audio: Import N HAL from flounder
HEAD:
Turn off excess logging
Change-Id: I77d6eaf4ac31d969fd42e9a96418203bc682476f
Change-Id: I1cbcbf16ab617b676defcce49335d6f4190e63a9
diff --git a/audio/audio_hw.h b/audio/audio_hw.h
new file mode 100644
index 0000000..8b097e1
--- /dev/null
+++ b/audio/audio_hw.h
@@ -0,0 +1,456 @@
+/*
+ * 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 NVIDIA_AUDIO_HW_H
+#define NVIDIA_AUDIO_HW_H
+
+#include <cutils/list.h>
+#include <hardware/audio.h>
+
+#include <tinyalsa/asoundlib.h>
+#include <tinycompress/tinycompress.h>
+/* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
+#include <audio_utils/resampler.h>
+#include <audio_route/audio_route.h>
+
+/* Retry for delay in FW loading*/
+#define RETRY_NUMBER 10
+#define RETRY_US 500000
+
+#ifdef __LP64__
+#define OFFLOAD_FX_LIBRARY_PATH "/system/lib64/soundfx/libnvvisualizer.so"
+#else
+#define OFFLOAD_FX_LIBRARY_PATH "/system/lib/soundfx/libnvvisualizer.so"
+#endif
+
+#define HTC_ACOUSTIC_LIBRARY_PATH "/vendor/lib/libhtcacoustic.so"
+
+#ifdef PREPROCESSING_ENABLED
+#include <audio_utils/echo_reference.h>
+#define MAX_PREPROCESSORS 3
+struct effect_info_s {
+ effect_handle_t effect_itfe;
+ size_t num_channel_configs;
+ channel_config_t *channel_configs;
+};
+#endif
+
+#ifdef __LP64__
+#define SOUND_TRIGGER_HAL_LIBRARY_PATH "/system/lib64/hw/sound_trigger.primary.flounder.so"
+#else
+#define SOUND_TRIGGER_HAL_LIBRARY_PATH "/system/lib/hw/sound_trigger.primary.flounder.so"
+#endif
+
+#define TTY_MODE_OFF 1
+#define TTY_MODE_FULL 2
+#define TTY_MODE_VCO 4
+#define TTY_MODE_HCO 8
+
+#define DUALMIC_CONFIG_NONE 0
+#define DUALMIC_CONFIG_1 1
+
+/* Sound devices specific to the platform
+ * The DEVICE_OUT_* and DEVICE_IN_* should be mapped to these sound
+ * devices to enable corresponding mixer paths
+ */
+enum {
+ SND_DEVICE_NONE = 0,
+
+ /* Playback devices */
+ SND_DEVICE_MIN,
+ SND_DEVICE_OUT_BEGIN = SND_DEVICE_MIN,
+ SND_DEVICE_OUT_HANDSET = SND_DEVICE_OUT_BEGIN,
+ SND_DEVICE_OUT_SPEAKER,
+ SND_DEVICE_OUT_HEADPHONES,
+ SND_DEVICE_OUT_SPEAKER_AND_HEADPHONES,
+ SND_DEVICE_OUT_VOICE_HANDSET,
+ SND_DEVICE_OUT_VOICE_SPEAKER,
+ SND_DEVICE_OUT_VOICE_HEADPHONES,
+ SND_DEVICE_OUT_HDMI,
+ SND_DEVICE_OUT_SPEAKER_AND_HDMI,
+ SND_DEVICE_OUT_BT_SCO,
+ SND_DEVICE_OUT_VOICE_TTY_FULL_HEADPHONES,
+ SND_DEVICE_OUT_VOICE_TTY_VCO_HEADPHONES,
+ SND_DEVICE_OUT_VOICE_TTY_HCO_HANDSET,
+ SND_DEVICE_OUT_END,
+
+ /*
+ * Note: IN_BEGIN should be same as OUT_END because total number of devices
+ * SND_DEVICES_MAX should not exceed MAX_RX + MAX_TX devices.
