Fix issue 1795088 Improve audio routing code

Initial commit for review.

3 files changed, 244 insertions, 59 deletions

diff --git a/include/hardware_legacy/AudioHardwareBase.h b/include/hardware_legacy/AudioHardwareBase.h
index 1065c39..ae2e3ef 100644
--- a/include/hardware_legacy/AudioHardwareBase.h
+++ b/include/hardware_legacy/AudioHardwareBase.h
@@ -33,19 +33,6 @@ class AudioHardwareBase : public AudioHardwareInterface
 public:
                         AudioHardwareBase();
     virtual             ~AudioHardwareBase() { }
-
-    /**  
-     * Audio routing methods. Routes defined in include/hardware_legacy/AudioSystem.h. 
-     * Audio routes can be (ROUTE_EARPIECE | ROUTE_SPEAKER | ROUTE_BLUETOOTH 
-     *                    | ROUTE_HEADSET)
-     * 
-     * setRouting sets the routes for a mode. This is called at startup. It is
-     * also called when a new device is connected, such as a wired headset is 
-     * plugged in or a Bluetooth headset is paired.
-     */
-    virtual status_t    setRouting(int mode, uint32_t routes);
-    
-    virtual status_t    getRouting(int mode, uint32_t* routes);
     
     /**
      * setMode is called when the audio mode changes. NORMAL mode is for
@@ -53,11 +40,9 @@ public:
      * when a call is in progress.
      */
     virtual status_t    setMode(int mode);
-    virtual status_t    getMode(int* mode);
 
-    // Temporary interface, do not use
-    // TODO: Replace with a more generic key:value get/set mechanism
-    virtual status_t    setParameter(const char* key, const char* value);
+    virtual status_t    setParameters(const String8& keyValuePairs);
+    virtual String8     getParameters(const String8& keys);
     
     virtual  size_t     getInputBufferSize(uint32_t sampleRate, int format, int channelCount);
     
@@ -66,7 +51,6 @@ public:
 
 protected:
     int             mMode;
-    uint32_t        mRoutes[AudioSystem::NUM_MODES];
 };
 
 }; // namespace android
diff --git a/include/hardware_legacy/AudioHardwareInterface.h b/include/hardware_legacy/AudioHardwareInterface.h
index 8d37e4f..46cea77 100644
--- a/include/hardware_legacy/AudioHardwareInterface.h
+++ b/include/hardware_legacy/AudioHardwareInterface.h
@@ -23,6 +23,7 @@
 #include <utils/Errors.h>
 #include <utils/Vector.h>
 #include <utils/String16.h>
+#include <utils/String8.h>
 
 #include <media/IAudioFlinger.h>
 #include "media/AudioSystem.h"
@@ -48,10 +49,9 @@ public:
     virtual size_t      bufferSize() const = 0;
 
     /**
-     * return number of output audio channels.
-     * Acceptable values are 1 (mono) or 2 (stereo)
+     * returns the output channel nask
      */
-    virtual int         channelCount() const = 0;
+    virtual uint32_t    channels() const = 0;
 
     /**
      * return audio format in 8bit or 16bit PCM format -
@@ -62,7 +62,7 @@ public:
     /**
      * return the frame size (number of bytes per sample).
      */
-    uint32_t    frameSize() const { return channelCount()*((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
+    uint32_t    frameSize() const { return AudioSystem::popCount(channels())*((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
 
     /**
      * return the audio hardware driver latency in milli seconds.
@@ -76,7 +76,7 @@ public:
      * This method might produce multiple PCM outputs or hardware accelerated
      * codecs, such as MP3 or AAC.
      */
-    virtual status_t    setVolume(float volume) = 0;
+    virtual status_t    setVolume(float left, float right) = 0;
 
     /** write audio buffer to driver. Returns number of bytes written */
     virtual ssize_t     write(const void* buffer, size_t bytes) = 0;
@@ -89,6 +89,15 @@ public:
 
     /** dump the state of the audio output device */
     virtual status_t dump(int fd, const Vector<String16>& args) = 0;
+
+    // set/get audio output parameters. The function accepts a list of parameters
+    // key value pairs in the form: key1=value1;key2=value2;...
+    // Some keys are reserved for standard parameters (See AudioParameter class).
+    // If the implementation does not accept a parameter change while the output is
+    // active but the parameter is acceptable otherwise, it must return INVALID_OPERATION.
+    // The audio flinger will put the output in standby and then change the parameter value.
+    virtual status_t    setParameters(const String8& keyValuePairs) = 0;
+    virtual String8     getParameters(const String8& keys) = 0;
 };
 
 /**
@@ -100,11 +109,14 @@ class AudioStreamIn {
 public:
     virtual             ~AudioStreamIn() = 0;
 
