initial audio HAL implementation for mako
alsa_sound is imported from codeaurora at:
c1217338f349fe746e0933fcf9b1b288b532808d
[remote "quic"]
url = git://git-android.quicinc.com/platform/hardware/alsa_sound.git
review = review-android.quicinc.com
projectname = platform/hardware/alsa_sound
fetch = +refs/heads/*:refs/remotes/quic/*
Change-Id: Ic985cc3a1088c3957b6e2ac5537e2c36caaf7212
Signed-off-by: Iliyan Malchev <malchev@google.com>
diff --git a/alsa_sound/AudioPolicyManagerALSA.h b/alsa_sound/AudioPolicyManagerALSA.h
new file mode 100644
index 0000000..75ed196
--- /dev/null
+++ b/alsa_sound/AudioPolicyManagerALSA.h
@@ -0,0 +1,108 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ * Copyright (C) 2011, Code Aurora Forum. All rights reserved.
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+
+#include <stdint.h>
+#include <sys/types.h>
+#include <utils/Timers.h>
+#include <utils/Errors.h>
+#include <utils/KeyedVector.h>
+#include <hardware_legacy/AudioPolicyManagerBase.h>
+
+
+namespace android_audio_legacy {
+
+// ----------------------------------------------------------------------------
+
+#define MAX_DEVICE_ADDRESS_LEN 20
+// Attenuation applied to STRATEGY_SONIFICATION streams when a headset is connected: 6dB
+#define SONIFICATION_HEADSET_VOLUME_FACTOR 0.5
+// Min volume for STRATEGY_SONIFICATION streams when limited by music volume: -36dB
+#define SONIFICATION_HEADSET_VOLUME_MIN 0.016
+// Time in seconds during which we consider that music is still active after a music
+// track was stopped - see computeVolume()
+#define SONIFICATION_HEADSET_MUSIC_DELAY 5
+class AudioPolicyManager: public AudioPolicyManagerBase
+{
+
+public:
+ AudioPolicyManager(AudioPolicyClientInterface *clientInterface)
+ : AudioPolicyManagerBase(clientInterface) {
+ mLPADecodeOutput = -1;
+ mLPAMuted = false;
+ mLPAStreamType = AudioSystem::DEFAULT;
+ }
+
+ virtual ~AudioPolicyManager() {}
+
+ // AudioPolicyInterface
+ virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
+ AudioSystem::device_connection_state state,
+ const char *device_address);
+ virtual void setPhoneState(int state);
+
+ // return appropriate device for streams handled by the specified strategy according to current
+ // phone state, connected devices...
+ // if fromCache is true, the device is returned from mDeviceForStrategy[], otherwise it is determined
+ // by current state (device connected, phone state, force use, a2dp output...)
+ // This allows to:
+ // 1 speed up process when the state is stable (when starting or stopping an output)
+ // 2 access to either current device selection (fromCache == true) or
+ // "future" device selection (fromCache == false) when called from a context
+ // where conditions are changing (setDeviceConnectionState(), setPhoneState()...) AND
+ // before updateDeviceForStrategy() is called.
+ virtual uint32_t getDeviceForStrategy(routing_strategy strategy, bool fromCache = true);
+
+#ifdef TUNNEL_LPA_ENABLED
+ virtual audio_io_handle_t getSession(AudioSystem::stream_type stream,
+ uint32_t format,
+ AudioSystem::output_flags flags,
+ int32_t sessionId,
+ uint32_t samplingRate, uint32_t channels);
+ virtual void pauseSession(audio_io_handle_t output, AudioSystem::stream_type stream);
+ virtual void resumeSession(audio_io_handle_t output, AudioSystem::stream_type stream);
+ virtual void releaseSession(audio_io_handle_t output);
+#endif
+
+ virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream, int session = 0);
+ virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream, int session = 0);
+ virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config);
+ status_t startInput(audio_io_handle_t input);
+
+protected:
+ // true is current platform implements a back microphone
+ virtual bool hasBackMicrophone() const { return false; }
+#ifdef WITH_A2DP
+ // true is current platform supports suplication of notifications and ringtones over A2DP output
+ virtual bool a2dpUsedForSonification() const { return true; }
+#endif
+ // change the route of the specified output
+ void setOutputDevice(audio_io_handle_t output, uint32_t device, bool force = false, int delayMs = 0);
+ // check that volume change is permitted, compute and send new volume to audio hardware
+ status_t checkAndSetVolume(int stream, int index, audio_io_handle_t output, uint32_t device, int delayMs = 0, bool force = false);
+ // select input device corresponding to requested audio source
+ virtual uint32_t getDeviceForInputSource(int inputSource);
+ // Mute or unmute the stream on the specified output
+ void setStreamMute(int stream, bool on, audio_io_handle_t output, int delayMs = 0);
+ audio_io_handle_t mLPADecodeOutput; // active output handler
+ audio_io_handle_t mLPAActiveOuput; // LPA Output Handler during inactive state
+
+ bool mLPAMuted;
+ AudioSystem::stream_type mLPAStreamType;
+ AudioSystem::stream_type mLPAActiveStreamType;
+};
+};