Initial mpq8092 HAL upload
Initial mpq8092 HAL upload
Depends-on: 531569 531557
Change-Id: Ic130ab0a5ae2ffee09d98b7ca2c3ee4374965466
diff --git a/hal_mpq/audio_hw.h b/hal_mpq/audio_hw.h
new file mode 100644
index 0000000..262fda8
--- /dev/null
+++ b/hal_mpq/audio_hw.h
@@ -0,0 +1,244 @@
+/*
+ * Copyright (c) 2013, The Linux Foundation. All rights reserved.
+ * Not a contribution.
+ *
+ * Copyright (C) 2013 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 QCOM_AUDIO_HW_H
+#define QCOM_AUDIO_HW_H
+
+#include <cutils/list.h>
+#include <hardware/audio.h>
+#include <tinyalsa/asoundlib.h>
+#include <tinycompress/tinycompress.h>
+
+#include <audio_route/audio_route.h>
+
+#define VISUALIZER_LIBRARY_PATH "/system/lib/soundfx/libqcomvisualizer.so"
+
+/* Flags used to initialize acdb_settings variable that goes to ACDB library */
+#define DMIC_FLAG 0x00000002
+#define QMIC_FLAG 0x00000004
+#define TTY_MODE_OFF 0x00000010
+#define TTY_MODE_FULL 0x00000020
+#define TTY_MODE_VCO 0x00000040
+#define TTY_MODE_HCO 0x00000080
+#define TTY_MODE_CLEAR 0xFFFFFF0F
+
+#define ACDB_DEV_TYPE_OUT 1
+#define ACDB_DEV_TYPE_IN 2
+
+#define MAX_SUPPORTED_CHANNEL_MASKS 2
+#define DEFAULT_HDMI_OUT_CHANNELS 2
+
+typedef int snd_device_t;
+
+/* These are the supported use cases by the hardware.
+ * Each usecase is mapped to a specific PCM device.
+ * Refer to pcm_device_table[].
+ */
+typedef enum {
+ USECASE_INVALID = -1,
+ /* Playback usecases */
+ USECASE_AUDIO_PLAYBACK_DEEP_BUFFER = 0,
+ USECASE_AUDIO_PLAYBACK_LOW_LATENCY,
+ USECASE_AUDIO_PLAYBACK_MULTI_CH,
+ USECASE_AUDIO_PLAYBACK_OFFLOAD,
+
+ /* FM usecase */
+ USECASE_AUDIO_PLAYBACK_FM,
+
+ /* Capture usecases */
+ USECASE_AUDIO_RECORD,
+ USECASE_AUDIO_RECORD_COMPRESS,
+ USECASE_AUDIO_RECORD_LOW_LATENCY,
+ USECASE_AUDIO_RECORD_FM_VIRTUAL,
+
+ /* Voice usecase */
+ USECASE_VOICE_CALL,
+
+ /* Voice extension usecases */
+ USECASE_VOICE2_CALL,
+ USECASE_VOLTE_CALL,
+ USECASE_QCHAT_CALL,
+ USECASE_COMPRESS_VOIP_CALL,
+
+ USECASE_INCALL_REC_UPLINK,
+ USECASE_INCALL_REC_DOWNLINK,
+ USECASE_INCALL_REC_UPLINK_AND_DOWNLINK,
+
+ USECASE_INCALL_MUSIC_UPLINK,
+ USECASE_INCALL_MUSIC_UPLINK2,
+
+ USECASE_AUDIO_SPKR_CALIB_RX,
+ USECASE_AUDIO_SPKR_CALIB_TX,
+ AUDIO_USECASE_MAX
+} audio_usecase_t;
+
+#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0]))
+
+/*
+ * tinyAlsa library interprets period size as number of frames
+ * one frame = channel_count * sizeof (pcm sample)
+ * so if format = 16-bit PCM and channels = Stereo, frame size = 2 ch * 2 = 4 bytes
+ * DEEP_BUFFER_OUTPUT_PERIOD_SIZE = 1024 means 1024 * 4 = 4096 bytes
+ * We should take care of returning proper size when AudioFlinger queries for
+ * the buffer size of an input/output stream
+ */
+
+enum {
+ OFFLOAD_CMD_EXIT, /* exit compress offload thread loop*/
+ OFFLOAD_CMD_DRAIN, /* send a full drain request to DSP */
+ OFFLOAD_CMD_PARTIAL_DRAIN, /* send a partial drain request to DSP */
+ OFFLOAD_CMD_WAIT_FOR_BUFFER, /* wait for buffer released by DSP */
+};
+
+enum {
+ OFFLOAD_STATE_IDLE,
+ OFFLOAD_STATE_PLAYING,
