776 lines
25 KiB
C
776 lines
25 KiB
C
/*
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* Copyright (c) 2013-2019, The Linux Foundation. All rights reserved.
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* Not a contribution.
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*
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* Copyright (C) 2013 The Android Open Source Project
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*
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* This file was modified by DTS, Inc. The portions of the
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* code modified by DTS, Inc are copyrighted and
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* licensed separately, as follows:
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*
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* (C) 2014 DTS, Inc.
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*
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* Licensed under the Apache License, Version 2.0 (the "License");
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* you may not use this file except in compliance with the License.
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* You may obtain a copy of the License at
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*
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* http://www.apache.org/licenses/LICENSE-2.0
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*
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* Unless required by applicable law or agreed to in writing, software
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* distributed under the License is distributed on an "AS IS" BASIS,
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* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
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* See the License for the specific language governing permissions and
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* limitations under the License.
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*/
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#ifndef QCOM_AUDIO_HW_H
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#define QCOM_AUDIO_HW_H
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#include <stdlib.h>
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#include <cutils/str_parms.h>
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#include <cutils/list.h>
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#include <hardware/audio.h>
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#include <tinyalsa/asoundlib.h>
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#include <tinycompress/tinycompress.h>
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#include <audio_route/audio_route.h>
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#include <audio_utils/ErrorLog.h>
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#include "audio_defs.h"
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#include "voice.h"
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#include "audio_hw_extn_api.h"
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#if LINUX_ENABLED
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#if defined(__LP64__)
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#define VISUALIZER_LIBRARY_PATH "/usr/lib64/libqcomvisualizer.so"
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#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/usr/lib64/libqcompostprocbundle.so"
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#define ADM_LIBRARY_PATH "/usr/lib64/libadm.so"
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#else
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#define VISUALIZER_LIBRARY_PATH "/usr/lib/libqcomvisualizer.so"
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#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/usr/lib/libqcompostprocbundle.so"
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#define ADM_LIBRARY_PATH "/usr/lib/libadm.so"
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#endif
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#else
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#define VISUALIZER_LIBRARY_PATH "/vendor/lib/soundfx/libqcomvisualizer.so"
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#define OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH "/vendor/lib/soundfx/libqcompostprocbundle.so"
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#define ADM_LIBRARY_PATH "/vendor/lib/libadm.so"
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#endif
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/* Flags used to initialize acdb_settings variable that goes to ACDB library */
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#define NONE_FLAG 0x00000000
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#define ANC_FLAG 0x00000001
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#define DMIC_FLAG 0x00000002
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#define QMIC_FLAG 0x00000004
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/* Include TMIC Flag after existing QMIC flag to avoid backward compatibility
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* issues since they are bit masked */
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#define TMIC_FLAG 0x00000008
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#define TTY_MODE_OFF 0x00000010
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#define TTY_MODE_FULL 0x00000020
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#define TTY_MODE_VCO 0x00000040
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#define TTY_MODE_HCO 0x00000080
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#define TTY_MODE_CLEAR 0xFFFFFF0F
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#define FLUENCE_MODE_CLEAR 0xFFFFFFF0
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#define ACDB_DEV_TYPE_OUT 1
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#define ACDB_DEV_TYPE_IN 2
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/* SCO SWB codec mode */
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#define SPEECH_MODE_INVALID 0xFFFF
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/* support positional and index masks to 8ch */
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#define MAX_SUPPORTED_CHANNEL_MASKS (2 * FCC_8)
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#define MAX_SUPPORTED_FORMATS 15
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#define MAX_SUPPORTED_SAMPLE_RATES 7
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#define DEFAULT_HDMI_OUT_CHANNELS 2
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#define DEFAULT_HDMI_OUT_SAMPLE_RATE 48000
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#define DEFAULT_HDMI_OUT_FORMAT AUDIO_FORMAT_PCM_16_BIT
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#define ERROR_LOG_ENTRIES 16
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#define SND_CARD_STATE_OFFLINE 0
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#define SND_CARD_STATE_ONLINE 1
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#define STREAM_DIRECTION_IN 0
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#define STREAM_DIRECTION_OUT 1
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#define MAX_PERF_LOCK_OPTS 20
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#define MAX_STREAM_PROFILE_STR_LEN 32
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typedef enum {
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EFFECT_NONE = 0,
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EFFECT_AEC,
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EFFECT_NS,
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EFFECT_MAX
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} effect_type_t;
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struct audio_effect_config {
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uint32_t module_id;
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uint32_t instance_id;
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uint32_t param_id;
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uint32_t param_value;
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};
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#define MAX_MIXER_PATH_LEN 64
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typedef enum card_status_t {
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CARD_STATUS_OFFLINE,
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CARD_STATUS_ONLINE
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} card_status_t;
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/* These are the supported use cases by the hardware.
