delay, &settings.stream_delay);
|
| SetSettingIfSpecified(FLAGS_stream_drift_samples,
|
| &settings.stream_drift_samples);
|
| + SetSettingIfSpecified(FLAGS_custom_call_order_file,
|
| + &settings.custom_call_order_filename);
|
| settings.report_performance = FLAGS_performance_report;
|
| settings.use_verbose_logging = FLAGS_verbose;
|
| settings.report_bitexactness = FLAGS_bitexactness_report;
|
| @@ -361,6 +364,11 @@ void PerformBasicParameterSanityChecks(const SimulationSettings& settings) {
|
| "Error: --bitexactness_report can only be used when operating on an "
|
| "aecdump\n");
|
|
|
| + ReportConditionalErrorAndExit(
|
| + settings.custom_call_order_filename && settings.aec_dump_input_filename,
|
| + "Error: --custom_call_order_file cannot be used when operating on an "
|
| + "aecdump\n");
|
| +
|
| auto valid_wav_name = [](const std::string& wav_file_name) {
|
| if (wav_file_name.size() < 5) {
|
| return false;
|
|
|