Add an optional artificial nearend signal for aecdump call recreation

webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc

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 694f2cd90c8f5cc702573a7ad1046b2e9983c151..f110af7de82d7b497671d491ca165d2dcb44c6d1 100644
--- a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
+++ b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.cc
@@ -62,6 +62,11 @@ bool VerifyFloatBitExactness(const webrtc::audioproc::Stream& msg,
 
 }  // namespace
 
+AecDumpBasedSimulator::AecDumpBasedSimulator(const SimulationSettings& settings)
+    : AudioProcessingSimulator(settings) {}
+
+AecDumpBasedSimulator::~AecDumpBasedSimulator() = default;
+
 void AecDumpBasedSimulator::PrepareProcessStreamCall(
     const webrtc::audioproc::Stream& msg,
     bool* set_stream_analog_level_called) {
@@ -96,6 +101,26 @@ void AecDumpBasedSimulator::PrepareProcessStreamCall(
     }
   }
 
+  if (artificial_nearend_buffer_reader_) {
+    bool samples_left_to_process =
+        artificial_nearend_buffer_reader_->Read(artificial_nearend_buf_.get());
+    if (samples_left_to_process) {
+      if (msg.has_input_data()) {
+        for (size_t k = 0; k < in_buf_->num_frames(); ++k) {
+          fwd_frame_.data_[0] += static_cast<int16_t>(
+              32767 * artificial_nearend_buf_->channels()[0][k]);

[ivoc, 2016/12/08 10:55:19]

Shouldn't this be fwd_frame_.data[k]? Also, why is it adding instead of
assigning?

[peah-webrtc, 2016/12/08 12:02:26]

Great find (I probably have not used this yet with the int16 interface.)

Regarding the adding, the idea is to add a nearend signal to the existing
microphone signal (which includes the echo which we want to maintain). Assigning
would replace the microphone signal.

Done.

[ivoc, 2016/12/08 12:13:24]

I see, that makes sense. I think this code can cause over/underflows as well,
should that be addressed, or is it not a problem in practice?

[peah-webrtc, 2016/12/09 07:19:25]

You are fully correct! I added explicit controlled saturation.

Done.

+        }
+      } else {
+        for (int i = 0; i < msg.input_channel_size(); ++i) {
+          for (size_t k = 0; k < in_buf_->num_frames(); ++k) {
+            in_buf_->channels()[i][k] +=
+                artificial_nearend_buf_->channels()[0][k];
+          }
+        }
+      }
+    }
+  }
+
   if (!settings_.stream_delay) {
     if (msg.has_delay()) {
       RTC_CHECK_EQ(AudioProcessing::kNoError,
@@ -189,6 +214,19 @@ void AecDumpBasedSimulator::Process() {
   CreateAudioProcessor();
   dump_input_file_ = OpenFile(settings_.aec_dump_input_filename->c_str(), "rb");
 
+  if (settings_.artificial_nearend_filename) {
+    std::unique_ptr<WavReader> artificial_nearend_file(
+        new WavReader(settings_.artificial_nearend_filename->c_str()));
+
+    if (artificial_nearend_file->num_channels() != 1) {
+      std::cout << "Only mono files for the artificial nearend are supported, "
+                   "reverted to not using the artificial nearend file";
+    } else {
+      artificial_nearend_buffer_reader_.reset(
+          new ChannelBufferWavReader(std::move(artificial_nearend_file)));
+    }
+  }
+
   webrtc::audioproc::Event event_msg;
   int num_forward_chunks_processed = 0;
   const float kOneBykChunksPerSecond =