Create a snapshot of the hardware_legacy
Create a snapshot of hardware_legacy for legacy HIDL, to avoid any build breakage due to new hardware_legacy for the new AIDL. Should be removed with the HIDL when we finish the switch
Bug: 205044134
Test: build and CtsWifiTest
Change-Id: Ib1068112f6c90f2a41b68e20027d959c95798120
diff --git a/wifi/1.6/default/hal_legacy/AudioHardwareInterface.h b/wifi/1.6/default/hal_legacy/AudioHardwareInterface.h
new file mode 100644
index 0000000..7befb79
--- /dev/null
+++ b/wifi/1.6/default/hal_legacy/AudioHardwareInterface.h
@@ -0,0 +1,296 @@
+/*
+ * Copyright (C) 2007 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_AUDIO_HARDWARE_INTERFACE_H
+#define ANDROID_AUDIO_HARDWARE_INTERFACE_H
+
+#include <stdint.h>
+#include <sys/types.h>
+
+#include <utils/Errors.h>
+#include <utils/String16.h>
+#include <utils/String8.h>
+#include <utils/Vector.h>
+
+#include <hardware_legacy/AudioSystemLegacy.h>
+
+#include <hardware/audio.h>
+#include <system/audio.h>
+
+#include <cutils/bitops.h>
+
+namespace android_audio_legacy {
+using android::String16;
+using android::String8;
+using android::Vector;
+
+// ----------------------------------------------------------------------------
+
+/**
+ * AudioStreamOut is the abstraction interface for the audio output hardware.
+ *
+ * It provides information about various properties of the audio output hardware driver.
+ */
+class AudioStreamOut {
+ public:
+ virtual ~AudioStreamOut() = 0;
+
+ /** return audio sampling rate in hz - eg. 44100 */
+ virtual uint32_t sampleRate() const = 0;
+
+ /** returns size of output buffer - eg. 4800 */
+ virtual size_t bufferSize() const = 0;
+
+ /**
+ * returns the output channel mask
+ */
+ virtual uint32_t channels() const = 0;
+
+ /**
+ * return audio format in 8bit or 16bit PCM format -
+ * eg. AudioSystem:PCM_16_BIT
+ */
+ virtual int format() const = 0;
+
+ /**
+ * return the frame size (number of bytes per sample).
+ */
+ uint32_t frameSize() const {
+ return audio_channel_count_from_out_mask(channels()) *
+ ((format() == AUDIO_FORMAT_PCM_16_BIT) ? sizeof(int16_t) : sizeof(int8_t));
+ }
+
+ /**
+ * return the audio hardware driver latency in milli seconds.
+ */
+ virtual uint32_t latency() const = 0;
+
+ /**
+ * Use this method in situations where audio mixing is done in the
+ * hardware. This method serves as a direct interface with hardware,
+ * allowing you to directly set the volume as apposed to via the framework.
+ * This method might produce multiple PCM outputs or hardware accelerated
+ * codecs, such as MP3 or AAC.
+ */
+ 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;
+
+ /**
+ * Put the audio hardware output into standby mode. Returns
+ * status based on include/utils/Errors.h
+ */
+ virtual status_t standby() = 0;
+
+ /** 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;
+
+ // return the number of audio frames written by the audio dsp to DAC since
+ // the output has exited standby
+ virtual status_t getRenderPosition(uint32_t* dspFrames) = 0;
+
+ /**
+ * get the local time at which the next write to the audio driver will be
+ * presented
+ */
+ virtual status_t getNextWriteTimestamp(int64_t* timestamp);
+
+ /**
+ * Return a recent count of the number of audio frames presented to an external observer.
+ */
+ virtual status_t getPresentationPosition(uint64_t* frames, struct timespec* timestamp);
+};
+
+/**
+ * AudioStreamIn is the abstraction interface for the audio input hardware.
+ *
+ * It defines the various properties of the audio hardware input driver.
+ */
+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 input channel mask */
+ virtual uint32_t channels() const = 0;
+
+ /**
+ * return audio format in 8bit or 16bit PCM format -
+ * eg. AudioSystem:PCM_16_BIT
+ */
+ virtual int format() const = 0;
+
+ /**
+ * return the frame size (number of bytes per sample).
