Index: webrtc/audio/audio_send_stream.cc |
diff --git a/webrtc/audio/audio_send_stream.cc b/webrtc/audio/audio_send_stream.cc |
index ec18125427282d4624543bc45b274d2b1a42692e..2d0c99b33a6bb23358b72ca85176ad0679630d6e 100644 |
--- a/webrtc/audio/audio_send_stream.cc |
+++ b/webrtc/audio/audio_send_stream.cc |
@@ -25,7 +25,7 @@ |
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
#include "webrtc/voice_engine/channel_proxy.h" |
#include "webrtc/voice_engine/include/voe_base.h" |
-#include "webrtc/voice_engine/include/voe_volume_control.h" |
+#include "webrtc/voice_engine/transmit_mixer.h" |
#include "webrtc/voice_engine/voice_engine_impl.h" |
namespace webrtc { |
@@ -193,16 +193,11 @@ webrtc::AudioSendStream::Stats AudioSendStream::GetStats() const { |
} |
} |
- // Local speech level. |
- { |
- ScopedVoEInterface<VoEVolumeControl> volume(voice_engine()); |
- unsigned int level = 0; |
- int error = volume->GetSpeechInputLevelFullRange(level); |
- RTC_DCHECK_EQ(0, error); |
- stats.audio_level = static_cast<int32_t>(level); |
- } |
- |
ScopedVoEInterface<VoEBase> base(voice_engine()); |
+ RTC_DCHECK(base->transmit_mixer()); |
+ stats.audio_level = base->transmit_mixer()->AudioLevelFullRange(); |
+ RTC_DCHECK_LE(0, stats.audio_level); |
+ |
RTC_DCHECK(base->audio_processing()); |
auto audio_processing_stats = base->audio_processing()->GetStatistics(); |
stats.echo_delay_median_ms = audio_processing_stats.delay_median; |