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* Each usecase is mapped to a specific PCM device.
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* Refer to pcm_device_table[].
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*/
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enum {
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USECASE_INVALID = -1,
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/* Playback usecases */
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USECASE_AUDIO_PLAYBACK_DEEP_BUFFER = 0,
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USECASE_AUDIO_PLAYBACK_LOW_LATENCY,
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USECASE_AUDIO_PLAYBACK_MULTI_CH,
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USECASE_AUDIO_PLAYBACK_OFFLOAD,
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USECASE_AUDIO_PLAYBACK_OFFLOAD2,
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USECASE_AUDIO_PLAYBACK_OFFLOAD3,
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USECASE_AUDIO_PLAYBACK_OFFLOAD4,
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USECASE_AUDIO_PLAYBACK_OFFLOAD5,
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USECASE_AUDIO_PLAYBACK_OFFLOAD6,
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USECASE_AUDIO_PLAYBACK_OFFLOAD7,
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USECASE_AUDIO_PLAYBACK_OFFLOAD8,
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USECASE_AUDIO_PLAYBACK_OFFLOAD9,
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USECASE_AUDIO_PLAYBACK_ULL,
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USECASE_AUDIO_PLAYBACK_MMAP,
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USECASE_AUDIO_PLAYBACK_WITH_HAPTICS,
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USECASE_AUDIO_PLAYBACK_HIFI,
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USECASE_AUDIO_PLAYBACK_TTS,
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/* FM usecase */
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USECASE_AUDIO_PLAYBACK_FM,
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/* HFP Use case*/
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USECASE_AUDIO_HFP_SCO,
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USECASE_AUDIO_HFP_SCO_WB,
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USECASE_AUDIO_HFP_SCO_DOWNLINK,
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USECASE_AUDIO_HFP_SCO_WB_DOWNLINK,
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/* Capture usecases */
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USECASE_AUDIO_RECORD,
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USECASE_AUDIO_RECORD_COMPRESS,
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USECASE_AUDIO_RECORD_COMPRESS2,
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USECASE_AUDIO_RECORD_COMPRESS3,
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USECASE_AUDIO_RECORD_COMPRESS4,
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USECASE_AUDIO_RECORD_COMPRESS5,
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USECASE_AUDIO_RECORD_COMPRESS6,
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USECASE_AUDIO_RECORD_LOW_LATENCY,
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USECASE_AUDIO_RECORD_FM_VIRTUAL,
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USECASE_AUDIO_RECORD_HIFI,
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USECASE_AUDIO_PLAYBACK_VOIP,
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USECASE_AUDIO_RECORD_VOIP,
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/* Voice usecase */
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USECASE_VOICE_CALL,
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USECASE_AUDIO_RECORD_MMAP,
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/* Voice extension usecases */
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USECASE_VOICE2_CALL,
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USECASE_VOLTE_CALL,
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USECASE_QCHAT_CALL,
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USECASE_VOWLAN_CALL,
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USECASE_VOICEMMODE1_CALL,
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USECASE_VOICEMMODE2_CALL,
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USECASE_COMPRESS_VOIP_CALL,
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USECASE_INCALL_REC_UPLINK,
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USECASE_INCALL_REC_DOWNLINK,
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USECASE_INCALL_REC_UPLINK_AND_DOWNLINK,
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USECASE_INCALL_REC_UPLINK_COMPRESS,
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USECASE_INCALL_REC_DOWNLINK_COMPRESS,
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USECASE_INCALL_REC_UPLINK_AND_DOWNLINK_COMPRESS,
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USECASE_INCALL_MUSIC_UPLINK,
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USECASE_INCALL_MUSIC_UPLINK2,
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USECASE_AUDIO_SPKR_CALIB_RX,
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USECASE_AUDIO_SPKR_CALIB_TX,
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USECASE_AUDIO_PLAYBACK_AFE_PROXY,
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USECASE_AUDIO_RECORD_AFE_PROXY,
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USECASE_AUDIO_DSM_FEEDBACK,
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USECASE_AUDIO_PLAYBACK_SILENCE,
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USECASE_AUDIO_TRANSCODE_LOOPBACK_RX,
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USECASE_AUDIO_TRANSCODE_LOOPBACK_TX,
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USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM1,
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USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM2,
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USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM3,
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USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM4,
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USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM5,
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USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM6,
