| 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());
|
| }
|
| }
|
|
|
| @@ -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);
|
| }
|
|
|
|
|