+ */
+ /* Capture devices */
+ SND_DEVICE_IN_BEGIN = SND_DEVICE_OUT_END,
+ SND_DEVICE_IN_HANDSET_MIC = SND_DEVICE_IN_BEGIN,
+ SND_DEVICE_IN_SPEAKER_MIC,
+ SND_DEVICE_IN_HEADSET_MIC,
+ SND_DEVICE_IN_HANDSET_MIC_AEC,
+ SND_DEVICE_IN_SPEAKER_MIC_AEC,
+ SND_DEVICE_IN_HEADSET_MIC_AEC,
+ SND_DEVICE_IN_VOICE_SPEAKER_MIC,
+ SND_DEVICE_IN_VOICE_HEADSET_MIC,
+ SND_DEVICE_IN_HDMI_MIC,
+ SND_DEVICE_IN_BT_SCO_MIC,
+ SND_DEVICE_IN_CAMCORDER_MIC,
+ SND_DEVICE_IN_VOICE_DMIC_1,
+ SND_DEVICE_IN_VOICE_SPEAKER_DMIC_1,
+ SND_DEVICE_IN_VOICE_TTY_FULL_HEADSET_MIC,
+ SND_DEVICE_IN_VOICE_TTY_VCO_HANDSET_MIC,
+ SND_DEVICE_IN_VOICE_TTY_HCO_HEADSET_MIC,
+ SND_DEVICE_IN_VOICE_REC_HEADSET_MIC,
+ SND_DEVICE_IN_VOICE_REC_MIC,
+ SND_DEVICE_IN_VOICE_REC_DMIC_1,
+ SND_DEVICE_IN_VOICE_REC_DMIC_NS_1,
+ SND_DEVICE_IN_LOOPBACK_AEC,
+ SND_DEVICE_IN_END,
+
+ SND_DEVICE_MAX = SND_DEVICE_IN_END,
+
+};
+
+
+#define MIXER_CARD 0
+#define SOUND_CARD 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
+ */
+#define PLAYBACK_PERIOD_SIZE 256
+#define PLAYBACK_PERIOD_COUNT 2
+#define PLAYBACK_DEFAULT_CHANNEL_COUNT 2
+#define PLAYBACK_DEFAULT_SAMPLING_RATE 48000
+#define PLAYBACK_START_THRESHOLD(size, count) (((size) * (count)) - 1)
+#define PLAYBACK_STOP_THRESHOLD(size, count) ((size) * ((count) + 2))
+#define PLAYBACK_AVAILABLE_MIN 1
+
+
+#define SCO_PERIOD_SIZE 168
+#define SCO_PERIOD_COUNT 2
+#define SCO_DEFAULT_CHANNEL_COUNT 2
+#define SCO_DEFAULT_SAMPLING_RATE 8000
+#define SCO_START_THRESHOLD 335
+#define SCO_STOP_THRESHOLD 336
+#define SCO_AVAILABLE_MIN 1
+
+#define PLAYBACK_HDMI_MULTI_PERIOD_SIZE 1024
+#define PLAYBACK_HDMI_MULTI_PERIOD_COUNT 4
+#define PLAYBACK_HDMI_MULTI_DEFAULT_CHANNEL_COUNT 6
+#define PLAYBACK_HDMI_MULTI_PERIOD_BYTES \
+ (PLAYBACK_HDMI_MULTI_PERIOD_SIZE * PLAYBACK_HDMI_MULTI_DEFAULT_CHANNEL_COUNT * 2)
+#define PLAYBACK_HDMI_MULTI_START_THRESHOLD 4095
+#define PLAYBACK_HDMI_MULTI_STOP_THRESHOLD 4096
+#define PLAYBACK_HDMI_MULTI_AVAILABLE_MIN 1
+
+#define PLAYBACK_HDMI_DEFAULT_CHANNEL_COUNT 2
+
+#define CAPTURE_PERIOD_SIZE 1024
+#define CAPTURE_PERIOD_SIZE_LOW_LATENCY 256
+#define CAPTURE_PERIOD_COUNT 2
+#define CAPTURE_PERIOD_COUNT_LOW_LATENCY 2
+#define CAPTURE_DEFAULT_CHANNEL_COUNT 2
+#define CAPTURE_DEFAULT_SAMPLING_RATE 48000
+#define CAPTURE_START_THRESHOLD 1
+
+#define COMPRESS_CARD 0
+#define COMPRESS_DEVICE 5
+#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024)
+#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
