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authorChih-hung Hsieh <chh@google.com>2015-12-01 17:07:48 +0000
committerandroid-build-merger <android-build-merger@google.com>2015-12-01 17:07:48 +0000
commita4acd9d6bc9b3b033d7d274316e75ee067df8d20 (patch)
tree672a185b294789cf991f385c3e395dd63bea9063 /webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
parent3681b90ba4fe7a27232dd3e27897d5d7ed9d651c (diff)
parentfe8b4a657979b49e1701bd92f6d5814a99e0b2be (diff)
downloadwebrtc-a4acd9d6bc9b3b033d7d274316e75ee067df8d20.tar.gz
Merge changes I7bbf776e,I1b827825
am: fe8b4a6579 * commit 'fe8b4a657979b49e1701bd92f6d5814a99e0b2be': (7237 commits) WIP: Changes after merge commit 'cb3f9bd' Make the nonlinear beamformer steerable Utilize bitrate above codec max to protect video. Enable VP9 internal resize by default. Filter overlapping RTP header extensions. Make VCMEncodedFrameCallback const. MediaCodecVideoEncoder: Add number of quality resolution downscales to Encoded callback. Remove redudant encoder rate calls. Create isolate files for nonparallel tests. Register header extensions in RtpRtcpObserver to avoid log spam. Make an enum class out of NetEqDecoder, and hide the neteq_decoders_ table ACM: Move NACK functionality inside NetEq Fix chromium-style warnings in webrtc/sound/. Create a 'webrtc_nonparallel_tests' target. Update scalability structure data according to updates in the RTP payload profile. audio_coding: rename interface -> include Rewrote perform_action_on_all_files to be parallell. Update reference indices according to updates in the RTP payload profile. Disable P2PTransport...TestFailoverControlledSide on Memcheck pass clangcl compile options to ignore warnings in gflags.cc ...
Diffstat (limited to 'webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h')
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diff --git a/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h b/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
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+++ b/webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h
@@ -0,0 +1,80 @@
+/*
+ * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_MAC_H
+#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_MAC_H
+
+#include "webrtc/modules/audio_device/include/audio_device.h"
+#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
+#include "webrtc/typedefs.h"
+
+#include <CoreAudio/CoreAudio.h>
+
+namespace webrtc {
+
+class AudioMixerManagerMac
+{
+public:
+ int32_t OpenSpeaker(AudioDeviceID deviceID);
+ int32_t OpenMicrophone(AudioDeviceID deviceID);
+ int32_t SetSpeakerVolume(uint32_t volume);
+ int32_t SpeakerVolume(uint32_t& volume) const;
+ int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
+ int32_t MinSpeakerVolume(uint32_t& minVolume) const;
+ int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
+ int32_t SpeakerVolumeIsAvailable(bool& available);
+ int32_t SpeakerMuteIsAvailable(bool& available);
+ int32_t SetSpeakerMute(bool enable);
+ int32_t SpeakerMute(bool& enabled) const;
+ int32_t StereoPlayoutIsAvailable(bool& available);
+ int32_t StereoRecordingIsAvailable(bool& available);
+ int32_t MicrophoneMuteIsAvailable(bool& available);
+ int32_t SetMicrophoneMute(bool enable);
+ int32_t MicrophoneMute(bool& enabled) const;
+ int32_t MicrophoneBoostIsAvailable(bool& available);
+ int32_t SetMicrophoneBoost(bool enable);
+ int32_t MicrophoneBoost(bool& enabled) const;
+ int32_t MicrophoneVolumeIsAvailable(bool& available);
+ int32_t SetMicrophoneVolume(uint32_t volume);
+ int32_t MicrophoneVolume(uint32_t& volume) const;
+ int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
+ int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
+ int32_t MicrophoneVolumeStepSize(uint16_t& stepSize) const;
+ int32_t Close();
+ int32_t CloseSpeaker();
+ int32_t CloseMicrophone();
+ bool SpeakerIsInitialized() const;
+ bool MicrophoneIsInitialized() const;
+
+public:
+ AudioMixerManagerMac(const int32_t id);
+ ~AudioMixerManagerMac();
+
+private:
+ static void logCAMsg(const TraceLevel level,
+ const TraceModule module,
+ const int32_t id, const char *msg,
+ const char *err);
+
+private:
+ CriticalSectionWrapper& _critSect;
+ int32_t _id;
+
+ AudioDeviceID _inputDeviceID;
+ AudioDeviceID _outputDeviceID;
+
+ uint16_t _noInputChannels;
+ uint16_t _noOutputChannels;
+
+};
+
+} // namespace webrtc
+
+#endif // AUDIO_MIXER_MAC_H