+ OFFLOAD_STATE_PAUSED,
+};
+
+struct offload_cmd {
+ struct listnode node;
+ int cmd;
+ int data[];
+};
+
+struct stream_out {
+ struct audio_stream_out stream;
+ pthread_mutex_t lock; /* see note below on mutex acquisition order */
+ pthread_cond_t cond;
+ struct pcm_config config;
+ struct compr_config compr_config;
+ struct pcm *pcm;
+ struct compress *compr;
+ int standby;
+ int pcm_device_id;
+ unsigned int sample_rate;
+ audio_channel_mask_t channel_mask;
+ audio_format_t format;
+ audio_devices_t devices;
+ audio_output_flags_t flags;
+ audio_usecase_t usecase;
+ /* 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];
+ bool muted;
+ uint64_t written; /* total frames written, not cleared when entering standby */
+ audio_io_handle_t handle;
+
+ int non_blocking;
+ int playback_started;
+ int offload_state;
+ pthread_cond_t offload_cond;
+ pthread_t offload_thread;
+ struct listnode offload_cmd_list;
+ bool offload_thread_blocked;
+
+ stream_callback_t offload_callback;
+ void *offload_cookie;
+ struct compr_gapless_mdata gapless_mdata;
+ int send_new_metadata;
+
+ struct audio_device *dev;
+};
+
+struct stream_in {
+ struct audio_stream_in stream;
+ pthread_mutex_t lock; /* see note below on mutex acquisition order */
+ struct pcm_config config;
+ struct pcm *pcm;
+ int standby;
+ int source;
+ int pcm_device_id;
+ int device;
+ audio_channel_mask_t channel_mask;
+ audio_usecase_t usecase;
+ bool enable_aec;
+ bool enable_ns;
+ audio_format_t format;
+
+ struct audio_device *dev;
+};
+
+typedef enum {
+ PCM_PLAYBACK,
+ PCM_CAPTURE,
+ VOICE_CALL,
+ VOIP_CALL
+} usecase_type_t;
+
+union stream_ptr {
+ struct stream_in *in;
+ struct stream_out *out;
+};
+
+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;
+ union stream_ptr stream;
+};
+
+struct audio_device {
+ struct audio_hw_device device;
+ pthread_mutex_t lock; /* see note below on mutex acquisition order */
+ struct mixer *mixer;
+ audio_mode_t mode;
+ audio_devices_t out_device;
+ struct stream_in *active_input;
+ struct stream_out *primary_output;
+ bool bluetooth_nrec;
+ bool screen_off;
+ int *snd_dev_ref_cnt;
+ struct listnode usecase_list;
+ struct audio_route *audio_route;
+ int acdb_settings;
+ bool speaker_lr_swap;
+ unsigned int cur_hdmi_channels;
+
+ void *platform;
+
+ void *visualizer_lib;
+ int (*visualizer_start_output)(audio_io_handle_t);
+ int (*visualizer_stop_output)(audio_io_handle_t);
+};
+
+int select_devices(struct audio_device *adev,
+ audio_usecase_t uc_id);
+int disable_audio_route(struct audio_device *adev,
+ struct audio_usecase *usecase,
+ bool update_mixer);
+int disable_snd_device(struct audio_device *adev,
+ snd_device_t snd_device,
+ bool update_mixer);
+int enable_snd_device(struct audio_device *adev,
+ snd_device_t snd_device,
+ bool update_mixer);
+int enable_audio_route(struct audio_device *adev,
+ struct audio_usecase *usecase,
+ bool update_mixer);
+struct audio_usecase *get_usecase_from_list(struct audio_device *adev,
+ audio_usecase_t uc_id);
+/*
+ * 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