+    /** return audio sampling rate in hz - eg. 44100 */
+    virtual uint32_t    sampleRate() const = 0;
+
     /** return the input buffer size allowed by audio driver */
     virtual size_t      bufferSize() const = 0;
 
-    /** return the number of audio input channels */
-    virtual int         channelCount() const = 0;
+    /** return input channel mask */
+    virtual uint32_t    channels() const = 0;
 
     /**
      * return audio format in 8bit or 16bit PCM format -
@@ -115,7 +127,7 @@ public:
     /**
      * return the frame size (number of bytes per sample).
      */
-    uint32_t    frameSize() const { return channelCount()*((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
+    uint32_t    frameSize() const { return AudioSystem::popCount(channels())*((format()==AudioSystem::PCM_16_BIT)?sizeof(int16_t):sizeof(int8_t)); }
 
     /** set the input gain for the audio driver. This method is for
      *  for future use */
@@ -133,6 +145,14 @@ public:
      */
     virtual status_t    standby() = 0;
 
+    // set/get audio input parameters. The function accepts a list of parameters
+    // key value pairs in the form: key1=value1;key2=value2;...
+    // Some keys are reserved for standard parameters (See AudioParameter class).
+    // If the implementation does not accept a parameter change while the output is
+    // active but the parameter is acceptable otherwise, it must return INVALID_OPERATION.
+    // The audio flinger will put the input in standby and then change the parameter value.
+    virtual status_t    setParameters(const String8& keyValuePairs) = 0;
+    virtual String8     getParameters(const String8& keys) = 0;
 };
 
 /**
@@ -169,53 +189,41 @@ public:
     virtual status_t    setMasterVolume(float volume) = 0;
 
     /**
-     * Audio routing methods. Routes defined in include/hardware_legacy/AudioSystem.h.
-     * Audio routes can be (ROUTE_EARPIECE | ROUTE_SPEAKER | ROUTE_BLUETOOTH
-     *                    | ROUTE_HEADSET)
-     *
-     * setRouting sets the routes for a mode. This is called at startup. It is
-     * also called when a new device is connected, such as a wired headset is
-     * plugged in or a Bluetooth headset is paired.
-     */
-    virtual status_t    setRouting(int mode, uint32_t routes) = 0;
-
-    virtual status_t    getRouting(int mode, uint32_t* routes) = 0;
-
-    /**
      * setMode is called when the audio mode changes. NORMAL mode is for
      * standard audio playback, RINGTONE when a ringtone is playing, and IN_CALL
      * when a call is in progress.
      */
     virtual status_t    setMode(int mode) = 0;
-    virtual status_t    getMode(int* mode) = 0;
 
     // mic mute
     virtual status_t    setMicMute(bool state) = 0;
     virtual status_t    getMicMute(bool* state) = 0;
 
-    // Temporary interface, do not use
-    // TODO: Replace with a more generic key:value get/set mechanism
-    virtual status_t    setParameter(const char* key, const char* value) = 0;
+    // set/get global audio parameters
+    virtual status_t    setParameters(const String8& keyValuePairs) = 0;
+    virtual String8     getParameters(const String8& keys) = 0;
 
     // Returns audio input buffer size according to parameters passed or 0 if one of the
     // parameters is not supported
     virtual size_t    getInputBufferSize(uint32_t sampleRate, int format, int channelCount) = 0;
-    
+
     /** This method creates and opens the audio hardware output stream */
     virtual AudioStreamOut* openOutputStream(
-                                int format=0,
-                                int channelCount=0,
-                                uint32_t sampleRate=0,
+                                uint32_t devices,
+                                int *format=0,
+                                uint32_t *channels=0,
+                                uint32_t *sampleRate=0,
                                 status_t *status=0) = 0;
-
+    virtual    void        closeOutputStream(AudioStreamOut* out) = 0;
     /** This method creates and opens the audio hardware input stream */
     virtual AudioStreamIn* openInputStream(
-                                int inputSource,
-                                int format,
-                                int channelCount,
-                                uint32_t sampleRate,
+                                uint32_t devices,
+                                int *format,
+                                uint32_t *channels,
+                                uint32_t *sampleRate,
                                 status_t *status,
                                 AudioSystem::audio_in_acoustics acoustics) = 0;
+    virtual    void        closeInputStream(AudioStreamIn* in) = 0;
 