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USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM7,
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USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM8,
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USECASE_AUDIO_EC_REF_LOOPBACK,
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USECASE_AUDIO_A2DP_ABR_FEEDBACK,
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/* car streams usecases */
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USECASE_AUDIO_PLAYBACK_MEDIA,
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USECASE_AUDIO_PLAYBACK_SYS_NOTIFICATION,
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USECASE_AUDIO_PLAYBACK_NAV_GUIDANCE,
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USECASE_AUDIO_PLAYBACK_PHONE,
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USECASE_AUDIO_PLAYBACK_REAR_SEAT,
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/*Audio FM Tuner usecase*/
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USECASE_AUDIO_FM_TUNER_EXT,
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AUDIO_USECASE_MAX
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};
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const char * const use_case_table[AUDIO_USECASE_MAX];
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#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
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/*
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* tinyAlsa library interprets period size as number of frames
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* one frame = channel_count * sizeof (pcm sample)
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* so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes
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* DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes
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* We should take care of returning proper size when AudioFlinger queries for
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* the buffer size of an input/output stream
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*/
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enum {
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OFFLOAD_CMD_EXIT, /* exit compress offload thread loop*/
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OFFLOAD_CMD_DRAIN, /* send a full drain request to DSP */
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OFFLOAD_CMD_PARTIAL_DRAIN, /* send a partial drain request to DSP */
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OFFLOAD_CMD_WAIT_FOR_BUFFER, /* wait for buffer released by DSP */
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OFFLOAD_CMD_ERROR, /* offload playback hit some error */
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};
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/*
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* Camera selection indicated via set_parameters "cameraFacing=front|back and
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* "rotation=0|90|180|270""
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*/
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enum {
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CAMERA_FACING_BACK = 0x0,
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CAMERA_FACING_FRONT = 0x1,
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CAMERA_FACING_MASK = 0x0F,
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CAMERA_ROTATION_LANDSCAPE = 0x0,
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CAMERA_ROTATION_INVERT_LANDSCAPE = 0x10,
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CAMERA_ROTATION_PORTRAIT = 0x20,
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CAMERA_ROTATION_MASK = 0xF0,
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CAMERA_BACK_LANDSCAPE = (CAMERA_FACING_BACK|CAMERA_ROTATION_LANDSCAPE),
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CAMERA_BACK_INVERT_LANDSCAPE = (CAMERA_FACING_BACK|CAMERA_ROTATION_INVERT_LANDSCAPE),
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CAMERA_BACK_PORTRAIT = (CAMERA_FACING_BACK|CAMERA_ROTATION_PORTRAIT),
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CAMERA_FRONT_LANDSCAPE = (CAMERA_FACING_FRONT|CAMERA_ROTATION_LANDSCAPE),
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CAMERA_FRONT_INVERT_LANDSCAPE = (CAMERA_FACING_FRONT|CAMERA_ROTATION_INVERT_LANDSCAPE),
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CAMERA_FRONT_PORTRAIT = (CAMERA_FACING_FRONT|CAMERA_ROTATION_PORTRAIT),
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CAMERA_DEFAULT = CAMERA_BACK_LANDSCAPE,
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};
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//FIXME: to be replaced by proper video capture properties API
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#define AUDIO_PARAMETER_KEY_CAMERA_FACING "cameraFacing"
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#define AUDIO_PARAMETER_VALUE_FRONT "front"
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#define AUDIO_PARAMETER_VALUE_BACK "back"
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enum {
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OFFLOAD_STATE_IDLE,
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OFFLOAD_STATE_PLAYING,
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OFFLOAD_STATE_PAUSED,
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};
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struct offload_cmd {
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struct listnode node;
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int cmd;
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int data[];
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};
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typedef enum render_mode {
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RENDER_MODE_AUDIO_NO_TIMESTAMP = 0,
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RENDER_MODE_AUDIO_MASTER,
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RENDER_MODE_AUDIO_STC_MASTER,
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} render_mode_t;
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/* This defines the physical car streams supported in audio HAL,
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* limited by the available frontend PCM driver.
