commit 5009286ebb46bdff5f21b9ea6de112410326b555
author bjornv@webrtc.org <bjornv@webrtc.org> Fri Feb 06 19:44:21 2015
committer bjornv@webrtc.org <bjornv@webrtc.org> Fri Feb 06 19:44:21 2015
tree 95961ba26a58c9aa5832eb3206a0a4b342a75e64
parent afcb4342721f1c35775a30ba5b55a2560d63272f

audio_processing: Now records mic volume level also when using new AGC

Previously only mic level calculated by the legacy agc was logged in aecdebug dumps.
Now we log it for any agc.
In addition, it is now possible to turn on and off debug recording in the test tool voe_cmd_test.

BUG=4274
TESTED=verified using voe_cmd_test
R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39839004

git-svn-id: http://webrtc.googlecode.com/svn/trunk/webrtc@8274 4adac7df-926f-26a2-2b94-8c16560cd09d

modules/audio_processing/audio_processing_impl.cc

diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 7eeda11..1319e16 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -591,7 +591,7 @@
     audioproc::Stream* msg = event_msg_->mutable_stream();
     msg->set_delay(stream_delay_ms_);
     msg->set_drift(echo_cancellation_->stream_drift_samples());
-    msg->set_level(gain_control_->stream_analog_level());
+    msg->set_level(gain_control()->stream_analog_level());
     msg->set_keypress(key_pressed_);
   }
 #endif

voice_engine/test/cmd_test/voe_cmd_test.cc

diff --git a/voice_engine/test/cmd_test/voe_cmd_test.cc b/voice_engine/test/cmd_test/voe_cmd_test.cc
index 1155234..454baff 100644
--- a/voice_engine/test/cmd_test/voe_cmd_test.cc
+++ b/voice_engine/test/cmd_test/voe_cmd_test.cc
@@ -233,6 +233,7 @@
   bool muted = false;
   bool opus_stereo = false;
   bool experimental_ns_enabled = false;
+  bool debug_recording_started = false;
 
 #if defined(WEBRTC_ANDROID)
   std::string resource_path = "/sdcard/";
@@ -448,6 +449,7 @@
       printf("%i. Set Opus maximum playback rate \n", option_index++);
       printf("%i. Set bit rate (only take effect on codecs that allow the "
              "change) \n", option_index++);
+      printf("%i. Toggle debug recording \n", option_index++);
 
       printf("Select action or %i to stop the call: ", option_index);
       int option_selection;
@@ -778,11 +780,25 @@
         ASSERT_EQ(1, scanf("%i", &cinst.rate));
         res = codec->SetSendCodec(chan, cinst);
         VALIDATE;
+      } else if (option_selection == option_index++) {
+        const char* kDebugFileName = "audio.aecdump";
+        if (debug_recording_started) {
+          apm->StopDebugRecording();
+          printf("Debug recording named %s stopped\n", kDebugFileName);
+        } else {
+          apm->StartDebugRecording(kDebugFileName);
+          printf("Debug recording named %s started\n", kDebugFileName);
+        }
+        debug_recording_started = !debug_recording_started;
       } else {
         break;
       }
     }
 
+    if (debug_recording_started) {
+      apm->StopDebugRecording();
+    }
+
     if (send) {
       printf("Stop Send \n");
       res = base1->StopSend(chan);