+ */
+ uint32_t frameSize() const {
+ return audio_channel_count_from_in_mask(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 */
+ virtual status_t setGain(float gain) = 0;
+
+ /** read audio buffer in from audio driver */
+ virtual ssize_t read(void* buffer, ssize_t bytes) = 0;
+
+ /** dump the state of the audio input device */
+ virtual status_t dump(int fd, const Vector<String16>& args) = 0;
+
+ /**
+ * Put the audio hardware input into standby mode. Returns
+ * status based on include/utils/Errors.h
+ */
+ 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;
+
+ // Return the number of input frames lost in the audio driver since the last call of this
+ // function. Audio driver is expected to reset the value to 0 and restart counting upon
+ // returning the current value by this function call. Such loss typically occurs when the user
+ // space process is blocked longer than the capacity of audio driver buffers. Unit: the number
+ // of input audio frames
+ virtual unsigned int getInputFramesLost() const = 0;
+
+ virtual status_t addAudioEffect(effect_handle_t effect) = 0;
+ virtual status_t removeAudioEffect(effect_handle_t effect) = 0;
+};
+
+/**
+ * AudioHardwareInterface.h defines the interface to the audio hardware abstraction layer.
+ *
+ * The interface supports setting and getting parameters, selecting audio routing
+ * paths, and defining input and output streams.
+ *
+ * AudioFlinger initializes the audio hardware and immediately opens an output stream.
+ * You can set Audio routing to output to handset, speaker, Bluetooth, or a headset.
+ *
+ * The audio input stream is initialized when AudioFlinger is called to carry out
+ * a record operation.
+ */
+class AudioHardwareInterface {
+ public:
+ virtual ~AudioHardwareInterface() {}
+
+ /**
+ * check to see if the audio hardware interface has been initialized.
+ * return status based on values defined in include/utils/Errors.h
+ */
+ virtual status_t initCheck() = 0;
+
+ /** set the audio volume of a voice call. Range is between 0.0 and 1.0 */
+ virtual status_t setVoiceVolume(float volume) = 0;
+
+ /**
+ * set the audio volume for all audio activities other than voice call.
+ * Range between 0.0 and 1.0. If any value other than NO_ERROR is returned,
+ * the software mixer will emulate this capability.
+ */
+ virtual status_t setMasterVolume(float volume) = 0;
+
+ /**
+ * Get the current master volume value for the HAL, if the HAL supports
+ * master volume control. AudioFlinger will query this value from the
+ * primary audio HAL when the service starts and use the value for setting
+ * the initial master volume across all HALs.
+ */
+ virtual status_t getMasterVolume(float* volume) = 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;
+
+ // mic mute
+ virtual status_t setMicMute(bool state) = 0;
+ virtual status_t getMicMute(bool* state) = 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(uint32_t devices, int* format = 0,
+ uint32_t* channels = 0, uint32_t* sampleRate = 0,
+ status_t* status = 0) = 0;
+ virtual AudioStreamOut* openOutputStreamWithFlags(
+ uint32_t devices, audio_output_flags_t flags = (audio_output_flags_t)0, 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(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;
+
+ virtual status_t setMasterMute(bool muted) = 0;
+
+ static AudioHardwareInterface* create();
+
+ virtual int createAudioPatch(unsigned int num_sources, const struct audio_port_config* sources,
+ unsigned int num_sinks, const struct audio_port_config* sinks,
+ audio_patch_handle_t* handle) = 0;
+
+ virtual int releaseAudioPatch(audio_patch_handle_t handle) = 0;
+
+ virtual int getAudioPort(struct audio_port* port) = 0;
+
+ virtual int setAudioPortConfig(const struct audio_port_config* config) = 0;
+
+ protected:
+ virtual status_t dump(int fd, const Vector<String16>& args) = 0;
+};
+
+// ----------------------------------------------------------------------------
+
+extern "C" AudioHardwareInterface* createAudioHardware(void);
+
+}; // namespace android_audio_legacy
+
+#endif // ANDROID_AUDIO_HARDWARE_INTERFACE_H