     /**This method dumps the state of the audio hardware */
     virtual status_t dumpState(int fd, const Vector<String16>& args) = 0;
@@ -223,13 +231,6 @@ public:
     static AudioHardwareInterface* create();
 
 protected:
-    /**
-     * doRouting actually initiates the routing. A call to setRouting
-     * or setMode may result in a routing change. The generic logic calls
-     * doRouting when required. If the device has any special requirements these
-     * methods can be overriden.
-     */
-    virtual status_t    doRouting() = 0;
 
     virtual status_t dump(int fd, const Vector<String16>& args) = 0;
 };
diff --git a/include/hardware_legacy/AudioPolicyInterface.h b/include/hardware_legacy/AudioPolicyInterface.h
new file mode 100644
index 0000000..57a60ae
--- /dev/null
+++ b/include/hardware_legacy/AudioPolicyInterface.h
@@ -0,0 +1,200 @@
+/*
+ * Copyright (C) 2009 The Android Open Source Project
+ *
+ * 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.
+ */
+
+#ifndef ANDROID_AUDIOPOLICYINTERFACE_H
+#define ANDROID_AUDIOPOLICYINTERFACE_H
+
+#include <media/AudioSystem.h>
+#include <media/ToneGenerator.h>
+#include <utils/String8.h>
+
+namespace android {
+
+
+// ----------------------------------------------------------------------------
+
+// The AudioPolicyInterface and AudioPolicyClientInterface classes define the communication interfaces
+// between the platform specific audio policy manager and Android generic audio policy manager.
+// The platform specific audio policy manager must implement methods of the AudioPolicyInterface class.
+// This implementation makes use of the AudioPolicyClientInterface to control the activity and
+// configuration of audio input and output streams.
+//
+// The platform specific audio policy manager is in charge of the audio routing and volume control
+// policies for a given platform.
+// The main roles of this module are:
+//   - keep track of current system state (removable device connections, phone state, user requests...).
+//   System state changes and user actions are notified to audio policy manager with methods of the AudioPolicyInterface.
+//   - process getOutput() queries received when AudioTrack objects are created: Those queries
+//   return a handler on an output that has been selected, configured and opened by the audio policy manager and that
+//   must be used by the AudioTrack when registering to the AudioFlinger with the createTrack() method.
+//   When the AudioTrack object is released, a putOutput() query is received and the audio policy manager can decide
+//   to close or reconfigure the output depending on other streams using this output and current system state.
+//   - similarly process getInput() and putInput() queries received from AudioRecord objects and configure audio inputs.
+//   - process volume control requests: the stream volume is converted from an index value (received from UI) to a float value
+//   applicable to each output as a function of platform specific settings and current output route (destination device). It
+//   also make sure that streams are not muted if not allowed (e.g. camera shutter sound in some countries).
+//
+// The platform specific audio policy manager is provided as a shared library by platform vendors (as for libaudio.so)
+// and is linked with libaudioflinger.so
+
+
+//    Audio Policy Manager Interface
+class AudioPolicyInterface
+{
+
+public:
+    virtual ~AudioPolicyInterface() {}
+    //
+    // configuration functions
+    //
+
+    // indicate a change in device connection status
+    virtual status_t setDeviceConnectionState(AudioSystem::audio_devices device,
+                                          AudioSystem::device_connection_state state,
+                                          const char *device_address) = 0;
+    // retreive a device connection status
+    virtual AudioSystem::device_connection_state getDeviceConnectionState(AudioSystem::audio_devices device,
+                                                                          const char *device_address) = 0;
+    // indicate a change in phone state. Valid phones states are defined by AudioSystem::audio_mode
+    virtual void setPhoneState(int state) = 0;
+    // indicate a change in ringer mode
+    virtual void setRingerMode(uint32_t mode, uint32_t mask) = 0;
+    // force using a specific device category for the specified usage
+    virtual void setForceUse(AudioSystem::force_use usage, AudioSystem::forced_config config) = 0;
+    // retreive current device category forced for a given usage
+    virtual AudioSystem::forced_config getForceUse(AudioSystem::force_use usage) = 0;
+    // set a system property (e.g. camera sound always audible)
+    virtual void setSystemProperty(const char* property, const char* value) = 0;
+
+
+    //
+    // Audio routing query functions
+    //
+
+    // request an output appriate for playback of the supplied stream type and parameters
+    virtual audio_io_handle_t getOutput(AudioSystem::stream_type stream,
+                                        uint32_t samplingRate = 0,
+                                        uint32_t format = AudioSystem::FORMAT_DEFAULT,
+                                        uint32_t channels = 0,
+                                        AudioSystem::output_flags flags = AudioSystem::OUTPUT_FLAG_INDIRECT) = 0;
+    // indicates to the audio policy manager that the output starts being used by corresponding stream.
+    virtual status_t startOutput(audio_io_handle_t output, AudioSystem::stream_type stream) = 0;
