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android / device / samsung / tuna / 1a42aa130cce5fd99e244d111a97d120b01e386b / . / audio / audio_hw.c
blob: 7a24bcfc8fd53dce281f862de3ecaeddd5035249 [file] [log] [blame]
/*
* Copyright (C) 2011 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.
*/
#define LOG_TAG "audio_hw_primary"
/*#define LOG_NDEBUG 0*/
#include <errno.h>
#include <pthread.h>
#include <stdint.h>
#include <sys/time.h>
#include <stdlib.h>
#include <cutils/log.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include <hardware/audio.h>
#include <tinyalsa/asoundlib.h>
#include <audio_utils/resampler.h>
#include <audio_utils/echo_reference.h>
#include <hardware/audio_effect.h>
#include <audio_effects/effect_aec.h>
#include "ril_interface.h"
/* Mixer control names */
#define MIXER_DL2_LEFT_EQUALIZER "DL2 Left Equalizer"
#define MIXER_DL2_RIGHT_EQUALIZER "DL2 Right Equalizer"
#define MIXER_DL1_MEDIA_PLAYBACK_VOLUME "DL1 Media Playback Volume"
#define MIXER_DL1_VOICE_PLAYBACK_VOLUME "DL1 Voice Playback Volume"
#define MIXER_DL1_TONES_PLAYBACK_VOLUME "DL1 Tones Playback Volume"
#define MIXER_DL2_MEDIA_PLAYBACK_VOLUME "DL2 Media Playback Volume"
#define MIXER_DL2_VOICE_PLAYBACK_VOLUME "DL2 Voice Playback Volume"
#define MIXER_DL2_TONES_PLAYBACK_VOLUME "DL2 Tones Playback Volume"
#define MIXER_SDT_DL_VOLUME "SDT DL Volume"
#define MIXER_SDT_UL_VOLUME "SDT UL Volume"
#define MIXER_HEADSET_PLAYBACK_VOLUME "Headset Playback Volume"
#define MIXER_HANDSFREE_PLAYBACK_VOLUME "Handsfree Playback Volume"
#define MIXER_EARPHONE_PLAYBACK_VOLUME "Earphone Playback Volume"
#define MIXER_BT_UL_VOLUME "BT UL Volume"
#define MIXER_DL1_EQUALIZER "DL1 Equalizer"
#define MIXER_DL1_MIXER_MULTIMEDIA "DL1 Mixer Multimedia"
#define MIXER_DL1_MIXER_VOICE "DL1 Mixer Voice"
#define MIXER_DL1_MIXER_TONES "DL1 Mixer Tones"
#define MIXER_DL2_MIXER_MULTIMEDIA "DL2 Mixer Multimedia"
#define MIXER_DL2_MIXER_VOICE "DL2 Mixer Voice"
#define MIXER_DL2_MIXER_TONES "DL2 Mixer Tones"
#define MIXER_SIDETONE_MIXER_PLAYBACK "Sidetone Mixer Playback"
#define MIXER_SIDETONE_MIXER_CAPTURE "Sidetone Mixer Capture"
#define MIXER_DL2_MONO_MIXER "DL2 Mono Mixer"
#define MIXER_DL1_PDM_SWITCH "DL1 PDM Switch"
#define MIXER_DL1_BT_VX_SWITCH "DL1 BT_VX Switch"
#define MIXER_VOICE_CAPTURE_MIXER_CAPTURE "Voice Capture Mixer Capture"
#define MIXER_HS_LEFT_PLAYBACK "HS Left Playback"
#define MIXER_HS_RIGHT_PLAYBACK "HS Right Playback"
#define MIXER_HF_LEFT_PLAYBACK "HF Left Playback"
#define MIXER_HF_RIGHT_PLAYBACK "HF Right Playback"
#define MIXER_EARPHONE_ENABLE_SWITCH "Earphone Enable Switch"
#define MIXER_ANALOG_LEFT_CAPTURE_ROUTE "Analog Left Capture Route"
#define MIXER_ANALOG_RIGHT_CAPTURE_ROUTE "Analog Right Capture Route"
#define MIXER_CAPTURE_PREAMPLIFIER_VOLUME "Capture Preamplifier Volume"
#define MIXER_CAPTURE_VOLUME "Capture Volume"
#define MIXER_AMIC_UL_VOLUME "AMIC UL Volume"
#define MIXER_AUDUL_VOICE_UL_VOLUME "AUDUL Voice UL Volume"
#define MIXER_MUX_VX0 "MUX_VX0"
#define MIXER_MUX_VX1 "MUX_VX1"
#define MIXER_MUX_UL10 "MUX_UL10"
#define MIXER_MUX_UL11 "MUX_UL11"
/* Mixer control gain and route values */
#define MIXER_ABE_GAIN_0DB 120
#define MIXER_PLAYBACK_HS_DAC "HS DAC"
#define MIXER_PLAYBACK_HF_DAC "HF DAC"
#define MIXER_MAIN_MIC "Main Mic"
#define MIXER_SUB_MIC "Sub Mic"
#define MIXER_HS_MIC "Headset Mic"
#define MIXER_AMIC0 "AMic0"
#define MIXER_AMIC1 "AMic1"
#define MIXER_BT_LEFT "BT Left"
#define MIXER_BT_RIGHT "BT Right"
#define MIXER_450HZ_HIGH_PASS "450Hz High-pass"
#define MIXER_FLAT_RESPONSE "Flat response"
#define MIXER_4KHZ_LPF_0DB "4Khz LPF 0dB"
/* ALSA cards for OMAP4 */
#define CARD_OMAP4_ABE 0
#define CARD_OMAP4_HDMI 1
#define CARD_TUNA_DEFAULT CARD_OMAP4_ABE
/* ALSA ports for OMAP4 */
#define PORT_MM 0
#define PORT_MM2_UL 1
#define PORT_VX 2
#define PORT_TONES 3
#define PORT_VIBRA 4
#define PORT_MODEM 5
#define PORT_MM_LP 6
#define PORT_SPDIF 9
#define PORT_HDMI 0
/* User serviceable */
/* #define to use mmap no-irq mode for playback, #undef for non-mmap irq mode */
#undef PLAYBACK_MMAP // was #define
/* short period (aka low latency) in milliseconds */
#define SHORT_PERIOD_MS 4 // was 22
/* long period (aka low power or deep buffer) in milliseconds */
#define LONG_PERIOD_MS 308
/* Constraint imposed by ABE: for playback, all period sizes must be multiples of 24 frames
* = 500 us at 48 kHz. It seems to be either 48 or 96 for capture, or maybe it is because the
* limitation is actually a min number of bytes which translates to a different amount of frames
* according to the number of channels.
*/
#define ABE_BASE_FRAME_COUNT 24
/* Derived from MM_FULL_POWER_SAMPLING_RATE=48000 and ABE_BASE_FRAME_COUNT=24 */
#define MULTIPLIER_FACTOR 2
/* number of base blocks in a short period (low latency) */
#define SHORT_PERIOD_MULTIPLIER (SHORT_PERIOD_MS * MULTIPLIER_FACTOR)
/* number of frames per short period (low latency) */
#define SHORT_PERIOD_SIZE (ABE_BASE_FRAME_COUNT * SHORT_PERIOD_MULTIPLIER)
/* number of short periods in a long period */
#define LONG_PERIOD_MULTIPLIER (LONG_PERIOD_MS / SHORT_PERIOD_MS)
/* number of frames per long period (low power) */
#define LONG_PERIOD_SIZE (SHORT_PERIOD_SIZE * LONG_PERIOD_MULTIPLIER)
/* number of periods for low power playback */
#define PLAYBACK_LONG_PERIOD_COUNT 2
/* Number of pseudo periods for low latency playback.
* These are called "pseudo" periods in that they are not known as periods by ALSA.
* Formerly, ALSA was configured in MMAP mode with 2 large periods, and this
* number was set to 4 (2 didn't work).
* The short periods size and count were only known by the audio HAL.
* Now for low latency, we are using non-MMAP mode and can set this to 2.
*/
#ifdef PLAYBACK_MMAP
#define PLAYBACK_SHORT_PERIOD_COUNT 4
#else
#define PLAYBACK_SHORT_PERIOD_COUNT 2
#endif
/* write function */
#ifdef PLAYBACK_MMAP
#define PCM_WRITE pcm_mmap_write
#else
#define PCM_WRITE pcm_write
#endif
/* User serviceable */
#define CAPTURE_PERIOD_MS 22
/* Number of frames per period for capture. This cannot be reduced below 96.
* Possibly related to the following rule in sound/soc/omap/omap-pcm.c:
* ret = snd_pcm_hw_constraint_step(runtime, 0, SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 384);
* (where 96 * 4 = 384)
* The only constraints I can find are periods_min = 2, period_bytes_min = 32.
* If you define RULES_DEBUG in sound/core/pcm_native.c, you can see which rule
* caused capture to fail.
* Decoupling playback and capture period size may have impacts on echo canceler behavior:
* to be verified. Currently 96 = 4 x 24 but it could be changed without noticing
* if we use separate defines.
*/
#define CAPTURE_PERIOD_SIZE (ABE_BASE_FRAME_COUNT * CAPTURE_PERIOD_MS * MULTIPLIER_FACTOR)
/* number of periods for capture */
#define CAPTURE_PERIOD_COUNT 2
/* minimum sleep time in out_write() when write threshold is not reached */
#define MIN_WRITE_SLEEP_US 5000
#define DEFAULT_OUT_SAMPLING_RATE 44100 // 48000 is possible but interacts poorly with HDMI
/* sampling rate when using MM low power port */
#define MM_LOW_POWER_SAMPLING_RATE 44100
/* sampling rate when using MM full power port */
#define MM_FULL_POWER_SAMPLING_RATE 48000 // affects MULTIPLIER_FACTOR
/* sampling rate when using VX port for narrow band */
#define VX_NB_SAMPLING_RATE 8000
/* sampling rate when using VX port for wide band */
#define VX_WB_SAMPLING_RATE 16000
/* conversions from dB to ABE and codec gains */
#define DB_TO_ABE_GAIN(x) ((x) + MIXER_ABE_GAIN_0DB)
#define DB_TO_CAPTURE_PREAMPLIFIER_VOLUME(x) (((x) + 6) / 6)
#define DB_TO_CAPTURE_VOLUME(x) (((x) - 6) / 6)
#define DB_TO_HEADSET_VOLUME(x) (((x) + 30) / 2)
#define DB_TO_SPEAKER_VOLUME(x) (((x) + 52) / 2)
#define DB_TO_EARPIECE_VOLUME(x) (((x) + 24) / 2)
/* conversions from codec and ABE gains to dB */
#define DB_FROM_SPEAKER_VOLUME(x) ((x) * 2 - 52)
/* use-case specific mic volumes, all in dB */
#define CAPTURE_MAIN_MIC_VOLUME 16
#define CAPTURE_SUB_MIC_VOLUME 18
#define CAPTURE_HEADSET_MIC_VOLUME 12
#define VOICE_RECOGNITION_MAIN_MIC_VOLUME 5
#define VOICE_RECOGNITION_SUB_MIC_VOLUME 18
#define VOICE_RECOGNITION_HEADSET_MIC_VOLUME 14
#define CAMCORDER_MAIN_MIC_VOLUME 13
#define CAMCORDER_SUB_MIC_VOLUME 10
#define CAMCORDER_HEADSET_MIC_VOLUME 12
#define VOIP_MAIN_MIC_VOLUME 13
#define VOIP_SUB_MIC_VOLUME 20
#define VOIP_HEADSET_MIC_VOLUME 12
#define VOICE_CALL_MAIN_MIC_VOLUME 0
#define VOICE_CALL_SUB_MIC_VOLUME_MAGURO -4
#define VOICE_CALL_SUB_MIC_VOLUME_TORO -2
#define VOICE_CALL_HEADSET_MIC_VOLUME 8
/* use-case specific output volumes */
#define NORMAL_SPEAKER_VOLUME_TORO 4
#define NORMAL_SPEAKER_VOLUME_MAGURO 2
#define NORMAL_HEADSET_VOLUME_TORO -12
#define NORMAL_HEADSET_VOLUME_MAGURO -12
#define NORMAL_HEADPHONE_VOLUME_TORO -6 /* allow louder output for headphones */
#define NORMAL_HEADPHONE_VOLUME_MAGURO -6
#define NORMAL_EARPIECE_VOLUME_TORO -2
#define NORMAL_EARPIECE_VOLUME_MAGURO -2
#define VOICE_CALL_SPEAKER_VOLUME_TORO 9
#define VOICE_CALL_SPEAKER_VOLUME_MAGURO 6
#define VOICE_CALL_HEADSET_VOLUME_TORO -6
#define VOICE_CALL_HEADSET_VOLUME_MAGURO 0
#define VOICE_CALL_EARPIECE_VOLUME_TORO 2
#define VOICE_CALL_EARPIECE_VOLUME_MAGURO 6
#define VOIP_SPEAKER_VOLUME_TORO 9
#define VOIP_SPEAKER_VOLUME_MAGURO 7
#define VOIP_HEADSET_VOLUME_TORO -6
#define VOIP_HEADSET_VOLUME_MAGURO -6
#define VOIP_EARPIECE_VOLUME_TORO 6
#define VOIP_EARPIECE_VOLUME_MAGURO 6
#define HEADPHONE_VOLUME_TTY -2
#define RINGTONE_HEADSET_VOLUME_OFFSET -14
/* product-specific defines */
#define PRODUCT_DEVICE_PROPERTY "ro.product.device"