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* Max number of physical streams supported is currently 16 and is
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* represented by stream bit flag.
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*/
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#define MAX_CAR_AUDIO_STREAMS 16
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enum {
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CAR_AUDIO_STREAM_MEDIA = 0x1,
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CAR_AUDIO_STREAM_SYS_NOTIFICATION = 0x2,
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CAR_AUDIO_STREAM_NAV_GUIDANCE = 0x4,
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CAR_AUDIO_STREAM_PHONE = 0x8,
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CAR_AUDIO_STREAM_REAR_SEAT = 0x100,
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};
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struct stream_app_type_cfg {
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int sample_rate;
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uint32_t bit_width;
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int app_type;
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int gain[2];
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};
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struct stream_config {
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unsigned int sample_rate;
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audio_channel_mask_t channel_mask;
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audio_format_t format;
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audio_devices_t devices;
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unsigned int bit_width;
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};
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struct stream_inout {
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pthread_mutex_t lock; /* see note below on mutex acquisition order */
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pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
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pthread_cond_t cond;
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struct stream_config in_config;
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struct stream_config out_config;
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struct stream_app_type_cfg out_app_type_cfg;
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char profile[MAX_STREAM_PROFILE_STR_LEN];
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struct audio_device *dev;
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void *adsp_hdlr_stream_handle;
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void *ip_hdlr_handle;
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stream_callback_t client_callback;
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void *client_cookie;
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};
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struct stream_out {
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struct audio_stream_out stream;
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pthread_mutex_t lock; /* see note below on mutex acquisition order */
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pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
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pthread_mutex_t compr_mute_lock; /* acquire before setting compress volume */
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pthread_mutex_t position_query_lock; /* acquire before updating/getting position of track offload*/
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pthread_cond_t cond;
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struct pcm_config config;
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struct compr_config compr_config;
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struct pcm *pcm;
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struct compress *compr;
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int standby;
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int pcm_device_id;
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unsigned int sample_rate;
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audio_channel_mask_t channel_mask;
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audio_format_t format;
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audio_devices_t devices;
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audio_output_flags_t flags;
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char profile[MAX_STREAM_PROFILE_STR_LEN];
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audio_usecase_t usecase;
