Index: webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc |
diff --git a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc |
index d1cd48424a1cfcb3708c170f2a1a91d866f35887..45e32f097402d23f19466ceb92adf9117eb1be9d 100644 |
--- a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc |
+++ b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc |
@@ -8,11 +8,15 @@ |
* be found in the AUTHORS file in the root of the source tree. |
*/ |
+#include <algorithm> |
#include <iostream> |
+#include <utility> |
#include "webrtc/modules/audio_processing/test/aec_dump_based_simulator.h" |
#include "webrtc/base/checks.h" |
+#include "webrtc/base/logging.h" |
+#include "webrtc/modules/audio_processing/test/fake_recording_device.h" |
#include "webrtc/modules/audio_processing/test/protobuf_utils.h" |
#include "webrtc/test/testsupport/trace_to_stderr.h" |
@@ -63,13 +67,17 @@ bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg, |
} // namespace |
AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings) |
- : AudioProcessingSimulator(settings) {} |
+ : AudioProcessingSimulator(settings) { |
+ if (settings_.simulate_mic_gain) { |
+ LOG(LS_VERBOSE) << "Simulating analog mic gain using AEC dump as input " |
+ << "(the unmodified mic gain level will be virtually restored)"; |
+ } |
+} |
AecDumpBasedSimulator::~AecDumpBasedSimulator() = default; |
void AecDumpBasedSimulator::PrepareProcessStreamCall( |
- const webrtc::audioproc::Stream& msg, |
- bool* set_stream_analog_level_called) { |
+ const webrtc::audioproc::Stream& msg) { |
if (msg.has_input_data()) { |
// Fixed interface processing. |
// Verify interface invariance. |
@@ -156,14 +164,20 @@ void AecDumpBasedSimulator::PrepareProcessStreamCall( |
ap_->set_stream_key_pressed(*settings_.use_ts); |
} |
- // TODO(peah): Add support for controlling the analog level via the |
- // command-line. |
- if (msg.has_level()) { |
- RTC_CHECK_EQ(AudioProcessing::kNoError, |
- ap_->gain_control()->set_stream_analog_level(msg.level())); |
- *set_stream_analog_level_called = true; |
+ // Level is always logged in AEC dumps. |
+ RTC_CHECK(msg.has_level()); |
+ |
+ RTC_DCHECK(fake_recording_device_); |
+ if (settings_.simulate_mic_gain) { |
+ // When the analog gain is simulated, set the undo level to |msg.level()| to |
+ // virtually restore the unmodified microphone signal level. |
+ fake_recording_device_->set_undo_mic_level(rtc::Optional<int>(msg.level())); |
} else { |
- *set_stream_analog_level_called = false; |
+ // When the analog gain is not simulated, the AEC dump level has to be used |
+ // in AudioProcessingSimulator::ProcessStream() - i.e., overriding any value |
+ // set from a gain controller once the previous audio frame has been |
+ // analyzed. |
+ fake_recording_device_->set_mic_level(msg.level()); |
AleBzk
2017/05/23 13:56:41
@Per: you asked to move this to the parent class a
peah-webrtc
2017/05/23 22:13:20
What I don't like with this approach is that it gi
AleBzk
2017/06/22 10:16:00
Done.
|
} |
} |
@@ -562,14 +576,8 @@ void AecDumpBasedSimulator::HandleMessage(const webrtc::audioproc::Init& msg) { |
void AecDumpBasedSimulator::HandleMessage( |
const webrtc::audioproc::Stream& msg) { |
- bool set_stream_analog_level_called = false; |
- PrepareProcessStreamCall(msg, &set_stream_analog_level_called); |
+ PrepareProcessStreamCall(msg); |
ProcessStream(interface_used_ == InterfaceType::kFixedInterface); |
- if (set_stream_analog_level_called) { |
- // Call stream analog level to ensure that any side-effects are triggered. |
- (void)ap_->gain_control()->stream_analog_level(); |
- } |
- |
VerifyProcessStreamBitExactness(msg); |
} |