+/* ToDo: Check and update a proper value in msec */
+#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96
+#define COMPRESS_PLAYBACK_VOLUME_MAX 0x10000 //NV suggested value
+
+#define DEEP_BUFFER_OUTPUT_SAMPLING_RATE 48000
+#define DEEP_BUFFER_OUTPUT_PERIOD_SIZE 480
+#define DEEP_BUFFER_OUTPUT_PERIOD_COUNT 8
+
+#define MAX_SUPPORTED_CHANNEL_MASKS 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 = 0,
+ USECASE_AUDIO_PLAYBACK_MULTI_CH,
+ USECASE_AUDIO_PLAYBACK_OFFLOAD,
+ USECASE_AUDIO_PLAYBACK_DEEP_BUFFER,
+
+ /* Capture usecases */
+ USECASE_AUDIO_CAPTURE,
+ USECASE_AUDIO_CAPTURE_HOTWORD,
+
+ USECASE_VOICE_CALL,
+ 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,
+ OFFLOAD_STATE_PAUSED_FLUSHED,
+};
+
+typedef enum {
+ PCM_PLAYBACK = 0x1,
+ PCM_CAPTURE = 0x2,
+ VOICE_CALL = 0x4,
+ PCM_HOTWORD_STREAMING = 0x8,
+ PCM_CAPTURE_LOW_LATENCY = 0x10,
+} usecase_type_t;
+
+struct offload_cmd {
+ struct listnode node;
+ int cmd;
+ int data[];
+};
+
+struct pcm_device_profile {
+ struct pcm_config config;
+ int card;
+ int id;
+ usecase_type_t type;
+ audio_devices_t devices;
+};
+
+struct pcm_device {
+ struct listnode stream_list_node;
+ struct pcm_device_profile* pcm_profile;
+ struct pcm* pcm;
+ int status;
+ /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
+ struct resampler_itfe* resampler;
+ int16_t* res_buffer;
+ size_t res_byte_count;
+ int sound_trigger_handle;
+};
+
+struct stream_out {
+ struct audio_stream_out stream;
+ pthread_mutex_t lock; /* see note below on mutex acquisition order */
+ pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
+ pthread_cond_t cond;
+ struct pcm_config config;
+ struct listnode pcm_dev_list;
+ struct compr_config compr_config;
+ struct compress* compr;
+ int standby;
+ 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;
+ /* total frames written, not cleared when entering standby */
+ uint64_t written;
+ audio_io_handle_t handle;
+
+ int non_blocking;
+ 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;
+
+#ifdef PREPROCESSING_ENABLED
+ struct echo_reference_itfe *echo_reference;
+ // echo_reference_generation indicates if the echo reference used by the output stream is
+ // in sync with the one known by the audio_device. When different from the generation stored
+ // in the audio_device the output stream must release the echo reference.
+ // always modified with audio device and stream mutex locked.