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/* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
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audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
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audio_format_t supported_formats[MAX_SUPPORTED_FORMATS+1];
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uint32_t supported_sample_rates[MAX_SUPPORTED_SAMPLE_RATES+1];
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bool muted;
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uint64_t written; /* total frames written, not cleared when entering standby */
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int64_t mmap_time_offset_nanos; /* fudge factor to correct inaccuracies in DSP */
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int mmap_shared_memory_fd; /* file descriptor associated with MMAP NOIRQ shared memory */
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audio_io_handle_t handle;
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struct stream_app_type_cfg app_type_cfg;
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int non_blocking;
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int playback_started;
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int offload_state;
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pthread_cond_t offload_cond;
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pthread_t offload_thread;
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struct listnode offload_cmd_list;
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bool offload_thread_blocked;
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struct timespec writeAt;
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void *adsp_hdlr_stream_handle;
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void *ip_hdlr_handle;
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stream_callback_t client_callback;
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void *client_cookie;
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struct compr_gapless_mdata gapless_mdata;
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int send_new_metadata;
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bool send_next_track_params;
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bool is_compr_metadata_avail;
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unsigned int bit_width;
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uint32_t hal_fragment_size;
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audio_format_t hal_ip_format;
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audio_format_t hal_op_format;
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void *convert_buffer;
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bool realtime;
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int af_period_multiplier;
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struct audio_device *dev;
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card_status_t card_status;
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void* qaf_stream_handle;
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void* qap_stream_handle;
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pthread_cond_t qaf_offload_cond;
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pthread_t qaf_offload_thread;
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struct listnode qaf_offload_cmd_list;
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uint32_t platform_latency;
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render_mode_t render_mode;
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bool drift_correction_enabled;
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struct audio_out_channel_map_param channel_map_param; /* input channel map */
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audio_offload_info_t info;
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int started;
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qahwi_stream_out_t qahwi_out;
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bool is_iec61937_info_available;
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bool a2dp_compress_mute;
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float volume_l;
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float volume_r;
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bool apply_volume;
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char pm_qos_mixer_path[MAX_MIXER_PATH_LEN];
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int hal_output_suspend_supported;
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int dynamic_pm_qos_config_supported;
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bool stream_config_changed;
