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..4c6375fafb59cd7f666fa94e3bd7c785c1602363 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 +101,30 @@ void AecDumpBasedSimulator::PrepareProcessStreamCall(
     }
   }
 
+  if (artificial_nearend_buffer_reader_) {
+    if (artificial_nearend_buffer_reader_->Read(

[hlundin-webrtc, 2016/12/09 07:48:26]

If the file ends before the rest of the simulation, we'll just silently not add
any more artificial nerend. Is this your intention? I think it is more honest to
the one running the test to kick and scream about it (essentially RTC_CHECK the
call to Read).

[peah-webrtc, 2016/12/09 08:07:51]

The intention was actually to do this silently but it can definitely be
discussed whether that is the best way. My view is that this is a
simulation/development tool and I don't think that it should limit the user in
any unnecessary way. If we CHECK on this, that will force the user to adjust the
length of the wav file just to comply with this restriction.

I'll add a printout to warn about this, because I agree that it could be that it
is not intentional for the .wav file to be that short.

WDYT?

[hlundin-webrtc, 2016/12/09 09:24:26]

Good.

+            artificial_nearend_buf_.get())) {
+      if (msg.has_input_data()) {
+        for (size_t k = 0; k < in_buf_->num_frames(); ++k) {
+          int32_t tmp = fwd_frame_.data_[k] +
+                        static_cast<int16_t>(
+                            32767 * artificial_nearend_buf_->channels()[0][k]);
+          fwd_frame_.data_[k] =
+              static_cast<int16_t>(std::min(32767, std::max(-32768, tmp)));

[hlundin-webrtc, 2016/12/09 07:48:26]

rtc::saturated_cast<int16_t>(tmp) will do the same thing.

[peah-webrtc, 2016/12/09 08:07:51]

Great! That I did not know. Changed the code to use that.

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];
+            in_buf_->channels()[i][k] = std::min(
+                32767.f, std::max(-32768.f, in_buf_->channels()[i][k]));
+          }
+        }
+      }
+    }
+  }
+
   if (!settings_.stream_delay) {
     if (msg.has_delay()) {
       RTC_CHECK_EQ(AudioProcessing::kNoError,
@@ -189,6 +218,24 @@ 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) {

[hlundin-webrtc, 2016/12/09 07:48:26]

Any point continuing here? Or simply
RTC_DCHECK_EQ(artificial_nearend_file->num_channels(), 1) << "Oh no...";

[peah-webrtc, 2016/12/09 08:07:51]

Good point! I added the DCHECK.

Done.

+      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)));
+
+      artificial_nearend_buf_.reset(new ChannelBuffer<float>(
+          rtc::CheckedDivExact(artificial_nearend_file->sample_rate(),
+                               kChunksPerSecond),
+          1));
+    }
+  }
+
   webrtc::audioproc::Event event_msg;
   int num_forward_chunks_processed = 0;
   const float kOneBykChunksPerSecond =