+    // indicates to the audio policy manager that the output stops being used by corresponding stream.
+    virtual status_t stopOutput(audio_io_handle_t output, AudioSystem::stream_type stream) = 0;
+    // releases the output.
+    virtual void releaseOutput(audio_io_handle_t output) = 0;
+
+    // request an input appriate for record from the supplied device with supplied parameters.
+    virtual audio_io_handle_t getInput(int inputSource,
+                                    uint32_t samplingRate = 0,
+                                    uint32_t Format = AudioSystem::FORMAT_DEFAULT,
+                                    uint32_t channels = 0,
+                                    AudioSystem::audio_in_acoustics acoustics = (AudioSystem::audio_in_acoustics)0) = 0;
+    // indicates to the audio policy manager that the input starts being used.
+    virtual status_t startInput(audio_io_handle_t input) = 0;
+    // indicates to the audio policy manager that the input stops being used.
+    virtual status_t stopInput(audio_io_handle_t input) = 0;
+    // releases the input.
+    virtual void releaseInput(audio_io_handle_t input) = 0;
+
+    //
+    // volume control functions
+    //
+
+    // initialises stream volume conversion parameters by specifying volume index range.
+    virtual void initStreamVolume(AudioSystem::stream_type stream,
+                                      int indexMin,
+                                      int indexMax) = 0;
+
+    // sets the new stream volume at a level corresponding to the supplied index
+    virtual status_t setStreamVolumeIndex(AudioSystem::stream_type stream, int index) = 0;
+    // retreive current volume index for the specified stream
+    virtual status_t getStreamVolumeIndex(AudioSystem::stream_type stream, int *index) = 0;
+};
+
+
+// Audio Policy client Interface
+class AudioPolicyClientInterface
+{
+public:
+    virtual ~AudioPolicyClientInterface() {}
+
+    //
+    // Audio output Control functions
+    //
+
+    // opens an audio output with the requested parameters. The parameter values can indicate to use the default values
+    // in case the audio policy manager has no specific requirements for the output being opened.
+    // When the function returns, the parameter values reflect the actual values used by the audio hardware output stream.
+    // The audio policy manager can check if the proposed parameters are suitable or not and act accordingly.
+    virtual audio_io_handle_t openOutput(uint32_t *pDevices,
+                                    uint32_t *pSamplingRate,
+                                    uint32_t *pFormat,
+                                    uint32_t *pChannels,
+                                    uint32_t *pLatencyMs,
+                                    AudioSystem::output_flags flags) = 0;
+    // creates a special output that is duplicated to the two outputs passed as arguments. The duplication is performed by
+    // a special mixer thread in the AudioFlinger.
+    virtual audio_io_handle_t openDuplicateOutput(audio_io_handle_t output1, audio_io_handle_t output2) = 0;
+    // closes the output stream
+    virtual status_t closeOutput(audio_io_handle_t output) = 0;
+    // suspends the output. When an output is suspended, the corresponding audio hardware output stream is placed in
+    // standby and the AudioTracks attached to the mixer thread are still processed but the output mix is discarded.
+    virtual status_t suspendOutput(audio_io_handle_t output) = 0;
+    // restores a suspended output.
+    virtual status_t restoreOutput(audio_io_handle_t output) = 0;
+
+    //
+    // Audio input Control functions
+    //
+
+    // opens an audio input
+    virtual audio_io_handle_t openInput(uint32_t *pDevices,
+                                    uint32_t *pSamplingRate,
+                                    uint32_t *pFormat,
+                                    uint32_t *pChannels,
+                                    uint32_t acoustics) = 0;
+    // closes an audio input
+    virtual status_t closeInput(audio_io_handle_t input) = 0;
+    //
+    // misc control functions
+    //
+
+    // set a stream volume for a particular output. For the same user setting, a given stream type can have different volumes
+    // for each output (destination device) it is attached to.
+    virtual status_t setStreamVolume(AudioSystem::stream_type stream, float volume, audio_io_handle_t output) = 0;
+
+    // reroute a given stream type to the specified output
+    virtual status_t setStreamOutput(AudioSystem::stream_type stream, audio_io_handle_t output) = 0;
+
+    // function enabling to send proprietary informations directly from audio policy manager to audio hardware interface.
+    virtual void setParameters(audio_io_handle_t ioHandle, const String8& keyValuePairs) = 0;
+    // function enabling to receive proprietary informations directly from audio hardware interface to audio policy manager.
+    virtual String8 getParameters(audio_io_handle_t ioHandle, const String8& keys) = 0;
+
+    // request the playback of a tone on the specified stream: used for instance to replace notification sounds when playing
+    // over a telephony device during a phone call.
+    virtual status_t startTone(ToneGenerator::tone_type tone, AudioSystem::stream_type stream) = 0;
+    virtual status_t stopTone() = 0;
+};
+
+extern "C" AudioPolicyInterface* createAudioPolicyManager(AudioPolicyClientInterface *clientInterface);
+extern "C" void destroyAudioPolicyManager(AudioPolicyInterface *interface);
+
+
+}; // namespace android
+
+#endif // ANDROID_AUDIOPOLICYINTERFACE_H