#define PRODUCT_NAME_PROPERTY "ro.product.name"
#define PRODUCT_DEVICE_TORO "toro"
#define PRODUCT_NAME_YAKJU "yakju"
enum tty_modes {
TTY_MODE_OFF,
TTY_MODE_VCO,
TTY_MODE_HCO,
TTY_MODE_FULL
};
struct pcm_config pcm_config_mm = {
.channels = 2,
.rate = MM_FULL_POWER_SAMPLING_RATE,
.period_size = LONG_PERIOD_SIZE,
.period_count = PLAYBACK_LONG_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
.start_threshold = LONG_PERIOD_SIZE,
.avail_min = LONG_PERIOD_SIZE,
};
struct pcm_config pcm_config_tones = {
.channels = 2,
.rate = MM_FULL_POWER_SAMPLING_RATE,
.period_size = SHORT_PERIOD_SIZE,
.period_count = PLAYBACK_SHORT_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
#ifdef PLAYBACK_MMAP
.start_threshold = SHORT_PERIOD_SIZE,
.avail_min = SHORT_PERIOD_SIZE,
#else
.start_threshold = 0,
.avail_min = 0,
#endif
};
struct pcm_config pcm_config_mm_ul = {
.channels = 2,
.rate = MM_FULL_POWER_SAMPLING_RATE,
.period_size = CAPTURE_PERIOD_SIZE,
.period_count = CAPTURE_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
};
struct pcm_config pcm_config_vx = {
.channels = 2,
.rate = VX_NB_SAMPLING_RATE,
.period_size = 160,
.period_count = 2,
.format = PCM_FORMAT_S16_LE,
};
#define MIN(x, y) ((x) > (y) ? (y) : (x))
struct route_setting
{
char *ctl_name;
int intval;
char *strval;
};
/* These are values that never change */
struct route_setting defaults[] = {
/* general */
{
.ctl_name = MIXER_DL2_LEFT_EQUALIZER,
.strval = MIXER_450HZ_HIGH_PASS,
},
{
.ctl_name = MIXER_DL2_RIGHT_EQUALIZER,
.strval = MIXER_450HZ_HIGH_PASS,
},
{
.ctl_name = MIXER_DL1_MEDIA_PLAYBACK_VOLUME,
.intval = MIXER_ABE_GAIN_0DB,
},
{
.ctl_name = MIXER_DL2_MEDIA_PLAYBACK_VOLUME,
.intval = MIXER_ABE_GAIN_0DB,
},
{
.ctl_name = MIXER_DL1_VOICE_PLAYBACK_VOLUME,
.intval = MIXER_ABE_GAIN_0DB,
},
{
.ctl_name = MIXER_DL2_VOICE_PLAYBACK_VOLUME,
.intval = MIXER_ABE_GAIN_0DB,
},
{
.ctl_name = MIXER_DL1_TONES_PLAYBACK_VOLUME,
.intval = MIXER_ABE_GAIN_0DB,
},
{
.ctl_name = MIXER_DL2_TONES_PLAYBACK_VOLUME,
.intval = MIXER_ABE_GAIN_0DB,
},
{
.ctl_name = MIXER_SDT_DL_VOLUME,
.intval = MIXER_ABE_GAIN_0DB,
},
{
.ctl_name = MIXER_AUDUL_VOICE_UL_VOLUME,
.intval = MIXER_ABE_GAIN_0DB,
},
{
.ctl_name = MIXER_CAPTURE_PREAMPLIFIER_VOLUME,
.intval = DB_TO_CAPTURE_PREAMPLIFIER_VOLUME(0),
},
{
.ctl_name = MIXER_CAPTURE_VOLUME,
.intval = DB_TO_CAPTURE_VOLUME(30),
},
{
.ctl_name = MIXER_SDT_UL_VOLUME,
.intval = MIXER_ABE_GAIN_0DB - 17,
},
{
.ctl_name = MIXER_SIDETONE_MIXER_CAPTURE,
.intval = 0,
},
/* headset */
{
.ctl_name = MIXER_SIDETONE_MIXER_PLAYBACK,
.intval = 1,
},
{
.ctl_name = MIXER_DL1_PDM_SWITCH,
.intval = 1,
},
/* bt */
{
.ctl_name = MIXER_BT_UL_VOLUME,
.intval = MIXER_ABE_GAIN_0DB,
},
{
.ctl_name = NULL,
},
};
struct route_setting hf_output[] = {
{
.ctl_name = MIXER_HF_LEFT_PLAYBACK,
.strval = MIXER_PLAYBACK_HF_DAC,
},
{
.ctl_name = MIXER_HF_RIGHT_PLAYBACK,
.strval = MIXER_PLAYBACK_HF_DAC,
},
{
.ctl_name = NULL,
},
};
struct route_setting hs_output[] = {
{
.ctl_name = MIXER_HS_LEFT_PLAYBACK,
.strval = MIXER_PLAYBACK_HS_DAC,
},
{
.ctl_name = MIXER_HS_RIGHT_PLAYBACK,
.strval = MIXER_PLAYBACK_HS_DAC,
},
{
.ctl_name = NULL,
},
};
/* MM UL front-end paths */
struct route_setting mm_ul2_bt[] = {
{
.ctl_name = MIXER_MUX_UL10,
.strval = MIXER_BT_LEFT,
},
{
.ctl_name = MIXER_MUX_UL11,
.strval = MIXER_BT_LEFT,
},
{
.ctl_name = NULL,
},
};
struct route_setting mm_ul2_amic_left[] = {
{
.ctl_name = MIXER_MUX_UL10,
.strval = MIXER_AMIC0,
},
{
.ctl_name = MIXER_MUX_UL11,
.strval = MIXER_AMIC0,
},
{
.ctl_name = NULL,
},
};
struct route_setting mm_ul2_amic_right[] = {
{
.ctl_name = MIXER_MUX_UL10,
.strval = MIXER_AMIC1,
},
{
.ctl_name = MIXER_MUX_UL11,
.strval = MIXER_AMIC1,
},
{
.ctl_name = NULL,
},
};
/* dual mic configuration with main mic on main channel and sub mic on aux channel.
* Used for handset mode (near talk) */
struct route_setting mm_ul2_amic_dual_main_sub[] = {
{
.ctl_name = MIXER_MUX_UL10,
.strval = MIXER_AMIC0,
},
{
.ctl_name = MIXER_MUX_UL11,
.strval = MIXER_AMIC1,
},
{
.ctl_name = NULL,
},
};
/* dual mic configuration with sub mic on main channel and main mic on aux channel.
* Used for speakerphone mode (far talk) */
struct route_setting mm_ul2_amic_dual_sub_main[] = {
{
.ctl_name = MIXER_MUX_UL10,
.strval = MIXER_AMIC1,
},
{
.ctl_name = MIXER_MUX_UL11,
.strval = MIXER_AMIC0,
},
{
.ctl_name = NULL,
},
};
/* VX UL front-end paths */
struct route_setting vx_ul_amic_left[] = {
{
.ctl_name = MIXER_MUX_VX0,
.strval = MIXER_AMIC0,
},
{
.ctl_name = MIXER_MUX_VX1,
.strval = MIXER_AMIC0,
},
{
.ctl_name = MIXER_VOICE_CAPTURE_MIXER_CAPTURE,
.intval = 1,
},
{
.ctl_name = NULL,
},
};
struct route_setting vx_ul_amic_right[] = {
{
.ctl_name = MIXER_MUX_VX0,
.strval = MIXER_AMIC1,
},
{
.ctl_name = MIXER_MUX_VX1,
.strval = MIXER_AMIC1,
},
{
.ctl_name = MIXER_VOICE_CAPTURE_MIXER_CAPTURE,
.intval = 1,
},
{
.ctl_name = NULL,
},
};
struct route_setting vx_ul_bt[] = {
{
.ctl_name = MIXER_MUX_VX0,
.strval = MIXER_BT_LEFT,
},
{
.ctl_name = MIXER_MUX_VX1,
.strval = MIXER_BT_LEFT,
},
{
.ctl_name = MIXER_VOICE_CAPTURE_MIXER_CAPTURE,
.intval = 1,
},
{
.ctl_name = NULL,
},
};
struct mixer_ctls
{
struct mixer_ctl *dl1_eq;
struct mixer_ctl *mm_dl2_volume;
struct mixer_ctl *vx_dl2_volume;
struct mixer_ctl *mm_dl1;
struct mixer_ctl *mm_dl2;
struct mixer_ctl *vx_dl1;
struct mixer_ctl *vx_dl2;
struct mixer_ctl *tones_dl1;
struct mixer_ctl *tones_dl2;
struct mixer_ctl *earpiece_enable;
struct mixer_ctl *dl2_mono;
struct mixer_ctl *dl1_headset;
struct mixer_ctl *dl1_bt;
struct mixer_ctl *left_capture;
struct mixer_ctl *right_capture;
struct mixer_ctl *amic_ul_volume;
struct mixer_ctl *voice_ul_volume;
struct mixer_ctl *sidetone_capture;
struct mixer_ctl *headset_volume;
struct mixer_ctl *speaker_volume;
struct mixer_ctl *earpiece_volume;
};
enum output_type {
OUTPUT_DEEP_BUF, // deep PCM buffers output stream
OUTPUT_LOW_LATENCY, // low latency output stream
OUTPUT_TOTAL
};
struct tuna_audio_device {
struct audio_hw_device hw_device;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
struct mixer *mixer;
struct mixer_ctls mixer_ctls;
audio_mode_t mode;
int devices;
struct pcm *pcm_modem_dl;
struct pcm *pcm_modem_ul;
int in_call;
float voice_volume;
struct tuna_stream_in *active_input;
struct tuna_stream_out *outputs[OUTPUT_TOTAL];
bool mic_mute;
int tty_mode;
struct echo_reference_itfe *echo_reference;
bool bluetooth_nrec;
bool device_is_toro;
int wb_amr;
/* RIL */
struct ril_handle ril;
};
enum pcm_type {
PCM_NORMAL = 0,
PCM_SPDIF,
PCM_HDMI,
PCM_TOTAL,
};
struct tuna_stream_out {
struct audio_stream_out stream;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
struct pcm_config config[PCM_TOTAL];
struct pcm *pcm[PCM_TOTAL];
struct resampler_itfe *resampler;
char *buffer;
size_t buffer_frames;
int standby;
struct echo_reference_itfe *echo_reference;
struct tuna_audio_device *dev;
};
#define MAX_PREPROCESSORS 3 /* maximum one AGC + one NS + one AEC per input stream */
struct effect_info_s {
effect_handle_t effect_itfe;
size_t num_channel_configs;
channel_config_t* channel_configs;
};
#define NUM_IN_AUX_CNL_CONFIGS 2
channel_config_t in_aux_cnl_configs[NUM_IN_AUX_CNL_CONFIGS] = {
{ AUDIO_CHANNEL_IN_FRONT , AUDIO_CHANNEL_IN_BACK},
{ AUDIO_CHANNEL_IN_STEREO , AUDIO_CHANNEL_IN_RIGHT}
};
struct tuna_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 device;
struct resampler_itfe *resampler;
struct resampler_buffer_provider buf_provider;
unsigned int requested_rate;
int standby;
int source;
struct echo_reference_itfe *echo_reference;
bool need_echo_reference;
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;
int16_t *ref_buf;
size_t ref_buf_size;
size_t ref_buf_frames;
int read_status;
int num_preprocessors;
struct effect_info_s preprocessors[MAX_PREPROCESSORS];
bool aux_channels_changed;
uint32_t main_channels;
uint32_t aux_channels;
struct tuna_audio_device *dev;
};
/**
* NOTE: when multiple mutexes have to be acquired, always respect the following order:
* hw device > in stream > out stream
*/
static void select_output_device(struct tuna_audio_device *adev);
static void select_input_device(struct tuna_audio_device *adev);
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume);
static int do_input_standby(struct tuna_stream_in *in);
static int do_output_standby(struct tuna_stream_out *out);
static void in_update_aux_channels(struct tuna_stream_in *in, effect_handle_t effect);
/* Returns true on devices that are toro, false otherwise */
static int is_device_toro(void)
{
char property[PROPERTY_VALUE_MAX];
property_get(PRODUCT_DEVICE_PROPERTY, property, PRODUCT_DEVICE_TORO);
/* return true if the property matches the given value */
return strcmp(property, PRODUCT_DEVICE_TORO) == 0;
}
/* The enable flag when 0 makes the assumption that enums are disabled by
* "Off" and integers/booleans by 0 */
static int set_route_by_array(struct mixer *mixer, struct route_setting *route,
int enable)
{
struct mixer_ctl *ctl;
unsigned int i, j;
/* Go through the route array and set each value */
i = 0;
while (route[i].ctl_name) {
ctl = mixer_get_ctl_by_name(mixer, route[i].ctl_name);
if (!ctl)
return -EINVAL;
if (route[i].strval) {
if (enable)
mixer_ctl_set_enum_by_string(ctl, route[i].strval);
else
mixer_ctl_set_enum_by_string(ctl, "Off");
} else {
/* This ensures multiple (i.e. stereo) values are set jointly */
for (j = 0; j < mixer_ctl_get_num_values(ctl); j++) {
if (enable)
mixer_ctl_set_value(ctl, j, route[i].intval);
else
mixer_ctl_set_value(ctl, j, 0);
}
}
i++;
}
return 0;
}
static int start_call(struct tuna_audio_device *adev)
{
ALOGE("Opening modem PCMs");
