Added a level indicator to new mixer.

webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc

Index: webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc
diff --git a/webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc b/webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc
index 48380fc550129a6ea62882fe96941a5148473701..b67dbda7091a1735eff4db4c81dfd82065b267d1 100644
--- a/webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc
+++ b/webrtc/modules/audio_mixer/new_audio_conference_mixer_impl.cc
@@ -73,7 +73,6 @@ void RemixFrame(AudioFrame* frame, size_t number_of_channels) {
 // |mixed_frame| always has at least as many channels as |frame|. Supports
 // stereo at most.
 //
-// TODO(andrew): consider not modifying |frame| here.

[minyue-webrtc, 2016/08/11 10:27:33]

you think there is no need to consider this anymore?

[aleloi, 2016/08/11 12:15:19]

I wrote that in a comment in one of the earlier patch sets. We will replace the
AudioFrame with something else and rewrite the mixer code that touches it.
Besides, I don't see a reason for why one would not modify the frame. Please
tell me if you can think of one :)

[minyue-webrtc, 2016/08/15 06:30:08]

I do not see any case, it is up to you to decide whether to keep the comment

 void MixFrames(AudioFrame* mixed_frame, AudioFrame* frame, bool use_limiter) {
   RTC_DCHECK_GE(mixed_frame->num_channels_, frame->num_channels_);
   if (use_limiter) {
@@ -237,8 +236,7 @@ void NewAudioConferenceMixerImpl::Mix(int sample_rate,
 
   time_stamp_ += static_cast<uint32_t>(sample_size_);
 
-  use_limiter_ = num_mixed_audio_sources_ > 1 &&
-                 output_frequency_ <= AudioProcessing::kMaxNativeSampleRateHz;
+  use_limiter_ = num_mixed_audio_sources_ > 1;
 
   // We only use the limiter if it supports the output sample rate and
   // we're actually mixing multiple streams.
@@ -257,6 +255,10 @@ void NewAudioConferenceMixerImpl::Mix(int sample_rate,
       LimitMixedAudio(audio_frame_for_mixing);
     }
   }
+
+  // Pass the final result to the level indicator.
+  audio_level_.ComputeLevel(*audio_frame_for_mixing);
+
   return;
 }
 
@@ -591,4 +593,20 @@ bool NewAudioConferenceMixerImpl::LimitMixedAudio(
   }
   return true;
 }
+
+uint32_t NewAudioConferenceMixerImpl::GetSpeechOutputLevel() {
+  int8_t currentLevel = audio_level_.Level();
+  uint32_t level = static_cast<uint32_t>(currentLevel);
+  WEBRTC_TRACE(kTraceStateInfo, kTraceAudioMixerServer, id_,
+               "GetSpeechOutputLevel() => level=%u", level);
+  return level;
+}
+
+uint32_t NewAudioConferenceMixerImpl::GetSpeechOutputLevelFullRange() {
+  int16_t currentLevel = audio_level_.LevelFullRange();
+  uint32_t level = static_cast<uint32_t>(currentLevel);
+  WEBRTC_TRACE(kTraceStateInfo, kTraceAudioMixerServer, id_,
+               "GetSpeechOutputLevelFullRange() => level=%u", level);
+  return level;
+}
 }  // namespace webrtc