Index: webrtc/modules/audio_processing/test/audio_processing_simulator.h |
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.h b/webrtc/modules/audio_processing/test/audio_processing_simulator.h |
index 1b838d97703440f5ebce58f28a981ec1e868afff..71e63449f52284cb810b0ad5a1c691535617695c 100644 |
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h |
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h |
@@ -23,6 +23,7 @@ |
#include "webrtc/base/timeutils.h" |
#include "webrtc/common_audio/channel_buffer.h" |
#include "webrtc/modules/audio_processing/include/audio_processing.h" |
+#include "webrtc/modules/audio_processing/test/fake_recording_device.h" |
#include "webrtc/modules/audio_processing/test/test_utils.h" |
namespace webrtc { |
@@ -76,6 +77,9 @@ struct SimulationSettings { |
rtc::Optional<int> vad_likelihood; |
rtc::Optional<int> ns_level; |
rtc::Optional<bool> use_refined_adaptive_filter; |
+ int initial_mic_level; |
+ bool simulate_mic_gain = false; |
+ rtc::Optional<int> simulated_mic_kind; |
bool report_performance = false; |
bool report_bitexactness = false; |
bool use_verbose_logging = false; |
@@ -166,6 +170,7 @@ class AudioProcessingSimulator { |
AudioFrame rev_frame_; |
AudioFrame fwd_frame_; |
bool bitexact_output_ = true; |
+ rtc::Optional<int> aec_dump_mic_level_; |
private: |
void SetupOutput(); |
@@ -177,6 +182,7 @@ class AudioProcessingSimulator { |
std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_; |
TickIntervalStats proc_time_; |
std::ofstream residual_echo_likelihood_graph_writer_; |
+ FakeRecordingDevice fake_recording_device_; |
rtc::TaskQueue worker_queue_; |
AleBzk
2017/06/29 11:43:36
Unrelated (merge)
|