pcm_config_vx.rate = adev->wb_amr ? VX_WB_SAMPLING_RATE : VX_NB_SAMPLING_RATE;
/* Open modem PCM channels */
if (adev->pcm_modem_dl == NULL) {
adev->pcm_modem_dl = pcm_open(0, PORT_MODEM, PCM_OUT, &pcm_config_vx);
if (!pcm_is_ready(adev->pcm_modem_dl)) {
ALOGE("cannot open PCM modem DL stream: %s", pcm_get_error(adev->pcm_modem_dl));
goto err_open_dl;
}
}
if (adev->pcm_modem_ul == NULL) {
adev->pcm_modem_ul = pcm_open(0, PORT_MODEM, PCM_IN, &pcm_config_vx);
if (!pcm_is_ready(adev->pcm_modem_ul)) {
ALOGE("cannot open PCM modem UL stream: %s", pcm_get_error(adev->pcm_modem_ul));
goto err_open_ul;
}
}
pcm_start(adev->pcm_modem_dl);
pcm_start(adev->pcm_modem_ul);
return 0;
err_open_ul:
pcm_close(adev->pcm_modem_ul);
adev->pcm_modem_ul = NULL;
err_open_dl:
pcm_close(adev->pcm_modem_dl);
adev->pcm_modem_dl = NULL;
return -ENOMEM;
}
static void end_call(struct tuna_audio_device *adev)
{
ALOGE("Closing modem PCMs");
pcm_stop(adev->pcm_modem_dl);
pcm_stop(adev->pcm_modem_ul);
pcm_close(adev->pcm_modem_dl);
pcm_close(adev->pcm_modem_ul);
adev->pcm_modem_dl = NULL;
adev->pcm_modem_ul = NULL;
}
static void set_eq_filter(struct tuna_audio_device *adev)
{
/* DL1_EQ can't be used for bt */
int dl1_eq_applicable = adev->devices & (AUDIO_DEVICE_OUT_WIRED_HEADSET |
AUDIO_DEVICE_OUT_WIRED_HEADPHONE | AUDIO_DEVICE_OUT_EARPIECE);
/* 4Khz LPF is used only in NB-AMR voicecall */
if ((adev->mode == AUDIO_MODE_IN_CALL) && dl1_eq_applicable &&
(adev->tty_mode == TTY_MODE_OFF) && !adev->wb_amr)
mixer_ctl_set_enum_by_string(adev->mixer_ctls.dl1_eq, MIXER_4KHZ_LPF_0DB);
else
mixer_ctl_set_enum_by_string(adev->mixer_ctls.dl1_eq, MIXER_FLAT_RESPONSE);
}
void audio_set_wb_amr_callback(void *data, int enable)
{
struct tuna_audio_device *adev = (struct tuna_audio_device *)data;
pthread_mutex_lock(&adev->lock);
if (adev->wb_amr != enable) {
adev->wb_amr = enable;
/* reopen the modem PCMs at the new rate */
if (adev->in_call) {
end_call(adev);
set_eq_filter(adev);
start_call(adev);
}
}
pthread_mutex_unlock(&adev->lock);
}
static void set_incall_device(struct tuna_audio_device *adev)
{
int device_type;
switch(adev->devices & AUDIO_DEVICE_OUT_ALL) {
case AUDIO_DEVICE_OUT_EARPIECE:
device_type = SOUND_AUDIO_PATH_HANDSET;
break;
case AUDIO_DEVICE_OUT_SPEAKER:
case AUDIO_DEVICE_OUT_AUX_DIGITAL:
case AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET:
device_type = SOUND_AUDIO_PATH_SPEAKER;
break;
case AUDIO_DEVICE_OUT_WIRED_HEADSET:
device_type = SOUND_AUDIO_PATH_HEADSET;
break;
case AUDIO_DEVICE_OUT_WIRED_HEADPHONE:
device_type = SOUND_AUDIO_PATH_HEADPHONE;
break;
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO:
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET:
case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT:
if (adev->bluetooth_nrec)
device_type = SOUND_AUDIO_PATH_BLUETOOTH;
else
device_type = SOUND_AUDIO_PATH_BLUETOOTH_NO_NR;
break;
default:
device_type = SOUND_AUDIO_PATH_HANDSET;
break;
}
/* if output device isn't supported, open modem side to handset by default */
ril_set_call_audio_path(&adev->ril, device_type);
}
static void set_input_volumes(struct tuna_audio_device *adev, int main_mic_on,
int headset_mic_on, int sub_mic_on)
{
unsigned int channel;
int volume = MIXER_ABE_GAIN_0DB;
if (adev->mode == AUDIO_MODE_IN_CALL) {
int sub_mic_volume = is_device_toro() ? VOICE_CALL_SUB_MIC_VOLUME_TORO :
VOICE_CALL_SUB_MIC_VOLUME_MAGURO;
/* special case: don't look at input source for IN_CALL state */
volume = DB_TO_ABE_GAIN(main_mic_on ? VOICE_CALL_MAIN_MIC_VOLUME :
(headset_mic_on ? VOICE_CALL_HEADSET_MIC_VOLUME :
(sub_mic_on ? sub_mic_volume : 0)));
} else if (adev->active_input) {
/* determine input volume by use case */
switch (adev->active_input->source) {
case AUDIO_SOURCE_MIC: /* general capture */
volume = DB_TO_ABE_GAIN(main_mic_on ? CAPTURE_MAIN_MIC_VOLUME :
(headset_mic_on ? CAPTURE_HEADSET_MIC_VOLUME :
(sub_mic_on ? CAPTURE_SUB_MIC_VOLUME : 0)));
break;
case AUDIO_SOURCE_CAMCORDER:
volume = DB_TO_ABE_GAIN(main_mic_on ? CAMCORDER_MAIN_MIC_VOLUME :
(headset_mic_on ? CAMCORDER_HEADSET_MIC_VOLUME :
(sub_mic_on ? CAMCORDER_SUB_MIC_VOLUME : 0)));
break;
case AUDIO_SOURCE_VOICE_RECOGNITION:
volume = DB_TO_ABE_GAIN(main_mic_on ? VOICE_RECOGNITION_MAIN_MIC_VOLUME :
(headset_mic_on ? VOICE_RECOGNITION_HEADSET_MIC_VOLUME :
(sub_mic_on ? VOICE_RECOGNITION_SUB_MIC_VOLUME : 0)));
break;
case AUDIO_SOURCE_VOICE_COMMUNICATION: /* VoIP */
volume = DB_TO_ABE_GAIN(main_mic_on ? VOIP_MAIN_MIC_VOLUME :
(headset_mic_on ? VOIP_HEADSET_MIC_VOLUME :
(sub_mic_on ? VOIP_SUB_MIC_VOLUME : 0)));
break;
default:
/* nothing to do */
break;
}
}
for (channel = 0; channel < 2; channel++)
mixer_ctl_set_value(adev->mixer_ctls.amic_ul_volume, channel, volume);
}
static void set_output_volumes(struct tuna_audio_device *adev, bool tty_volume)
{
unsigned int channel;
int speaker_volume;
int headset_volume;
int earpiece_volume;
bool toro = adev->device_is_toro;
int headphone_on = adev->devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
int speaker_on = adev->devices & AUDIO_DEVICE_OUT_SPEAKER;
int speaker_volume_overrange = MIXER_ABE_GAIN_0DB;
int speaker_max_db =
DB_FROM_SPEAKER_VOLUME(mixer_ctl_get_range_max(adev->mixer_ctls.speaker_volume));
struct mixer_ctl *mixer_ctl_overrange = adev->mixer_ctls.mm_dl2_volume;
if (adev->mode == AUDIO_MODE_IN_CALL) {
/* Voice call */
speaker_volume = toro ? VOICE_CALL_SPEAKER_VOLUME_TORO :
VOICE_CALL_SPEAKER_VOLUME_MAGURO;
headset_volume = toro ? VOICE_CALL_HEADSET_VOLUME_TORO :
VOICE_CALL_HEADSET_VOLUME_MAGURO;
earpiece_volume = toro ? VOICE_CALL_EARPIECE_VOLUME_TORO :
VOICE_CALL_EARPIECE_VOLUME_MAGURO;
mixer_ctl_overrange = adev->mixer_ctls.vx_dl2_volume;
} else if (adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
/* VoIP */
speaker_volume = toro ? VOIP_SPEAKER_VOLUME_TORO :
VOIP_SPEAKER_VOLUME_MAGURO;
headset_volume = toro ? VOIP_HEADSET_VOLUME_TORO :
VOIP_HEADSET_VOLUME_MAGURO;
earpiece_volume = toro ? VOIP_EARPIECE_VOLUME_TORO :
VOIP_EARPIECE_VOLUME_MAGURO;
} else {
/* Media */
speaker_volume = toro ? NORMAL_SPEAKER_VOLUME_TORO :
NORMAL_SPEAKER_VOLUME_MAGURO;
if (headphone_on)
headset_volume = toro ? NORMAL_HEADPHONE_VOLUME_TORO :
NORMAL_HEADPHONE_VOLUME_MAGURO;
else
headset_volume = toro ? NORMAL_HEADSET_VOLUME_TORO :
NORMAL_HEADSET_VOLUME_MAGURO;
earpiece_volume = toro ? NORMAL_EARPIECE_VOLUME_TORO :
NORMAL_EARPIECE_VOLUME_MAGURO;
}
if (tty_volume)
headset_volume = HEADPHONE_VOLUME_TTY;
else if (adev->mode == AUDIO_MODE_RINGTONE)
headset_volume += RINGTONE_HEADSET_VOLUME_OFFSET;
/* If we have run out of range in the codec (analog) speaker volume,
we have to apply the remainder of the dB increase to the DL2
media/voice mixer volume, which is a digital gain */
if (speaker_volume > speaker_max_db) {
speaker_volume_overrange += (speaker_volume - speaker_max_db);
speaker_volume = speaker_max_db;
}
for (channel = 0; channel < 2; channel++) {
mixer_ctl_set_value(adev->mixer_ctls.speaker_volume, channel,
DB_TO_SPEAKER_VOLUME(speaker_volume));
mixer_ctl_set_value(adev->mixer_ctls.headset_volume, channel,
DB_TO_HEADSET_VOLUME(headset_volume));
}
if (speaker_on)
mixer_ctl_set_value(mixer_ctl_overrange, 0, speaker_volume_overrange);
else
mixer_ctl_set_value(mixer_ctl_overrange, 0, MIXER_ABE_GAIN_0DB);
mixer_ctl_set_value(adev->mixer_ctls.earpiece_volume, 0,
DB_TO_EARPIECE_VOLUME(earpiece_volume));
}
static void force_all_standby(struct tuna_audio_device *adev)
{
struct tuna_stream_in *in;
struct tuna_stream_out *out;
/* only needed for low latency output streams as other streams are not used
* for voice use cases */
if (adev->outputs[OUTPUT_LOW_LATENCY] != NULL &&
!adev->outputs[OUTPUT_LOW_LATENCY]->standby) {
out = adev->outputs[OUTPUT_LOW_LATENCY];
pthread_mutex_lock(&out->lock);
do_output_standby(out);
pthread_mutex_unlock(&out->lock);
}
if (adev->active_input) {
in = adev->active_input;
pthread_mutex_lock(&in->lock);
do_input_standby(in);
pthread_mutex_unlock(&in->lock);
}
}
static void select_mode(struct tuna_audio_device *adev)
{
if (adev->mode == AUDIO_MODE_IN_CALL) {
ALOGE("Entering IN_CALL state, in_call=%d", adev->in_call);
if (!adev->in_call) {
force_all_standby(adev);
/* force earpiece route for in call state if speaker is the
only currently selected route. This prevents having to tear
down the modem PCMs to change route from speaker to earpiece
after the ringtone is played, but doesn't cause a route
change if a headset or bt device is already connected. If
speaker is not the only thing active, just remove it from
the route. We'll assume it'll never be used initally during
a call. This works because we're sure that the audio policy
manager will update the output device after the audio mode
change, even if the device selection did not change. */
if ((adev->devices & AUDIO_DEVICE_OUT_ALL) == AUDIO_DEVICE_OUT_SPEAKER)
adev->devices = AUDIO_DEVICE_OUT_EARPIECE |
AUDIO_DEVICE_IN_BUILTIN_MIC;
else
adev->devices &= ~AUDIO_DEVICE_OUT_SPEAKER;
select_output_device(adev);
start_call(adev);
adev_set_voice_volume(&adev->hw_device, adev->voice_volume);
adev->in_call = 1;
}
} else {
ALOGE("Leaving IN_CALL state, in_call=%d, mode=%d",
adev->in_call, adev->mode);
if (adev->in_call) {
adev->in_call = 0;
end_call(adev);
force_all_standby(adev);
select_output_device(adev);
select_input_device(adev);
}
}
}
static void select_output_device(struct tuna_audio_device *adev)
{
int headset_on;
int headphone_on;
int speaker_on;
int earpiece_on;
int bt_on;
int dl1_on;
int sidetone_capture_on = 0;
bool tty_volume = false;
unsigned int channel;
/* Mute VX_UL to avoid pop noises in the tx path
* during call before switch changes.