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mix_matrix_params_t pan_scale_params;
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mix_matrix_params_t downmix_params;
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bool set_dual_mono;
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bool prev_card_status_offline;
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error_log_t *error_log;
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bool pspd_coeff_sent;
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char address[AUDIO_DEVICE_MAX_ADDRESS_LEN];
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int car_audio_stream;
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union {
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char *addr;
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struct {
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int controller;
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int stream;
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} cs;
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} extconn;
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};
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struct stream_in {
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struct audio_stream_in stream;
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pthread_mutex_t lock; /* see note below on mutex acquisition order */
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pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
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struct pcm_config config;
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struct pcm *pcm;
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int standby;
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int source;
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int pcm_device_id;
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audio_devices_t device;
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audio_channel_mask_t channel_mask;
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audio_usecase_t usecase;
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bool enable_aec;
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bool enable_ns;
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audio_format_t format;
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bool enable_ec_port;
|
|
bool ec_opened;
|
|
struct listnode aec_list;
|
|
struct listnode ns_list;
|
|
int64_t mmap_time_offset_nanos; /* fudge factor to correct inaccuracies in DSP */
|
|
int mmap_shared_memory_fd; /* file descriptor associated with MMAP NOIRQ shared memory */
|
|
audio_io_handle_t capture_handle;
|
|
audio_input_flags_t flags;
|
|
char profile[MAX_STREAM_PROFILE_STR_LEN];
|
|
bool is_st_session;
|
|
bool is_st_session_active;
|
|
unsigned int sample_rate;
|
|
unsigned int bit_width;
|
|
bool realtime;
|
|
int af_period_multiplier;
|
|
struct stream_app_type_cfg app_type_cfg;
|
|
void *cin_extn;
|
|
qahwi_stream_in_t qahwi_in;
|
|
|
|
struct audio_device *dev;
|
|
card_status_t card_status;
|
|
int capture_started;
|
|
float zoom;
|
|
audio_microphone_direction_t direction;
|
|
|
|
volatile int32_t capture_stopped;
|
|
|
|
/* Array of supported channel mask configurations. +1 so that the last entry is always 0 */
|
|
audio_channel_mask_t supported_channel_masks[MAX_SUPPORTED_CHANNEL_MASKS + 1];
|
|
audio_format_t supported_formats[MAX_SUPPORTED_FORMATS + 1];
|
|
uint32_t supported_sample_rates[MAX_SUPPORTED_SAMPLE_RATES + 1];
|
|
|
|
int64_t frames_read; /* total frames read, not cleared when entering standby */
|
|
int64_t frames_muted; /* total frames muted, not cleared when entering standby */
|
|
|
|
error_log_t *error_log;
|
|
};
|
|
|
|
typedef enum {
|
|
PCM_PLAYBACK,
|
|
PCM_CAPTURE,
|
|
VOICE_CALL,
|
|
VOIP_CALL,
|
|
PCM_HFP_CALL,
|
|
TRANSCODE_LOOPBACK_RX,
|
|
TRANSCODE_LOOPBACK_TX,
|
|
PCM_PASSTHROUGH,
|
|
USECASE_TYPE_MAX
|
|
} usecase_type_t;
|
|
|
|
union stream_ptr {
|
|
struct stream_in *in;
|
|
struct stream_out *out;
|
|
struct stream_inout *inout;
|
|
};
|
|
|
|
struct audio_usecase {
|
|
struct listnode list;
|
|
audio_usecase_t id;
|
|
usecase_type_t type;
|
|
audio_devices_t devices;
|
|
snd_device_t out_snd_device;
|
|
snd_device_t in_snd_device;
|
|
struct stream_app_type_cfg out_app_type_cfg;
|
|
struct stream_app_type_cfg in_app_type_cfg;
|
|
union stream_ptr stream;
|
|
};
|
|
|
|
struct stream_format {
|
|
struct listnode list;
|
|
audio_format_t format;
|
|
};
|
|
|
|
struct stream_sample_rate {
|
|
struct listnode list;
|
|
uint32_t sample_rate;
|
|
};
|
|
|
|
typedef union {
|
|
audio_output_flags_t out_flags;
|
|
audio_input_flags_t in_flags;
|
|
} audio_io_flags_t;
|
|
|
|
struct streams_io_cfg {
|
|
struct listnode list;
|
|
audio_io_flags_t flags;
|
|
char profile[MAX_STREAM_PROFILE_STR_LEN];
