Index: content/renderer/media/webrtc_audio_device_impl.cc |
diff --git a/content/renderer/media/webrtc_audio_device_impl.cc b/content/renderer/media/webrtc_audio_device_impl.cc |
index 240ea8974dc811378b494939b588ac78fb4823b8..3b589a8a023228a3cab882f6fd7c9e9a3c61d3db 100644 |
--- a/content/renderer/media/webrtc_audio_device_impl.cc |
+++ b/content/renderer/media/webrtc_audio_device_impl.cc |
@@ -6,6 +6,7 @@ |
#include "base/bind.h" |
#include "base/string_util.h" |
+#include "content/browser/renderer_host/media/audio_input_device_manager.h" |
#include "content/common/view_messages.h" |
#include "content/renderer/render_thread_impl.h" |
#include "media/audio/audio_util.h" |
@@ -173,11 +174,11 @@ void WebRtcAudioDeviceImpl::Capture( |
} |
} |
-void WebRtcAudioDeviceImpl::OnDeviceStarted(int device_index) { |
- VLOG(1) << "OnDeviceStarted (device_index=" << device_index << ")"; |
+void WebRtcAudioDeviceImpl::OnDeviceStarted(const std::string& device_uid) { |
+ VLOG(1) << "OnDeviceStarted (device_uid=" << device_uid << ")"; |
// -1 is an invalid device index. Do nothing if a valid device has |
// been started. Otherwise update the |recording_| state to false. |
- if (device_index != -1) |
+ if (device_uid != media_stream::AudioInputDeviceManager::kInvalidDeviceUId) |
return; |
base::AutoLock auto_lock(lock_); |
@@ -371,7 +372,7 @@ int32_t WebRtcAudioDeviceImpl::Init() { |
DLOG(ERROR) << "Only 48kHz sample rate is supported on Linux."; |
return -1; |
} |
- input_channels = 1; |
+ input_channels = 2; |
output_channels = 1; |
// Based on tests using the current ALSA implementation in Chrome, we have |