*/
if (adev->mode == AUDIO_MODE_IN_CALL) {
for (channel = 0; channel < 2; channel++)
mixer_ctl_set_value(adev->mixer_ctls.voice_ul_volume,
channel, 0);
}
headset_on = adev->devices & AUDIO_DEVICE_OUT_WIRED_HEADSET;
headphone_on = adev->devices & AUDIO_DEVICE_OUT_WIRED_HEADPHONE;
speaker_on = adev->devices & AUDIO_DEVICE_OUT_SPEAKER;
earpiece_on = adev->devices & AUDIO_DEVICE_OUT_EARPIECE;
bt_on = adev->devices & AUDIO_DEVICE_OUT_ALL_SCO;
/* force rx path according to TTY mode when in call */
if (adev->mode == AUDIO_MODE_IN_CALL && !bt_on) {
switch(adev->tty_mode) {
case TTY_MODE_FULL:
case TTY_MODE_VCO:
/* rx path to headphones */
headphone_on = 1;
headset_on = 0;
speaker_on = 0;
earpiece_on = 0;
tty_volume = true;
break;
case TTY_MODE_HCO:
/* rx path to device speaker */
headphone_on = 0;
headset_on = 0;
speaker_on = 1;
earpiece_on = 0;
break;
case TTY_MODE_OFF:
default:
/* force speaker on when in call and HDMI or S/PDIF is selected
* as voice DL audio cannot be routed there by ABE */
if (adev->devices &
(AUDIO_DEVICE_OUT_AUX_DIGITAL |
AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET))
speaker_on = 1;
break;
}
}
dl1_on = headset_on | headphone_on | earpiece_on | bt_on;
/* Select front end */
mixer_ctl_set_value(adev->mixer_ctls.mm_dl2, 0, speaker_on);
mixer_ctl_set_value(adev->mixer_ctls.tones_dl2, 0, speaker_on);
mixer_ctl_set_value(adev->mixer_ctls.vx_dl2, 0,
speaker_on && (adev->mode == AUDIO_MODE_IN_CALL));
mixer_ctl_set_value(adev->mixer_ctls.mm_dl1, 0, dl1_on);
mixer_ctl_set_value(adev->mixer_ctls.tones_dl1, 0, dl1_on);
mixer_ctl_set_value(adev->mixer_ctls.vx_dl1, 0,
dl1_on && (adev->mode == AUDIO_MODE_IN_CALL));
/* Select back end */
mixer_ctl_set_value(adev->mixer_ctls.dl1_headset, 0,
headset_on | headphone_on | earpiece_on);
mixer_ctl_set_value(adev->mixer_ctls.dl1_bt, 0, bt_on);
mixer_ctl_set_value(adev->mixer_ctls.dl2_mono, 0,
(adev->mode != AUDIO_MODE_IN_CALL) && speaker_on);
mixer_ctl_set_value(adev->mixer_ctls.earpiece_enable, 0, earpiece_on);
/* select output stage */
set_route_by_array(adev->mixer, hs_output, headset_on | headphone_on);
set_route_by_array(adev->mixer, hf_output, speaker_on);
set_eq_filter(adev);
set_output_volumes(adev, tty_volume);
/* Special case: select input path if in a call, otherwise
in_set_parameters is used to update the input route
todo: use sub mic for handsfree case */
if (adev->mode == AUDIO_MODE_IN_CALL) {
if (bt_on)
set_route_by_array(adev->mixer, vx_ul_bt, bt_on);
else {
/* force tx path according to TTY mode when in call */
switch(adev->tty_mode) {
case TTY_MODE_FULL:
case TTY_MODE_HCO:
/* tx path from headset mic */
headphone_on = 0;
headset_on = 1;
speaker_on = 0;
earpiece_on = 0;
break;
case TTY_MODE_VCO:
/* tx path from device sub mic */
headphone_on = 0;
headset_on = 0;
speaker_on = 1;
earpiece_on = 0;
break;
case TTY_MODE_OFF:
default:
break;
}
if (headset_on || headphone_on || earpiece_on)
set_route_by_array(adev->mixer, vx_ul_amic_left, 1);
else if (speaker_on)
set_route_by_array(adev->mixer, vx_ul_amic_right, 1);
else
set_route_by_array(adev->mixer, vx_ul_amic_left, 0);
mixer_ctl_set_enum_by_string(adev->mixer_ctls.left_capture,
(earpiece_on || headphone_on) ? MIXER_MAIN_MIC :
(headset_on ? MIXER_HS_MIC : "Off"));
mixer_ctl_set_enum_by_string(adev->mixer_ctls.right_capture,
speaker_on ? MIXER_SUB_MIC : "Off");
set_input_volumes(adev, earpiece_on || headphone_on,
headset_on, speaker_on);
/* enable sidetone mixer capture if needed */
sidetone_capture_on = earpiece_on && adev->device_is_toro;
}
set_incall_device(adev);
/* Unmute VX_UL after the switch */
for (channel = 0; channel < 2; channel++) {
mixer_ctl_set_value(adev->mixer_ctls.voice_ul_volume,
channel, MIXER_ABE_GAIN_0DB);
}
}
mixer_ctl_set_value(adev->mixer_ctls.sidetone_capture, 0, sidetone_capture_on);
}
static void select_input_device(struct tuna_audio_device *adev)
{
int headset_on = 0;
int main_mic_on = 0;
int sub_mic_on = 0;
int bt_on = adev->devices & AUDIO_DEVICE_IN_ALL_SCO;
if (!bt_on) {
if ((adev->mode != AUDIO_MODE_IN_CALL) && (adev->active_input != 0)) {
/* sub mic is used for camcorder or VoIP on speaker phone */
sub_mic_on = (adev->active_input->source == AUDIO_SOURCE_CAMCORDER) ||
((adev->devices & AUDIO_DEVICE_OUT_SPEAKER) &&
(adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION));
}
if (!sub_mic_on) {
headset_on = adev->devices & AUDIO_DEVICE_IN_WIRED_HEADSET;
main_mic_on = adev->devices & AUDIO_DEVICE_IN_BUILTIN_MIC;
}
}
/* TODO: check how capture is possible during voice calls or if
* both use cases are mutually exclusive.
*/
if (bt_on)
set_route_by_array(adev->mixer, mm_ul2_bt, 1);
else {
/* Select front end */
if ((adev->active_input != 0) && (adev->active_input->aux_channels)) {
ALOGV("select input device(): multi-mic configuration main mic %s sub mic %s",
main_mic_on ? "ON" : "OFF", sub_mic_on ? "ON" : "OFF");
if (main_mic_on) {
set_route_by_array(adev->mixer, mm_ul2_amic_dual_main_sub, 1);
sub_mic_on = 1;
}
else if (sub_mic_on) {
set_route_by_array(adev->mixer, mm_ul2_amic_dual_sub_main, 1);
main_mic_on = 1;
}
else {
set_route_by_array(adev->mixer, mm_ul2_amic_dual_main_sub, 0);
}
} else {
ALOGV("select input device(): single mic configuration");
if (main_mic_on || headset_on)
set_route_by_array(adev->mixer, mm_ul2_amic_left, 1);
else if (sub_mic_on)
set_route_by_array(adev->mixer, mm_ul2_amic_right, 1);
else
set_route_by_array(adev->mixer, mm_ul2_amic_left, 0);
}
/* Select back end */
mixer_ctl_set_enum_by_string(adev->mixer_ctls.right_capture,
sub_mic_on ? MIXER_SUB_MIC : "Off");
mixer_ctl_set_enum_by_string(adev->mixer_ctls.left_capture,
main_mic_on ? MIXER_MAIN_MIC :
(headset_on ? MIXER_HS_MIC : "Off"));
}
set_input_volumes(adev, main_mic_on, headset_on, sub_mic_on);
}
/* must be called with hw device and output stream mutexes locked */
static int start_output_stream_low_latency(struct tuna_stream_out *out)
{
struct tuna_audio_device *adev = out->dev;
#ifdef PLAYBACK_MMAP
unsigned int flags = PCM_OUT | PCM_MMAP | PCM_NOIRQ;
#else
unsigned int flags = PCM_OUT;
#endif
int i;
bool success = true;
if (adev->mode != AUDIO_MODE_IN_CALL) {
select_output_device(adev);
}
/* default to low power: will be corrected in out_write if necessary before first write to
* tinyalsa.
*/
if (adev->devices & (AUDIO_DEVICE_OUT_ALL &
~(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET | AUDIO_DEVICE_OUT_AUX_DIGITAL))) {
/* Something not a dock in use */
out->config[PCM_NORMAL] = pcm_config_tones;
out->config[PCM_NORMAL].rate = MM_FULL_POWER_SAMPLING_RATE;
out->pcm[PCM_NORMAL] = pcm_open(CARD_TUNA_DEFAULT, PORT_TONES,
flags, &out->config[PCM_NORMAL]);
}
if (adev->devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) {
/* SPDIF output in use */
out->config[PCM_SPDIF] = pcm_config_tones;
out->config[PCM_SPDIF].rate = MM_FULL_POWER_SAMPLING_RATE;
out->pcm[PCM_SPDIF] = pcm_open(CARD_TUNA_DEFAULT, PORT_SPDIF,
flags, &out->config[PCM_SPDIF]);
}
if(adev->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
/* HDMI output in use */
out->config[PCM_HDMI] = pcm_config_tones;
out->config[PCM_HDMI].rate = MM_LOW_POWER_SAMPLING_RATE;
out->pcm[PCM_HDMI] = pcm_open(CARD_OMAP4_HDMI, PORT_HDMI,
flags, &out->config[PCM_HDMI]);
}
/* Close any PCMs that could not be opened properly and return an error */
for (i = 0; i < PCM_TOTAL; i++) {
if (out->pcm[i] && !pcm_is_ready(out->pcm[i])) {
ALOGE("cannot open pcm_out driver: %s", pcm_get_error(out->pcm[i]));
pcm_close(out->pcm[i]);
out->pcm[i] = NULL;
success = false;
}
}
if (success) {
out->buffer_frames = pcm_config_tones.period_size * 2;
if (out->buffer == NULL)
out->buffer = malloc(out->buffer_frames * audio_stream_frame_size(&out->stream.common));
if (adev->echo_reference != NULL)
out->echo_reference = adev->echo_reference;
out->resampler->reset(out->resampler);
return 0;
}
return -ENOMEM;
}
/* must be called with hw device and output stream mutexes locked */
static int start_output_stream_deep_buffer(struct tuna_stream_out *out)
{
struct tuna_audio_device *adev = out->dev;
if (adev->mode != AUDIO_MODE_IN_CALL) {
select_output_device(adev);
}
out->config[PCM_NORMAL] = pcm_config_mm;
out->config[PCM_NORMAL].rate = MM_FULL_POWER_SAMPLING_RATE;
out->pcm[PCM_NORMAL] = pcm_open(CARD_TUNA_DEFAULT, PORT_MM,
PCM_OUT | PCM_MMAP | PCM_NOIRQ, &out->config[PCM_NORMAL]);
if (out->pcm[PCM_NORMAL] && !pcm_is_ready(out->pcm[PCM_NORMAL])) {
ALOGE("cannot open pcm_out driver: %s", pcm_get_error(out->pcm[PCM_NORMAL]));
pcm_close(out->pcm[PCM_NORMAL]);
out->pcm[PCM_NORMAL] = NULL;
return -ENOMEM;
}
out->buffer_frames = pcm_config_mm.period_size * 2;
if (out->buffer == NULL)
out->buffer = malloc(out->buffer_frames * audio_stream_frame_size(&out->stream.common));
return 0;
}
static int check_input_parameters(uint32_t sample_rate, audio_format_t format, int channel_count)
{
if (format != AUDIO_FORMAT_PCM_16_BIT)
return -EINVAL;
if ((channel_count < 1) || (channel_count > 2))
return -EINVAL;
switch(sample_rate) {
case 8000:
case 11025:
case 16000:
case 22050:
case 24000:
case 32000:
case 44100:
case 48000:
break;
default:
return -EINVAL;
}
return 0;
}
static size_t get_input_buffer_size(uint32_t sample_rate, audio_format_t format, int channel_count)
{
size_t size;
size_t device_rate;
if (check_input_parameters(sample_rate, format, channel_count) != 0)
return 0;
/* take resampling into account and return the closest majoring
multiple of 16 frames, as audioflinger expects audio buffers to
be a multiple of 16 frames */
size = (pcm_config_mm_ul.period_size * sample_rate) / pcm_config_mm_ul.rate;
size = ((size + 15) / 16) * 16;
return size * channel_count * sizeof(short);
}
static void add_echo_reference(struct tuna_stream_out *out,
struct echo_reference_itfe *reference)
{
pthread_mutex_lock(&out->lock);
out->echo_reference = reference;
pthread_mutex_unlock(&out->lock);
}
static void remove_echo_reference(struct tuna_stream_out *out,
struct echo_reference_itfe *reference)
{
pthread_mutex_lock(&out->lock);
if (out->echo_reference == reference) {
/* stop writing to echo reference */
reference->write(reference, NULL);
out->echo_reference = NULL;
}
pthread_mutex_unlock(&out->lock);
}
static void put_echo_reference(struct tuna_audio_device *adev,
struct echo_reference_itfe *reference)
{
if (adev->echo_reference != NULL &&
reference == adev->echo_reference) {
/* echo reference is taken from the low latency output stream used
* for voice use cases */
if (adev->outputs[OUTPUT_LOW_LATENCY] != NULL &&
!adev->outputs[OUTPUT_LOW_LATENCY]->standby)
remove_echo_reference(adev->outputs[OUTPUT_LOW_LATENCY], reference);
release_echo_reference(reference);
adev->echo_reference = NULL;
}
}
static struct echo_reference_itfe *get_echo_reference(struct tuna_audio_device *adev,
audio_format_t format,
uint32_t channel_count,
uint32_t sampling_rate)
{
put_echo_reference(adev, adev->echo_reference);
/* echo reference is taken from the low latency output stream used
* for voice use cases */
if (adev->outputs[OUTPUT_LOW_LATENCY] != NULL &&
!adev->outputs[OUTPUT_LOW_LATENCY]->standby) {
struct audio_stream *stream =
&adev->outputs[OUTPUT_LOW_LATENCY]->stream.common;
uint32_t wr_channel_count = popcount(stream->get_channels(stream));
uint32_t wr_sampling_rate = stream->get_sample_rate(stream);
int status = create_echo_reference(AUDIO_FORMAT_PCM_16_BIT,
channel_count,
sampling_rate,
AUDIO_FORMAT_PCM_16_BIT,
wr_channel_count,
wr_sampling_rate,
&adev->echo_reference);
if (status == 0)
add_echo_reference(adev->outputs[OUTPUT_LOW_LATENCY],
adev->echo_reference);
}
return adev->echo_reference;
}
static int get_playback_delay(struct tuna_stream_out *out,
size_t frames,
struct echo_reference_buffer *buffer)
{
size_t kernel_frames;
int status;
int primary_pcm = 0;
/* Find the first active PCM to act as primary */
while ((primary_pcm < PCM_TOTAL) && !out->pcm[primary_pcm])
primary_pcm++;
status = pcm_get_htimestamp(out->pcm[primary_pcm], &kernel_frames, &buffer->time_stamp);
if (status < 0) {
buffer->time_stamp.tv_sec = 0;
buffer->time_stamp.tv_nsec = 0;
buffer->delay_ns = 0;
ALOGV("get_playback_delay(): pcm_get_htimestamp error,"
"setting playbackTimestamp to 0");
return status;
}
kernel_frames = pcm_get_buffer_size(out->pcm[primary_pcm]) - kernel_frames;
/* adjust render time stamp with delay added by current driver buffer.