|
|
struct listnode format_list;
|
|
struct listnode sample_rate_list;
|
|
struct stream_app_type_cfg app_type_cfg;
|
|
};
|
|
|
|
typedef struct streams_input_ctxt {
|
|
struct listnode list;
|
|
struct stream_in *input;
|
|
} streams_input_ctxt_t;
|
|
|
|
typedef struct streams_output_ctxt {
|
|
struct listnode list;
|
|
struct stream_out *output;
|
|
} streams_output_ctxt_t;
|
|
|
|
typedef void* (*adm_init_t)();
|
|
typedef void (*adm_deinit_t)(void *);
|
|
typedef void (*adm_register_output_stream_t)(void *, audio_io_handle_t, audio_output_flags_t);
|
|
typedef void (*adm_register_input_stream_t)(void *, audio_io_handle_t, audio_input_flags_t);
|
|
typedef void (*adm_deregister_stream_t)(void *, audio_io_handle_t);
|
|
typedef void (*adm_request_focus_t)(void *, audio_io_handle_t);
|
|
typedef void (*adm_abandon_focus_t)(void *, audio_io_handle_t);
|
|
typedef void (*adm_set_config_t)(void *, audio_io_handle_t,
|
|
struct pcm *,
|
|
struct pcm_config *);
|
|
typedef void (*adm_request_focus_v2_t)(void *, audio_io_handle_t, long);
|
|
typedef bool (*adm_is_noirq_avail_t)(void *, int, int, int);
|
|
typedef void (*adm_on_routing_change_t)(void *, audio_io_handle_t);
|
|
typedef int (*adm_request_focus_v2_1_t)(void *, audio_io_handle_t, long);
|
|
|
|
struct audio_device {
|
|
struct audio_hw_device device;
|
|
|
|
pthread_mutex_t lock; /* see note below on mutex acquisition order */
|
|
pthread_mutex_t cal_lock;
|
|
struct mixer *mixer;
|
|
audio_mode_t mode;
|
|
audio_devices_t out_device;
|
|
struct stream_out *primary_output;
|
|
struct stream_out *voice_tx_output;
|
|
struct stream_out *current_call_output;
|
|
bool bluetooth_nrec;
|
|
bool screen_off;
|
|
int *snd_dev_ref_cnt;
|
|
struct listnode usecase_list;
|
|
struct listnode streams_output_cfg_list;
|
|
struct listnode streams_input_cfg_list;
|
|
struct audio_route *audio_route;
|
|
int acdb_settings;
|
|
bool speaker_lr_swap;
|
|
struct voice voice;
|
|
unsigned int cur_hdmi_channels;
|
|
audio_format_t cur_hdmi_format;
|
|
unsigned int cur_hdmi_sample_rate;
|
|
unsigned int cur_hdmi_bit_width;
|
|
unsigned int cur_wfd_channels;
|
|
bool bt_wb_speech_enabled;
|
|
unsigned int swb_speech_mode;
|
|
bool allow_afe_proxy_usage;
|
|
bool is_charging; // from battery listener
|
|
bool mic_break_enabled;
|
|
bool enable_hfp;
|
|
bool mic_muted;
|
|
bool enable_voicerx;
|
|
unsigned int num_va_sessions;
|
|
|
|
int snd_card;
|
|
card_status_t card_status;
|
|
unsigned int cur_codec_backend_samplerate;
|
|
unsigned int cur_codec_backend_bit_width;
|
|
bool is_channel_status_set;
|
|
void *platform;
|
|
void *extspk;
|
|
unsigned int offload_usecases_state;
|
|
unsigned int pcm_record_uc_state;
|
|
void *visualizer_lib;
|
|
int (*visualizer_start_output)(audio_io_handle_t, int);
|
|
int (*visualizer_stop_output)(audio_io_handle_t, int);
|
|
void *offload_effects_lib;
|
|
int (*offload_effects_start_output)(audio_io_handle_t, int, struct mixer *);
|
|
int (*offload_effects_stop_output)(audio_io_handle_t, int);
|
|
|
|
int (*offload_effects_set_hpx_state)(bool);
|
|
|
|
void *adm_data;
|
|
void *adm_lib;
|
|
adm_init_t adm_init;
|
|
adm_deinit_t adm_deinit;
|
|
adm_register_input_stream_t adm_register_input_stream;
|
|
adm_register_output_stream_t adm_register_output_stream;
|
|
adm_deregister_stream_t adm_deregister_stream;
|
|
adm_request_focus_t adm_request_focus;
|
|
adm_abandon_focus_t adm_abandon_focus;
|
|
adm_set_config_t adm_set_config;
|
|
adm_request_focus_v2_t adm_request_focus_v2;
|
|
adm_is_noirq_avail_t adm_is_noirq_avail;
|
|
adm_on_routing_change_t adm_on_routing_change;
|
|
adm_request_focus_v2_1_t adm_request_focus_v2_1;
|
|
|
|
void (*offload_effects_get_parameters)(struct str_parms *,
|
|
struct str_parms *);
|
|
void (*offload_effects_set_parameters)(struct str_parms *);
|
|
|
|
bool multi_offload_enable;
|
|
int perf_lock_handle;
|
|
int perf_lock_opts[MAX_PERF_LOCK_OPTS];
|
|
int perf_lock_opts_size;
|
|
bool native_playback_enabled;
|
|
bool asrc_mode_enabled;
|
|
qahwi_device_t qahwi_dev;
|
|
bool vr_audio_mode_enabled;
|
|
uint32_t dsp_bit_width_enforce_mode;
|
|
bool bt_sco_on;
|
|
struct audio_device_config_param *device_cfg_params;
|
|
unsigned int interactive_usecase_state;
|
|
bool dp_allowed_for_voice;
|
|
void *ext_hw_plugin;
|
|
|
|
struct pcm_config haptics_config;
|
|
struct pcm *haptic_pcm;
|
|
int haptic_pcm_device_id;
|
|
uint8_t *haptic_buffer;
|
|
size_t haptic_buffer_size;
|
|
|
|
/* logging */
|
|
snd_device_t last_logged_snd_device[AUDIO_USECASE_MAX][2]; /* [out, in] */
|
|
|
|
/* The pcm_params use_case_table is loaded by adev_verify_devices() upon
|
|
* calling adev_open().
|
|
*
|
|
* If an entry is not NULL, it can be used to determine if extended precision
|
|
* or other capabilities are present for the device corresponding to that usecase.
