Index: content/renderer/media/media_stream_audio_processor.cc |
diff --git a/content/renderer/media/media_stream_audio_processor.cc b/content/renderer/media/media_stream_audio_processor.cc |
index c83ffff7eb408417dd9a641fc3c93f92a032119f..dcd9d40dbc9d52e22af5fdbc09ec14fa2b6ab74d 100644 |
--- a/content/renderer/media/media_stream_audio_processor.cc |
+++ b/content/renderer/media/media_stream_audio_processor.cc |
@@ -11,6 +11,7 @@ |
#include "content/public/common/content_switches.h" |
#include "content/renderer/media/media_stream_audio_processor_options.h" |
#include "content/renderer/media/rtc_media_constraints.h" |
+#include "content/renderer/media/webrtc_audio_device_impl.h" |
#include "media/audio/audio_parameters.h" |
#include "media/base/audio_converter.h" |
#include "media/base/audio_fifo.h" |
@@ -474,6 +475,7 @@ int MediaStreamAudioProcessor::ProcessData(webrtc::AudioFrame* audio_frame, |
audio_processing_->set_stream_delay_ms(total_delay_ms); |
+ DCHECK_LE(volume, WebRtcAudioDeviceImpl::kMaxVolumeLevel); |
webrtc::GainControl* agc = audio_processing_->gain_control(); |
int err = agc->set_stream_analog_level(volume); |
DCHECK_EQ(err, 0) << "set_stream_analog_level() error: " << err; |