* Add the duration of current frame as we want the render time of the last
* sample being written. */
buffer->delay_ns = (long)(((int64_t)(kernel_frames + frames)* 1000000000)/
MM_FULL_POWER_SAMPLING_RATE);
return 0;
}
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
return DEFAULT_OUT_SAMPLING_RATE;
}
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
return 0;
}
static size_t out_get_buffer_size_low_latency(const struct audio_stream *stream)
{
struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
/* take resampling into account and return the closest majoring
multiple of 16 frames, as audioflinger expects audio buffers to
be a multiple of 16 frames. Note: we use the default rate here
from pcm_config_tones.rate. */
size_t size = (SHORT_PERIOD_SIZE * DEFAULT_OUT_SAMPLING_RATE) / pcm_config_tones.rate;
size = ((size + 15) / 16) * 16;
return size * audio_stream_frame_size((struct audio_stream *)stream);
}
static size_t out_get_buffer_size_deep_buffer(const struct audio_stream *stream)
{
struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
/* take resampling into account and return the closest majoring
multiple of 16 frames, as audioflinger expects audio buffers to
be a multiple of 16 frames. Note: we use the default rate here
from pcm_config_mm.rate. */
size_t size = (LONG_PERIOD_SIZE * DEFAULT_OUT_SAMPLING_RATE) / pcm_config_mm.rate;
size = ((size + 15) / 16) * 16;
return size * audio_stream_frame_size((struct audio_stream *)stream);
}
static uint32_t out_get_channels(const struct audio_stream *stream)
{
return AUDIO_CHANNEL_OUT_STEREO;
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
return AUDIO_FORMAT_PCM_16_BIT;
}
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
return 0;
}
/* must be called with hw device and output stream mutexes locked */
static int do_output_standby(struct tuna_stream_out *out)
{
struct tuna_audio_device *adev = out->dev;
int i;
bool all_outputs_in_standby = true;
if (!out->standby) {
out->standby = 1;
for (i = 0; i < PCM_TOTAL; i++) {
if (out->pcm[i]) {
pcm_close(out->pcm[i]);
out->pcm[i] = NULL;
}
}
for (i = 0; i < OUTPUT_TOTAL; i++) {
if (adev->outputs[i] != NULL && !adev->outputs[i]->standby) {
all_outputs_in_standby = false;
break;
}
}
/* if in call, don't turn off the output stage. This will
be done when the call is ended */
if (all_outputs_in_standby && adev->mode != AUDIO_MODE_IN_CALL) {
set_route_by_array(adev->mixer, hs_output, 0);
set_route_by_array(adev->mixer, hf_output, 0);
}
/* stop writing to echo reference */
if (out->echo_reference != NULL) {
out->echo_reference->write(out->echo_reference, NULL);
out->echo_reference = NULL;
}
}
return 0;
}
static int out_standby(struct audio_stream *stream)
{
struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
int status;
pthread_mutex_lock(&out->dev->lock);
pthread_mutex_lock(&out->lock);
status = do_output_standby(out);
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&out->dev->lock);
return status;
}
static int out_dump(const struct audio_stream *stream, int fd)
{
return 0;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
struct tuna_audio_device *adev = out->dev;
struct tuna_stream_in *in;
struct str_parms *parms;
char *str;
char value[32];
int ret, val = 0;
bool force_input_standby = false;
parms = str_parms_create_str(kvpairs);
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
if (((adev->devices & AUDIO_DEVICE_OUT_ALL) != val) && (val != 0)) {
/* this is needed only when changing device on low latency output
* as other output streams are not used for voice use cases nor
* handle duplication to HDMI or SPDIF */
if (out == adev->outputs[OUTPUT_LOW_LATENCY] && !out->standby) {
/* a change in output device may change the microphone selection */
if (adev->active_input &&
adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
force_input_standby = true;
}
/* force standby if moving to/from HDMI/SPDIF or if the output
* device changes when in HDMI/SPDIF mode */
/* FIXME workaround for audio being dropped when switching path without forcing standby
* (several hundred ms of audio can be lost: e.g beginning of a ringtone. We must understand
* the root cause in audio HAL, driver or ABE.
if (((val & AUDIO_DEVICE_OUT_AUX_DIGITAL) ^
(adev->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)) ||
((val & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) ^
(adev->devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) ||
(adev->devices & (AUDIO_DEVICE_OUT_AUX_DIGITAL |
AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)))
*/
if (((val & AUDIO_DEVICE_OUT_AUX_DIGITAL) ^
(adev->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)) ||
((val & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) ^
(adev->devices & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) ||
(adev->devices & (AUDIO_DEVICE_OUT_AUX_DIGITAL |
AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) ||
((val & AUDIO_DEVICE_OUT_SPEAKER) ^
(adev->devices & AUDIO_DEVICE_OUT_SPEAKER)))
do_output_standby(out);
}
adev->devices &= ~AUDIO_DEVICE_OUT_ALL;
adev->devices |= val;
select_output_device(adev);
}
pthread_mutex_unlock(&out->lock);
if (force_input_standby) {
in = adev->active_input;
pthread_mutex_lock(&in->lock);
do_input_standby(in);
pthread_mutex_unlock(&in->lock);
}
pthread_mutex_unlock(&adev->lock);
}
str_parms_destroy(parms);
return ret;
}
static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
{
return strdup("");
}
static uint32_t out_get_latency_low_latency(const struct audio_stream_out *stream)
{
struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
/* Note: we use the default rate here from pcm_config_mm.rate */
return (SHORT_PERIOD_SIZE * PLAYBACK_SHORT_PERIOD_COUNT * 1000) / pcm_config_tones.rate;
}
static uint32_t out_get_latency_deep_buffer(const struct audio_stream_out *stream)
{
struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
/* Note: we use the default rate here from pcm_config_mm.rate */
return (LONG_PERIOD_SIZE * PLAYBACK_LONG_PERIOD_COUNT * 1000) / pcm_config_mm.rate;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
return -ENOSYS;
}
static ssize_t out_write_low_latency(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
int ret;
struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
struct tuna_audio_device *adev = out->dev;
size_t frame_size = audio_stream_frame_size(&out->stream.common);
size_t in_frames = bytes / frame_size;
size_t out_frames = in_frames;
bool force_input_standby = false;
struct tuna_stream_in *in;
int i;
/* acquiring hw device mutex systematically is useful if a low priority thread is waiting
* on the output stream mutex - e.g. executing select_mode() while holding the hw device
* mutex
*/
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
if (out->standby) {
ret = start_output_stream_low_latency(out);
if (ret != 0) {
pthread_mutex_unlock(&adev->lock);
goto exit;
}
out->standby = 0;
/* a change in output device may change the microphone selection */
if (adev->active_input &&
adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION)
force_input_standby = true;
}
pthread_mutex_unlock(&adev->lock);
for (i = 0; i < PCM_TOTAL; i++) {
/* only use resampler if required */
if (out->pcm[i] && (out->config[i].rate != DEFAULT_OUT_SAMPLING_RATE)) {
out_frames = out->buffer_frames;
out->resampler->resample_from_input(out->resampler,
(int16_t *)buffer,
&in_frames,
(int16_t *)out->buffer,
&out_frames);
break;
}
}
if (out->echo_reference != NULL) {
struct echo_reference_buffer b;
b.raw = (void *)buffer;
b.frame_count = in_frames;
get_playback_delay(out, out_frames, &b);
out->echo_reference->write(out->echo_reference, &b);
}
/* Write to all active PCMs */
for (i = 0; i < PCM_TOTAL; i++) {
if (out->pcm[i]) {
if (out->config[i].rate == DEFAULT_OUT_SAMPLING_RATE) {
/* PCM uses native sample rate */
ret = PCM_WRITE(out->pcm[i], (void *)buffer, bytes);
} else {
/* PCM needs resampler */
ret = PCM_WRITE(out->pcm[i], (void *)out->buffer, out_frames * frame_size);
}
if (ret)
break;
}
}
exit:
pthread_mutex_unlock(&out->lock);
if (ret != 0) {
usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) /
out_get_sample_rate(&stream->common));
}
if (force_input_standby) {
pthread_mutex_lock(&adev->lock);
if (adev->active_input) {
in = adev->active_input;
pthread_mutex_lock(&in->lock);
do_input_standby(in);
pthread_mutex_unlock(&in->lock);
}
pthread_mutex_unlock(&adev->lock);
}
return bytes;
}
static ssize_t out_write_deep_buffer(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
int ret;
struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
struct tuna_audio_device *adev = out->dev;
size_t frame_size = audio_stream_frame_size(&out->stream.common);
size_t in_frames = bytes / frame_size;
/* acquiring hw device mutex systematically is useful if a low priority thread is waiting
* on the output stream mutex - e.g. executing select_mode() while holding the hw device
* mutex
*/
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
if (out->standby) {
ret = start_output_stream_deep_buffer(out);
if (ret != 0) {
pthread_mutex_unlock(&adev->lock);
goto exit;
}
out->standby = 0;
}
pthread_mutex_unlock(&adev->lock);
/* only use resampler if required */
if (out->config[PCM_NORMAL].rate != DEFAULT_OUT_SAMPLING_RATE) {
size_t out_frames = out->buffer_frames;
out->resampler->resample_from_input(out->resampler,
(int16_t *)buffer,
&in_frames,
(int16_t *)out->buffer,
&out_frames);
ret = pcm_mmap_write(out->pcm[PCM_NORMAL], (void *)out->buffer, out_frames * frame_size);
} else
ret = pcm_mmap_write(out->pcm[PCM_NORMAL], (void *)buffer, bytes);
exit:
pthread_mutex_unlock(&out->lock);
if (ret != 0) {
usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) /
out_get_sample_rate(&stream->common));
}
return bytes;
}
static int out_get_render_position(const struct audio_stream_out *stream,
uint32_t *dsp_frames)
{
return -EINVAL;
}
static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
/** audio_stream_in implementation **/
/* must be called with hw device and input stream mutexes locked */
static int start_input_stream(struct tuna_stream_in *in)
{
int ret = 0;
struct tuna_audio_device *adev = in->dev;
adev->active_input = in;
if (adev->mode != AUDIO_MODE_IN_CALL) {
