Chromium Code Reviews| 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..e7be0fa7a5c80417c315abd0c1253c819b84b575 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 +8,14 @@ |
| * 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/protobuf_utils.h" |
| #include "webrtc/test/testsupport/trace_to_stderr.h" |
| @@ -68,8 +71,7 @@ AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings) |
| 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. |
| @@ -158,12 +160,24 @@ void AecDumpBasedSimulator::PrepareProcessStreamCall( |
| // TODO(peah): Add support for controlling the analog level via the |
|
peah-webrtc
2017/05/05 06:28:41
You can remove this TODO in this CL :-)
AleBzk
2017/05/05 12:20:17
Done.
|
| // 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; |
| - } else { |
| - *set_stream_analog_level_called = false; |
| + |
| + // Level is always logged in AEC dumps. |
| + RTC_CHECK(msg.has_level()); |
| + |
| + // When the analog gain is simulated, set the undo level to |msg.level()| to |
| + // virtually restore the unmodified microphone signal level. |
| + if (settings_.simulate_mic_gain) { |
| + fake_recording_device_.NotifyAudioDeviceLevel(msg.level()); |
|
peah-webrtc
2017/05/05 06:28:41
I'd prefer all the handling of the fake_recording_
AleBzk
2017/05/05 12:20:17
Right. I made fake_recording_device_ private inste
|
| + LOG(LS_VERBOSE) << "setting mic gain undo level from AEC dump to " |
|
peah-webrtc
2017/05/05 06:28:41
setting->Setting?
AleBzk
2017/05/05 12:20:17
Done.
|
| + << msg.level(); |
|
aleloi
2017/05/04 12:47:13
This would get logged once for every audio frame.
peah-webrtc
2017/05/05 06:28:41
Yes, I agree. The logging here is only in test cod
AleBzk
2017/05/05 12:20:17
I moved this in the ctor so it's shown once.
|
| + } |
| + |
| + // When the analog gain is not simulated, the AEC dump level has to be used in |
| + // AudioProcessingSimulator::ProcessStream(). |
| + if (!settings_.simulate_mic_gain) { |
|
aleloi
2017/05/04 12:47:13
Suggest merge into the if statement above as an el
AleBzk
2017/05/05 12:20:17
Done.
AleBzk
2017/05/05 12:20:17
Done.
|
| + fake_recording_device_.set_analog_level(msg.level()); |
| + LOG(LS_VERBOSE) << "AEC dump overriding AGC suggested level to " |
| + << msg.level(); |
| } |
| } |
| @@ -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); |
| } |