+ int32_t echo_reference_generation;
+#endif
+
+ bool is_fastmixer_affinity_set;
+};
+
+struct stream_in {
+ struct audio_stream_in stream;
+ pthread_mutex_t lock; /* see note below on mutex acquisition order */
+ pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by
+ capture thread */
+ struct pcm_config config;
+ struct listnode pcm_dev_list;
+ int standby;
+ audio_source_t source;
+ audio_devices_t devices;
+ uint32_t main_channels;
+ audio_usecase_t usecase;
+ usecase_type_t usecase_type;
+ bool enable_aec;
+ audio_input_flags_t input_flags;
+
+ /* TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8 */
+ unsigned int requested_rate;
+ struct resampler_itfe* resampler;
+ struct resampler_buffer_provider buf_provider;
+ int read_status;
+ int16_t* read_buf;
+ size_t read_buf_size;
+ size_t read_buf_frames;
+
+ int16_t *proc_buf_in;
+ int16_t *proc_buf_out;
+ size_t proc_buf_size;
+ size_t proc_buf_frames;
+
+#ifdef PREPROCESSING_ENABLED
+ struct echo_reference_itfe *echo_reference;
+ int16_t *ref_buf;
+ size_t ref_buf_size;
+ size_t ref_buf_frames;
+
+#ifdef HW_AEC_LOOPBACK
+ bool hw_echo_reference;
+ int16_t* hw_ref_buf;
+ size_t hw_ref_buf_size;
+#endif
+
+ int num_preprocessors;
+ struct effect_info_s preprocessors[MAX_PREPROCESSORS];
+
+ bool aux_channels_changed;
+ uint32_t aux_channels;
+#endif
+
+ struct audio_device* dev;
+ bool is_fastcapture_affinity_set;
+};
+
+struct mixer_card {
+ struct listnode adev_list_node;
+ struct listnode uc_list_node[AUDIO_USECASE_MAX];
+ int card;
+ struct mixer* mixer;
+ struct audio_route* audio_route;
+};
+
+struct audio_usecase {
+ struct listnode adev_list_node;
+ 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 audio_stream* stream;
+ struct listnode mixer_list;
+};
+
+
+struct audio_device {
+ struct audio_hw_device device;
+ pthread_mutex_t lock; /* see note below on mutex acquisition order */
+ struct listnode mixer_list;
+ audio_mode_t mode;
+ struct stream_in* active_input;
+ struct stream_out* primary_output;
+ int in_call;
+ float voice_volume;
+ bool mic_mute;
+ int tty_mode;
+ bool bluetooth_nrec;
+ bool screen_off;
+ int* snd_dev_ref_cnt;
+ struct listnode usecase_list;
+ bool speaker_lr_swap;
+ unsigned int cur_hdmi_channels;
+ int dualmic_config;
+ bool ns_in_voice_rec;
+
+ void* offload_fx_lib;
+ int (*offload_fx_start_output)(audio_io_handle_t);
+ int (*offload_fx_stop_output)(audio_io_handle_t);
+
+#ifdef PREPROCESSING_ENABLED
+ struct echo_reference_itfe* echo_reference;
+ // echo_reference_generation indicates if the echo reference used by the output stream is
+ // in sync with the one known by the audio_device.
+ // incremented atomically with a memory barrier and audio device mutex locked but WITHOUT
+ // stream mutex locked: the stream will load it atomically with a barrier and re-read it
+ // with audio device mutex if needed
+ volatile int32_t echo_reference_generation;
+#endif
+
+ void* htc_acoustic_lib;
+ int (*htc_acoustic_init_rt5506)();
+ int (*htc_acoustic_set_rt5506_amp)(int, int);
+ int (*htc_acoustic_set_amp_mode)(int, int, int, int, bool);
+ int (*htc_acoustic_spk_reverse)(bool);
+
+ void* sound_trigger_lib;
+ int (*sound_trigger_open_for_streaming)();
+ size_t (*sound_trigger_read_samples)(int, void*, size_t);
+ int (*sound_trigger_close_for_streaming)(int);
+
+ int tfa9895_init;
+ int tfa9895_mode_change;
+ pthread_mutex_t tfa9895_lock;
+
+ int dummybuf_thread_timeout;
+ int dummybuf_thread_cancel;
+ int dummybuf_thread_active;
+ audio_devices_t dummybuf_thread_devices;
+ pthread_mutex_t dummybuf_thread_lock;
+ pthread_t dummybuf_thread;
+
+ pthread_mutex_t lock_inputs; /* see note below on mutex acquisition order */
+};
+
+/*
+ * NOTE: when multiple mutexes have to be acquired, always take the
+ * lock_inputs, stream_in, stream_out, audio_device, then tfa9895 mutex.
+ * stream_in mutex must always be before stream_out mutex
+ * if both have to be taken (see get_echo_reference(), put_echo_reference()...)
+ * dummybuf_thread mutex is not related to the other mutexes with respect to order.
+ * lock_inputs must be held in order to either close the input stream, or prevent closure.
+ */
+
+#endif // NVIDIA_AUDIO_HW_H