adev->devices &= ~AUDIO_DEVICE_IN_ALL;
adev->devices |= in->device;
select_input_device(adev);
}
if (in->aux_channels_changed)
{
in->aux_channels_changed = false;
in->config.channels = popcount(in->main_channels | in->aux_channels);
if (in->resampler) {
/* release and recreate the resampler with the new number of channel of the input */
release_resampler(in->resampler);
in->resampler = NULL;
ret = create_resampler(in->config.rate,
in->requested_rate,
in->config.channels,
RESAMPLER_QUALITY_DEFAULT,
&in->buf_provider,
&in->resampler);
}
ALOGV("start_input_stream(): New channel configuration, "
"main_channels = [%04x], aux_channels = [%04x], config.channels = %d",
in->main_channels, in->aux_channels, in->config.channels);
}
if (in->need_echo_reference && in->echo_reference == NULL)
in->echo_reference = get_echo_reference(adev,
AUDIO_FORMAT_PCM_16_BIT,
in->config.channels,
in->requested_rate);
/* this assumes routing is done previously */
in->pcm = pcm_open(0, PORT_MM2_UL, PCM_IN, &in->config);
if (!pcm_is_ready(in->pcm)) {
ALOGE("cannot open pcm_in driver: %s", pcm_get_error(in->pcm));
pcm_close(in->pcm);
adev->active_input = NULL;
return -ENOMEM;
}
/* force read and proc buf reallocation case of frame size or channel count change */
in->read_buf_frames = 0;
in->read_buf_size = 0;
in->proc_buf_frames = 0;
in->proc_buf_size = 0;
/* if no supported sample rate is available, use the resampler */
if (in->resampler) {
in->resampler->reset(in->resampler);
}
return 0;
}
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
struct tuna_stream_in *in = (struct tuna_stream_in *)stream;
return in->requested_rate;
}
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
return 0;
}
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
struct tuna_stream_in *in = (struct tuna_stream_in *)stream;
return get_input_buffer_size(in->requested_rate,
AUDIO_FORMAT_PCM_16_BIT,
popcount(in->main_channels));
}
static uint32_t in_get_channels(const struct audio_stream *stream)
{
struct tuna_stream_in *in = (struct tuna_stream_in *)stream;
return in->main_channels;
}
static audio_format_t in_get_format(const struct audio_stream *stream)
{
return AUDIO_FORMAT_PCM_16_BIT;
}
static int in_set_format(struct audio_stream *stream, audio_format_t format)
{
return 0;
}
/* must be called with hw device and input stream mutexes locked */
static int do_input_standby(struct tuna_stream_in *in)
{
struct tuna_audio_device *adev = in->dev;
if (!in->standby) {
pcm_close(in->pcm);
in->pcm = NULL;
adev->active_input = 0;
if (adev->mode != AUDIO_MODE_IN_CALL) {
adev->devices &= ~AUDIO_DEVICE_IN_ALL;
select_input_device(adev);
}
if (in->echo_reference != NULL) {
/* stop reading from echo reference */
in->echo_reference->read(in->echo_reference, NULL);
put_echo_reference(adev, in->echo_reference);
in->echo_reference = NULL;
}
in->standby = 1;
}
return 0;
}
static int in_standby(struct audio_stream *stream)
{
struct tuna_stream_in *in = (struct tuna_stream_in *)stream;
int status;
pthread_mutex_lock(&in->dev->lock);
pthread_mutex_lock(&in->lock);
status = do_input_standby(in);
pthread_mutex_unlock(&in->lock);
pthread_mutex_unlock(&in->dev->lock);
return status;
}
static int in_dump(const struct audio_stream *stream, int fd)
{
return 0;
}
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
struct tuna_stream_in *in = (struct tuna_stream_in *)stream;
struct tuna_audio_device *adev = in->dev;
struct str_parms *parms;
char *str;
char value[32];
int ret, val = 0;
bool do_standby = false;
parms = str_parms_create_str(kvpairs);
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value));
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&in->lock);
if (ret >= 0) {
val = atoi(value);
/* no audio source uses val == 0 */
if ((in->source != val) && (val != 0)) {
in->source = val;
do_standby = true;
}
}
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
if ((in->device != val) && (val != 0)) {
in->device = val;
do_standby = true;
/* make sure new device selection is incompatible with multi-mic pre processing
* configuration */
in_update_aux_channels(in, NULL);
}
}
if (do_standby)
do_input_standby(in);
pthread_mutex_unlock(&in->lock);
pthread_mutex_unlock(&adev->lock);
str_parms_destroy(parms);
return ret;
}
static char * in_get_parameters(const struct audio_stream *stream,
const char *keys)
{
return strdup("");
}
static int in_set_gain(struct audio_stream_in *stream, float gain)
{
return 0;
}
static void get_capture_delay(struct tuna_stream_in *in,
size_t frames,
struct echo_reference_buffer *buffer)
{
/* read frames available in kernel driver buffer */
size_t kernel_frames;
struct timespec tstamp;
long buf_delay;
long rsmp_delay;
long kernel_delay;
long delay_ns;
if (pcm_get_htimestamp(in->pcm, &kernel_frames, &tstamp) < 0) {
buffer->time_stamp.tv_sec = 0;
buffer->time_stamp.tv_nsec = 0;
buffer->delay_ns = 0;
ALOGW("read get_capture_delay(): pcm_htimestamp error");
return;
}
/* read frames available in audio HAL input buffer
* add number of frames being read as we want the capture time of first sample
* in current buffer */
/* frames in in->buffer are at driver sampling rate while frames in in->proc_buf are
* at requested sampling rate */
buf_delay = (long)(((int64_t)(in->read_buf_frames) * 1000000000) / in->config.rate +
((int64_t)(in->proc_buf_frames) * 1000000000) /
in->requested_rate);
/* add delay introduced by resampler */
rsmp_delay = 0;
if (in->resampler) {
rsmp_delay = in->resampler->delay_ns(in->resampler);
}
kernel_delay = (long)(((int64_t)kernel_frames * 1000000000) / in->config.rate);
delay_ns = kernel_delay + buf_delay + rsmp_delay;
buffer->time_stamp = tstamp;
buffer->delay_ns = delay_ns;
ALOGV("get_capture_delay time_stamp = [%ld].[%ld], delay_ns: [%d],"
" kernel_delay:[%ld], buf_delay:[%ld], rsmp_delay:[%ld], kernel_frames:[%d], "
"in->read_buf_frames:[%d], in->proc_buf_frames:[%d], frames:[%d]",
buffer->time_stamp.tv_sec , buffer->time_stamp.tv_nsec, buffer->delay_ns,
kernel_delay, buf_delay, rsmp_delay, kernel_frames,
in->read_buf_frames, in->proc_buf_frames, frames);
}
static int32_t update_echo_reference(struct tuna_stream_in *in, size_t frames)
{
struct echo_reference_buffer b;
b.delay_ns = 0;
ALOGV("update_echo_reference, frames = [%d], in->ref_buf_frames = [%d], "
"b.frame_count = [%d]",
frames, in->ref_buf_frames, frames - in->ref_buf_frames);
if (in->ref_buf_frames < frames) {
if (in->ref_buf_size < frames) {
in->ref_buf_size = frames;
in->ref_buf = (int16_t *)realloc(in->ref_buf, pcm_frames_to_bytes(in->pcm, frames));
ALOG_ASSERT((in->ref_buf != NULL),
"update_echo_reference() failed to reallocate ref_buf");
ALOGV("update_echo_reference(): ref_buf %p extended to %d bytes",
in->ref_buf, pcm_frames_to_bytes(in->pcm, frames));
}
b.frame_count = frames - in->ref_buf_frames;
b.raw = (void *)(in->ref_buf + in->ref_buf_frames * in->config.channels);
get_capture_delay(in, frames, &b);
if (in->echo_reference->read(in->echo_reference, &b) == 0)
{
in->ref_buf_frames += b.frame_count;
ALOGV("update_echo_reference(): in->ref_buf_frames:[%d], "
"in->ref_buf_size:[%d], frames:[%d], b.frame_count:[%d]",
in->ref_buf_frames, in->ref_buf_size, frames, b.frame_count);
}
} else
ALOGW("update_echo_reference(): NOT enough frames to read ref buffer");
return b.delay_ns;
}
static int set_preprocessor_param(effect_handle_t handle,
effect_param_t *param)
{
uint32_t size = sizeof(int);
uint32_t psize = ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) +
param->vsize;
int status = (*handle)->command(handle,
EFFECT_CMD_SET_PARAM,
sizeof (effect_param_t) + psize,
param,
&size,
&param->status);
if (status == 0)
status = param->status;
return status;
}
static int set_preprocessor_echo_delay(effect_handle_t handle,
int32_t delay_us)
{
uint32_t buf[sizeof(effect_param_t) / sizeof(uint32_t) + 2];
effect_param_t *param = (effect_param_t *)buf;
param->psize = sizeof(uint32_t);
param->vsize = sizeof(uint32_t);
*(uint32_t *)param->data = AEC_PARAM_ECHO_DELAY;
*((int32_t *)param->data + 1) = delay_us;
return set_preprocessor_param(handle, param);
}
static void push_echo_reference(struct tuna_stream_in *in, size_t frames)
{
/* read frames from echo reference buffer and update echo delay
* in->ref_buf_frames is updated with frames available in in->ref_buf */
int32_t delay_us = update_echo_reference(in, frames)/1000;
int i;
audio_buffer_t buf;
if (in->ref_buf_frames < frames)
frames = in->ref_buf_frames;
buf.frameCount = frames;
buf.raw = in->ref_buf;
for (i = 0; i < in->num_preprocessors; i++) {
if ((*in->preprocessors[i].effect_itfe)->process_reverse == NULL)
continue;
(*in->preprocessors[i].effect_itfe)->process_reverse(in->preprocessors[i].effect_itfe,
&buf,
NULL);
set_preprocessor_echo_delay(in->preprocessors[i].effect_itfe, delay_us);
}
in->ref_buf_frames -= buf.frameCount;
if (in->ref_buf_frames) {
memcpy(in->ref_buf,
in->ref_buf + buf.frameCount * in->config.channels,
in->ref_buf_frames * in->config.channels * sizeof(int16_t));
}
}
static int get_next_buffer(struct resampler_buffer_provider *buffer_provider,
struct resampler_buffer* buffer)
{
struct tuna_stream_in *in;
if (buffer_provider == NULL || buffer == NULL)
return -EINVAL;
in = (struct tuna_stream_in *)((char *)buffer_provider -
offsetof(struct tuna_stream_in, buf_provider));
if (in->pcm == NULL) {
buffer->raw = NULL;
buffer->frame_count = 0;
in->read_status = -ENODEV;
return -ENODEV;
}
if (in->read_buf_frames == 0) {
size_t size_in_bytes = pcm_frames_to_bytes(in->pcm, in->config.period_size);
if (in->read_buf_size < in->config.period_size) {
in->read_buf_size = in->config.period_size;
in->read_buf = (int16_t *) realloc(in->read_buf, size_in_bytes);
ALOG_ASSERT((in->read_buf != NULL),
"get_next_buffer() failed to reallocate read_buf");
ALOGV("get_next_buffer(): read_buf %p extended to %d bytes",
in->read_buf, size_in_bytes);
}
in->read_status = pcm_read(in->pcm, (void*)in->read_buf, size_in_bytes);
if (in->read_status != 0) {
ALOGE("get_next_buffer() pcm_read error %d", in->read_status);
buffer->raw = NULL;
buffer->frame_count = 0;
return in->read_status;
}
in->read_buf_frames = in->config.period_size;
}
buffer->frame_count = (buffer->frame_count > in->read_buf_frames) ?