|
|
*/
|
|
struct pcm_params *use_case_table[AUDIO_USECASE_MAX];
|
|
struct listnode active_inputs_list;
|
|
struct listnode active_outputs_list;
|
|
bool use_old_pspd_mix_ctrl;
|
|
int camera_orientation; /* CAMERA_BACK_LANDSCAPE ... CAMERA_FRONT_PORTRAIT */
|
|
bool adm_routing_changed;
|
|
struct listnode audio_patch_record_list;
|
|
unsigned int audio_patch_index;
|
|
};
|
|
|
|
struct audio_patch_record {
|
|
struct listnode list;
|
|
audio_patch_handle_t handle;
|
|
audio_usecase_t usecase;
|
|
audio_io_handle_t input_io_handle;
|
|
audio_io_handle_t output_io_handle;
|
|
struct audio_port_config source;
|
|
struct audio_port_config sink;
|
|
};
|
|
|
|
int select_devices(struct audio_device *adev,
|
|
audio_usecase_t uc_id);
|
|
int disable_audio_route(struct audio_device *adev,
|
|
struct audio_usecase *usecase);
|
|
int disable_snd_device(struct audio_device *adev,
|
|
snd_device_t snd_device);
|
|
int enable_snd_device(struct audio_device *adev,
|
|
snd_device_t snd_device);
|
|
|
|
int enable_audio_route(struct audio_device *adev,
|
|
struct audio_usecase *usecase);
|
|
|
|
struct audio_usecase *get_usecase_from_list(const struct audio_device *adev,
|
|
audio_usecase_t uc_id);
|
|
|
|
bool is_offload_usecase(audio_usecase_t uc_id);
|
|
|
|
bool audio_is_true_native_stream_active(struct audio_device *adev);
|
|
|
|
bool audio_is_dsd_native_stream_active(struct audio_device *adev);
|
|
|
|
uint32_t adev_get_dsp_bit_width_enforce_mode();
|
|
|
|
int pcm_ioctl(struct pcm *pcm, int request, ...);
|
|
|
|
audio_usecase_t get_usecase_id_from_usecase_type(const struct audio_device *adev,
|
|
usecase_type_t type);
|
|
|
|
int check_a2dp_restore(struct audio_device *adev, struct stream_out *out, bool restore);
|
|
|
|
int adev_open_output_stream(struct audio_hw_device *dev,
|
|
audio_io_handle_t handle,
|
|
audio_devices_t devices,
|
|
audio_output_flags_t flags,
|
|
struct audio_config *config,
|
|
struct audio_stream_out **stream_out,
|
|
const char *address __unused);
|
|
void adev_close_output_stream(struct audio_hw_device *dev __unused,
|
|
struct audio_stream_out *stream);
|
|
|
|
bool is_interactive_usecase(audio_usecase_t uc_id);
|
|
|
|
streams_input_ctxt_t *in_get_stream(struct audio_device *dev,
|
|
audio_io_handle_t input);
|
|
streams_output_ctxt_t *out_get_stream(struct audio_device *dev,
|
|
audio_io_handle_t output);
|
|
|
|
size_t get_output_period_size(uint32_t sample_rate,
|
|
audio_format_t format,
|
|
int channel_count,
|
|
int duration /*in millisecs*/);
|
|
|
|
#define LITERAL_TO_STRING(x) #x
|
|
#define CHECK(condition) LOG_ALWAYS_FATAL_IF(!(condition), "%s",\
|
|
__FILE__ ":" LITERAL_TO_STRING(__LINE__)\
|
|
" ASSERT_FATAL(" #condition ") failed.")
|
|
|
|
static inline bool is_loopback_input_device(audio_devices_t device) {
|
|
if (!audio_is_output_device(device) &&
|
|
((device & AUDIO_DEVICE_IN_LOOPBACK) == AUDIO_DEVICE_IN_LOOPBACK))
|
|
return true;
|
|
else
|
|
return false;
|
|
}
|
|
|
|
/*
|
|
* NOTE: when multiple mutexes have to be acquired, always take the
|
|
* stream_in or stream_out mutex first, followed by the audio_device mutex.
|
|
*/
|
|
|
|
#endif // QCOM_AUDIO_HW_H
|