in->read_buf_frames : buffer->frame_count;
buffer->i16 = in->read_buf + (in->config.period_size - in->read_buf_frames) *
in->config.channels;
return in->read_status;
}
static void release_buffer(struct resampler_buffer_provider *buffer_provider,
struct resampler_buffer* buffer)
{
struct tuna_stream_in *in;
if (buffer_provider == NULL || buffer == NULL)
return;
in = (struct tuna_stream_in *)((char *)buffer_provider -
offsetof(struct tuna_stream_in, buf_provider));
in->read_buf_frames -= buffer->frame_count;
}
/* read_frames() reads frames from kernel driver, down samples to capture rate
* if necessary and output the number of frames requested to the buffer specified */
static ssize_t read_frames(struct tuna_stream_in *in, void *buffer, ssize_t frames)
{
ssize_t frames_wr = 0;
while (frames_wr < frames) {
size_t frames_rd = frames - frames_wr;
if (in->resampler != NULL) {
in->resampler->resample_from_provider(in->resampler,
(int16_t *)((char *)buffer +
pcm_frames_to_bytes(in->pcm ,frames_wr)),
&frames_rd);
} else {
struct resampler_buffer buf = {
{ raw : NULL, },
frame_count : frames_rd,
};
get_next_buffer(&in->buf_provider, &buf);
if (buf.raw != NULL) {
memcpy((char *)buffer +
pcm_frames_to_bytes(in->pcm, frames_wr),
buf.raw,
pcm_frames_to_bytes(in->pcm, buf.frame_count));
frames_rd = buf.frame_count;
}
release_buffer(&in->buf_provider, &buf);
}
/* in->read_status is updated by getNextBuffer() also called by
* in->resampler->resample_from_provider() */
if (in->read_status != 0)
return in->read_status;
frames_wr += frames_rd;
}
return frames_wr;
}
/* process_frames() reads frames from kernel driver (via read_frames()),
* calls the active audio pre processings and output the number of frames requested
* to the buffer specified */
static ssize_t process_frames(struct tuna_stream_in *in, void* buffer, ssize_t frames)
{
ssize_t frames_wr = 0;
audio_buffer_t in_buf;
audio_buffer_t out_buf;
int i;
bool has_aux_channels = (~in->main_channels & in->aux_channels);
void *proc_buf_out;
if (has_aux_channels)
proc_buf_out = in->proc_buf_out;
else
proc_buf_out = buffer;
/* since all the processing below is done in frames and using the config.channels
* as the number of channels, no changes is required in case aux_channels are present */
while (frames_wr < frames) {
/* first reload enough frames at the end of process input buffer */
if (in->proc_buf_frames < (size_t)frames) {
ssize_t frames_rd;
if (in->proc_buf_size < (size_t)frames) {
size_t size_in_bytes = pcm_frames_to_bytes(in->pcm, frames);
in->proc_buf_size = (size_t)frames;
in->proc_buf_in = (int16_t *)realloc(in->proc_buf_in, size_in_bytes);
ALOG_ASSERT((in->proc_buf_in != NULL),
"process_frames() failed to reallocate proc_buf_in");
if (has_aux_channels) {
in->proc_buf_out = (int16_t *)realloc(in->proc_buf_out, size_in_bytes);
ALOG_ASSERT((in->proc_buf_out != NULL),
"process_frames() failed to reallocate proc_buf_out");
proc_buf_out = in->proc_buf_out;
}
ALOGV("process_frames(): proc_buf_in %p extended to %d bytes",
in->proc_buf_in, size_in_bytes);
}
frames_rd = read_frames(in,
in->proc_buf_in +
in->proc_buf_frames * in->config.channels,
frames - in->proc_buf_frames);
if (frames_rd < 0) {
frames_wr = frames_rd;
break;
}
in->proc_buf_frames += frames_rd;
}
if (in->echo_reference != NULL)
push_echo_reference(in, in->proc_buf_frames);
/* in_buf.frameCount and out_buf.frameCount indicate respectively
* the maximum number of frames to be consumed and produced by process() */
in_buf.frameCount = in->proc_buf_frames;
in_buf.s16 = in->proc_buf_in;
out_buf.frameCount = frames - frames_wr;
out_buf.s16 = (int16_t *)proc_buf_out + frames_wr * in->config.channels;
/* FIXME: this works because of current pre processing library implementation that
* does the actual process only when the last enabled effect process is called.
* The generic solution is to have an output buffer for each effect and pass it as
* input to the next.
*/
for (i = 0; i < in->num_preprocessors; i++) {
(*in->preprocessors[i].effect_itfe)->process(in->preprocessors[i].effect_itfe,
&in_buf,
&out_buf);
}
/* process() has updated the number of frames consumed and produced in
* in_buf.frameCount and out_buf.frameCount respectively
* move remaining frames to the beginning of in->proc_buf_in */
in->proc_buf_frames -= in_buf.frameCount;
if (in->proc_buf_frames) {
memcpy(in->proc_buf_in,
in->proc_buf_in + in_buf.frameCount * in->config.channels,
in->proc_buf_frames * in->config.channels * sizeof(int16_t));
}
/* if not enough frames were passed to process(), read more and retry. */
if (out_buf.frameCount == 0) {
ALOGW("No frames produced by preproc");
continue;
}
if ((frames_wr + (ssize_t)out_buf.frameCount) <= frames) {
frames_wr += out_buf.frameCount;
} else {
/* The effect does not comply to the API. In theory, we should never end up here! */
ALOGE("preprocessing produced too many frames: %d + %d > %d !",
(unsigned int)frames_wr, out_buf.frameCount, (unsigned int)frames);
frames_wr = frames;
}
}
/* Remove aux_channels that have been added on top of main_channels
* Assumption is made that the channels are interleaved and that the main
* channels are first. */
if (has_aux_channels)
{
size_t src_channels = in->config.channels;
size_t dst_channels = popcount(in->main_channels);
int16_t* src_buffer = (int16_t *)proc_buf_out;
int16_t* dst_buffer = (int16_t *)buffer;
if (dst_channels == 1) {
for (i = frames_wr; i > 0; i--)
{
*dst_buffer++ = *src_buffer;
src_buffer += src_channels;
}
} else {
for (i = frames_wr; i > 0; i--)
{
memcpy(dst_buffer, src_buffer, dst_channels*sizeof(int16_t));
dst_buffer += dst_channels;
src_buffer += src_channels;
}
}
}
return frames_wr;
}
static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
size_t bytes)
{
int ret = 0;
struct tuna_stream_in *in = (struct tuna_stream_in *)stream;
struct tuna_audio_device *adev = in->dev;
size_t frames_rq = bytes / audio_stream_frame_size(&stream->common);
/* acquiring hw device mutex systematically is useful if a low priority thread is waiting
* on the input stream mutex - e.g. executing select_mode() while holding the hw device
* mutex
*/
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&in->lock);
if (in->standby) {
ret = start_input_stream(in);
if (ret == 0)
in->standby = 0;
}
pthread_mutex_unlock(&adev->lock);
if (ret < 0)
goto exit;
if (in->num_preprocessors != 0)
ret = process_frames(in, buffer, frames_rq);
else if (in->resampler != NULL)
ret = read_frames(in, buffer, frames_rq);
else
ret = pcm_read(in->pcm, buffer, bytes);
if (ret > 0)
ret = 0;
if (ret == 0 && adev->mic_mute)
memset(buffer, 0, bytes);
exit:
if (ret < 0)
usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) /
in_get_sample_rate(&stream->common));
pthread_mutex_unlock(&in->lock);
return bytes;
}
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
{
return 0;
}
#define GET_COMMAND_STATUS(status, fct_status, cmd_status) \
do { \
if (fct_status != 0) \
status = fct_status; \
else if (cmd_status != 0) \
status = cmd_status; \
} while(0)
static int in_configure_reverse(struct tuna_stream_in *in)
{
int32_t cmd_status;
uint32_t size = sizeof(int);
effect_config_t config;
int32_t status = 0;
int32_t fct_status = 0;
int i;
if (in->num_preprocessors > 0) {
config.inputCfg.channels = in->main_channels;
config.outputCfg.channels = in->main_channels;
config.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT;
config.inputCfg.samplingRate = in->requested_rate;
config.outputCfg.samplingRate = in->requested_rate;
config.inputCfg.mask =
( EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT );
config.outputCfg.mask =
( EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT );
for (i = 0; i < in->num_preprocessors; i++)
{
if ((*in->preprocessors[i].effect_itfe)->process_reverse == NULL)
continue;
fct_status = (*(in->preprocessors[i].effect_itfe))->command(
in->preprocessors[i].effect_itfe,
EFFECT_CMD_SET_CONFIG_REVERSE,
sizeof(effect_config_t),
&config,
&size,
&cmd_status);
GET_COMMAND_STATUS(status, fct_status, cmd_status);
}
}
return status;
}
#define MAX_NUM_CHANNEL_CONFIGS 10
static void in_read_audio_effect_channel_configs(struct tuna_stream_in *in,
struct effect_info_s *effect_info)
{
/* size and format of the cmd are defined in hardware/audio_effect.h */
effect_handle_t effect = effect_info->effect_itfe;
uint32_t cmd_size = 2 * sizeof(uint32_t);
uint32_t cmd[] = { EFFECT_FEATURE_AUX_CHANNELS, MAX_NUM_CHANNEL_CONFIGS };
/* reply = status + number of configs (n) + n x channel_config_t */
uint32_t reply_size =
2 * sizeof(uint32_t) + (MAX_NUM_CHANNEL_CONFIGS * sizeof(channel_config_t));
int32_t reply[reply_size];
int32_t cmd_status;
ALOG_ASSERT((effect_info->num_channel_configs == 0),
"in_read_audio_effect_channel_configs() num_channel_configs not cleared");
ALOG_ASSERT((effect_info->channel_configs == NULL),
"in_read_audio_effect_channel_configs() channel_configs not cleared");
/* if this command is not supported, then the effect is supposed to return -EINVAL.
* This error will be interpreted as if the effect supports the main_channels but does not
* support any aux_channels */
cmd_status = (*effect)->command(effect,
EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS,
cmd_size,
(void*)&cmd,
&reply_size,
(void*)&reply);
if (cmd_status != 0) {
ALOGV("in_read_audio_effect_channel_configs(): "
"fx->command returned %d", cmd_status);
return;
}
if (reply[0] != 0) {
ALOGW("in_read_audio_effect_channel_configs(): "
"command EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS error %d num configs %d",
reply[0], (reply[0] == -ENOMEM) ? reply[1] : MAX_NUM_CHANNEL_CONFIGS);
return;
}
/* the feature is not supported */
ALOGV("in_read_audio_effect_channel_configs()(): "
"Feature supported and adding %d channel configs to the list", reply[1]);
effect_info->num_channel_configs = reply[1];
effect_info->channel_configs =
(channel_config_t *) malloc(sizeof(channel_config_t) * reply[1]); /* n x configs */
memcpy(effect_info->channel_configs, (reply + 2), sizeof(channel_config_t) * reply[1]);
}
static uint32_t in_get_aux_channels(struct tuna_stream_in *in)
{
int i;
channel_config_t new_chcfg = {0, 0};
if (in->num_preprocessors == 0)
return 0;
/* do not enable dual mic configurations when capturing from other microphones than
* main or sub */
if (!(in->device & (AUDIO_DEVICE_IN_BUILTIN_MIC | AUDIO_DEVICE_IN_BACK_MIC)))
return 0;
/* retain most complex aux channels configuration compatible with requested main channels and
* supported by audio driver and all pre processors */
for (i = 0; i < NUM_IN_AUX_CNL_CONFIGS; i++) {
channel_config_t *cur_chcfg = &in_aux_cnl_configs[i];
if (cur_chcfg->main_channels == in->main_channels) {
size_t match_cnt;
size_t idx_preproc;
for (idx_preproc = 0, match_cnt = 0;
/* no need to continue if at least one preprocessor doesn't match */
idx_preproc < (size_t)in->num_preprocessors && match_cnt == idx_preproc;
idx_preproc++) {
struct effect_info_s *effect_info = &in->preprocessors[idx_preproc];
size_t idx_chcfg;
for (idx_chcfg = 0; idx_chcfg < effect_info->num_channel_configs; idx_chcfg++) {
if (memcmp(effect_info->channel_configs + idx_chcfg,
cur_chcfg,
sizeof(channel_config_t)) == 0) {
match_cnt++;
break;
}
}
}
/* if all preprocessors match, we have a candidate */
if (match_cnt == (size_t)in->num_preprocessors) {
/* retain most complex aux channels configuration */
if (popcount(cur_chcfg->aux_channels) > popcount(new_chcfg.aux_channels)) {
new_chcfg = *cur_chcfg;
}
}
}
}
ALOGV("in_get_aux_channels(): return %04x", new_chcfg.aux_channels);
return new_chcfg.aux_channels;
}
static int in_configure_effect_channels(effect_handle_t effect,
channel_config_t *channel_config)
{
int status = 0;
int fct_status;
int32_t cmd_status;
uint32_t reply_size;
effect_config_t config;
uint32_t cmd[(sizeof(uint32_t) + sizeof(channel_config_t) - 1) / sizeof(uint32_t) + 1];
ALOGV("in_configure_effect_channels(): configure effect with channels: [%04x][%04x]",
channel_config->main_channels,
channel_config->aux_channels);
config.inputCfg.mask = EFFECT_CONFIG_CHANNELS;
config.outputCfg.mask = EFFECT_CONFIG_CHANNELS;
reply_size = sizeof(effect_config_t);
fct_status = (*effect)->command(effect,
EFFECT_CMD_GET_CONFIG,
0,
NULL,
&reply_size,
&config);
if (fct_status != 0) {
ALOGE("in_configure_effect_channels(): EFFECT_CMD_GET_CONFIG failed");
return fct_status;
}
config.inputCfg.channels = channel_config->main_channels | channel_config->aux_channels;
config.outputCfg.channels = config.inputCfg.channels;
reply_size = sizeof(uint32_t);
fct_status = (*effect)->command(effect,
EFFECT_CMD_SET_CONFIG,
sizeof(effect_config_t),
&config,
&reply_size,
&cmd_status);
GET_COMMAND_STATUS(status, fct_status, cmd_status);
cmd[0] = EFFECT_FEATURE_AUX_CHANNELS;
memcpy(cmd + 1, channel_config, sizeof(channel_config_t));
reply_size = sizeof(uint32_t);
fct_status = (*effect)->command(effect,
EFFECT_CMD_SET_FEATURE_CONFIG,
sizeof(cmd), //sizeof(uint32_t) + sizeof(channel_config_t),
cmd,
&reply_size,
&cmd_status);
GET_COMMAND_STATUS(status, fct_status, cmd_status);
/* some implementations need to be re-enabled after a config change */
reply_size = sizeof(uint32_t);
fct_status = (*effect)->command(effect,
EFFECT_CMD_ENABLE,
0,
NULL,
&reply_size,
&cmd_status);
GET_COMMAND_STATUS(status, fct_status, cmd_status);
return status;
}
static int in_reconfigure_channels(struct tuna_stream_in *in,
effect_handle_t effect,
channel_config_t *channel_config,
bool config_changed) {
int status = 0;
ALOGV("in_reconfigure_channels(): config_changed %d effect %p",
config_changed, effect);
/* if config changed, reconfigure all previously added effects */
if (config_changed) {
int i;
for (i = 0; i < in->num_preprocessors; i++)
{
int cur_status = in_configure_effect_channels(in->preprocessors[i].effect_itfe,
channel_config);
if (cur_status != 0) {
ALOGV("in_reconfigure_channels(): error %d configuring effect "
"%d with channels: [%04x][%04x]",
cur_status,
i,
channel_config->main_channels,
channel_config->aux_channels);
status = cur_status;
}
}
} else if (effect != NULL && channel_config->aux_channels) {
/* if aux channels config did not change but aux channels are present,
* we still need to configure the effect being added */
status = in_configure_effect_channels(effect, channel_config);
}
return status;
}
static void in_update_aux_channels(struct tuna_stream_in *in,
effect_handle_t effect)
{
uint32_t aux_channels;
channel_config_t channel_config;
int status;
aux_channels = in_get_aux_channels(in);
channel_config.main_channels = in->main_channels;
channel_config.aux_channels = aux_channels;
status = in_reconfigure_channels(in,
effect,
&channel_config,
(aux_channels != in->aux_channels));
if (status != 0) {
ALOGV("in_update_aux_channels(): in_reconfigure_channels error %d", status);
/* resetting aux channels configuration */
aux_channels = 0;
channel_config.aux_channels = 0;
in_reconfigure_channels(in, effect, &channel_config, true);
}
if (in->aux_channels != aux_channels) {
in->aux_channels_changed = true;
in->aux_channels = aux_channels;
do_input_standby(in);
}
}
static int in_add_audio_effect(const struct audio_stream *stream,
effect_handle_t effect)
{
struct tuna_stream_in *in = (struct tuna_stream_in *)stream;
int status;
effect_descriptor_t desc;
pthread_mutex_lock(&in->dev->lock);
pthread_mutex_lock(&in->lock);
if (in->num_preprocessors >= MAX_PREPROCESSORS) {
status = -ENOSYS;
goto exit;
}
status = (*effect)->get_descriptor(effect, &desc);
if (status != 0)
goto exit;
in->preprocessors[in->num_preprocessors].effect_itfe = effect;
/* add the supported channel of the effect in the channel_configs */
in_read_audio_effect_channel_configs(in, &in->preprocessors[in->num_preprocessors]);
in->num_preprocessors++;
/* check compatibility between main channel supported and possible auxiliary channels */
in_update_aux_channels(in, effect);
ALOGV("in_add_audio_effect(), effect type: %08x", desc.type.timeLow);
if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) {
in->need_echo_reference = true;
do_input_standby(in);
in_configure_reverse(in);
}
exit:
ALOGW_IF(status != 0, "in_add_audio_effect() error %d", status);
pthread_mutex_unlock(&in->lock);
pthread_mutex_unlock(&in->dev->lock);
return status;
}
static int in_remove_audio_effect(const struct audio_stream *stream,
effect_handle_t effect)
{
struct tuna_stream_in *in = (struct tuna_stream_in *)stream;
int i;
int status = -EINVAL;
effect_descriptor_t desc;
pthread_mutex_lock(&in->dev->lock);
pthread_mutex_lock(&in->lock);
if (in->num_preprocessors <= 0) {
status = -ENOSYS;
goto exit;
}
for (i = 0; i < in->num_preprocessors; i++) {
if (status == 0) { /* status == 0 means an effect was removed from a previous slot */
in->preprocessors[i - 1].effect_itfe = in->preprocessors[i].effect_itfe;
in->preprocessors[i - 1].channel_configs = in->preprocessors[i].channel_configs;
in->preprocessors[i - 1].num_channel_configs = in->preprocessors[i].num_channel_configs;
ALOGV("in_remove_audio_effect moving fx from %d to %d", i, i - 1);
continue;
}
if (in->preprocessors[i].effect_itfe == effect) {
ALOGV("in_remove_audio_effect found fx at index %d", i);
free(in->preprocessors[i].channel_configs);
status = 0;
}
}
if (status != 0)
goto exit;
in->num_preprocessors--;
/* if we remove one effect, at least the last preproc should be reset */
in->preprocessors[in->num_preprocessors].num_channel_configs = 0;
in->preprocessors[in->num_preprocessors].effect_itfe = NULL;
in->preprocessors[in->num_preprocessors].channel_configs = NULL;
/* check compatibility between main channel supported and possible auxiliary channels */
in_update_aux_channels(in, NULL);
status = (*effect)->get_descriptor(effect, &desc);
if (status != 0)
goto exit;
ALOGV("in_remove_audio_effect(), effect type: %08x", desc.type.timeLow);
if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) {
in->need_echo_reference = false;
do_input_standby(in);
}
exit:
ALOGW_IF(status != 0, "in_remove_audio_effect() error %d", status);
pthread_mutex_unlock(&in->lock);
pthread_mutex_unlock(&in->dev->lock);
return status;
}
static 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)
{
struct tuna_audio_device *ladev = (struct tuna_audio_device *)dev;
struct tuna_stream_out *out;
int ret;
int output_type;
*stream_out = NULL;
out = (struct tuna_stream_out *)calloc(1, sizeof(struct tuna_stream_out));
if (!out)
return -ENOMEM;
if (flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
ALOGV("adev_open_output_stream() deep buffer");
if (ladev->outputs[OUTPUT_DEEP_BUF] != NULL)
return -ENOSYS;
output_type = OUTPUT_DEEP_BUF;
out->stream.common.get_buffer_size = out_get_buffer_size_deep_buffer;
out->stream.get_latency = out_get_latency_deep_buffer;
out->stream.write = out_write_deep_buffer;
} else {
ALOGV("adev_open_output_stream() normal buffer");
if (ladev->outputs[OUTPUT_LOW_LATENCY] != NULL)
return -ENOSYS;
output_type = OUTPUT_LOW_LATENCY;
out->stream.common.get_buffer_size = out_get_buffer_size_low_latency;
out->stream.get_latency = out_get_latency_low_latency;
out->stream.write = out_write_low_latency;
}
ret = create_resampler(DEFAULT_OUT_SAMPLING_RATE,
MM_FULL_POWER_SAMPLING_RATE,
2,
RESAMPLER_QUALITY_DEFAULT,
NULL,
&out->resampler);
if (ret != 0)
goto err_open;
out->stream.common.get_sample_rate = out_get_sample_rate;
out->stream.common.set_sample_rate = out_set_sample_rate;
out->stream.common.get_channels = out_get_channels;
out->stream.common.get_format = out_get_format;
out->stream.common.set_format = out_set_format;
out->stream.common.standby = out_standby;
out->stream.common.dump = out_dump;
out->stream.common.set_parameters = out_set_parameters;
out->stream.common.get_parameters = out_get_parameters;
out->stream.common.add_audio_effect = out_add_audio_effect;
out->stream.common.remove_audio_effect = out_remove_audio_effect;
out->stream.set_volume = out_set_volume;
out->stream.get_render_position = out_get_render_position;
out->dev = ladev;
out->standby = 1;
/* FIXME: when we support multiple output devices, we will want to
* do the following:
* adev->devices &= ~AUDIO_DEVICE_OUT_ALL;
* adev->devices |= out->device;
* select_output_device(adev);
* This is because out_set_parameters() with a route is not
* guaranteed to be called after an output stream is opened. */
config->format = out_get_format(&out->stream.common);
config->channel_mask = out_get_channels(&out->stream.common);
config->sample_rate = out_get_sample_rate(&out->stream.common);
*stream_out = &out->stream;
ladev->outputs[output_type] = out;
return 0;
err_open:
free(out);
return ret;
}
static void adev_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
struct tuna_audio_device *ladev = (struct tuna_audio_device *)dev;
struct tuna_stream_out *out = (struct tuna_stream_out *)stream;
int i;
out_standby(&stream->common);
for (i = 0; i < OUTPUT_TOTAL; i++) {
if (ladev->outputs[i] == out) {
ladev->outputs[i] = NULL;
break;
}
}
if (out->buffer)
free(out->buffer);
if (out->resampler)
release_resampler(out->resampler);
free(stream);
}
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
struct tuna_audio_device *adev = (struct tuna_audio_device *)dev;
struct str_parms *parms;
char *str;
char value[32];
int ret;
parms = str_parms_create_str(kvpairs);
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_TTY_MODE, value, sizeof(value));
if (ret >= 0) {
int tty_mode;
if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_OFF) == 0)
tty_mode = TTY_MODE_OFF;
else if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_VCO) == 0)
tty_mode = TTY_MODE_VCO;
else if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_HCO) == 0)
tty_mode = TTY_MODE_HCO;
else if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_FULL) == 0)
tty_mode = TTY_MODE_FULL;
else
return -EINVAL;
pthread_mutex_lock(&adev->lock);
if (tty_mode != adev->tty_mode) {
adev->tty_mode = tty_mode;
if (adev->mode == AUDIO_MODE_IN_CALL)
select_output_device(adev);
}
pthread_mutex_unlock(&adev->lock);
}
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value));
if (ret >= 0) {
if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
adev->bluetooth_nrec = true;
else
adev->bluetooth_nrec = false;
}
str_parms_destroy(parms);
return ret;
}
static char * adev_get_parameters(const struct audio_hw_device *dev,
const char *keys)
{
return strdup("");
}
static int adev_init_check(const struct audio_hw_device *dev)
{
return 0;
}
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
struct tuna_audio_device *adev = (struct tuna_audio_device *)dev;
adev->voice_volume = volume;
if (adev->mode == AUDIO_MODE_IN_CALL)
ril_set_call_volume(&adev->ril, SOUND_TYPE_VOICE, volume);
return 0;
}
static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
{
return -ENOSYS;
}
static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
struct tuna_audio_device *adev = (struct tuna_audio_device *)dev;
pthread_mutex_lock(&adev->lock);
if (adev->mode != mode) {
adev->mode = mode;
select_mode(adev);
}
pthread_mutex_unlock(&adev->lock);
return 0;
}
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
struct tuna_audio_device *adev = (struct tuna_audio_device *)dev;
adev->mic_mute = state;
return 0;
}
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
struct tuna_audio_device *adev = (struct tuna_audio_device *)dev;
*state = adev->mic_mute;
return 0;
}
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
const struct audio_config *config)
{
size_t size;
int channel_count = popcount(config->channel_mask);
if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
return 0;
return get_input_buffer_size(config->sample_rate, config->format, channel_count);
}
static int adev_open_input_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
struct audio_config *config,
struct audio_stream_in **stream_in)
{
struct tuna_audio_device *ladev = (struct tuna_audio_device *)dev;
struct tuna_stream_in *in;
int ret;
int channel_count = popcount(config->channel_mask);
*stream_in = NULL;
if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
return -EINVAL;
in = (struct tuna_stream_in *)calloc(1, sizeof(struct tuna_stream_in));
if (!in)
return -ENOMEM;
in->stream.common.get_sample_rate = in_get_sample_rate;
in->stream.common.set_sample_rate = in_set_sample_rate;
in->stream.common.get_buffer_size = in_get_buffer_size;
in->stream.common.get_channels = in_get_channels;
in->stream.common.get_format = in_get_format;
in->stream.common.set_format = in_set_format;
in->stream.common.standby = in_standby;
in->stream.common.dump = in_dump;
in->stream.common.set_parameters = in_set_parameters;
in->stream.common.get_parameters = in_get_parameters;
in->stream.common.add_audio_effect = in_add_audio_effect;
in->stream.common.remove_audio_effect = in_remove_audio_effect;
in->stream.set_gain = in_set_gain;
in->stream.read = in_read;
in->stream.get_input_frames_lost = in_get_input_frames_lost;
in->requested_rate = config->sample_rate;
memcpy(&in->config, &pcm_config_mm_ul, sizeof(pcm_config_mm_ul));
in->config.channels = channel_count;
in->main_channels = config->channel_mask;
/* initialisation of preprocessor structure array is implicit with the calloc.
* same for in->aux_channels and in->aux_channels_changed */
if (in->requested_rate != in->config.rate) {
in->buf_provider.get_next_buffer = get_next_buffer;
in->buf_provider.release_buffer = release_buffer;
ret = create_resampler(in->config.rate,
in->requested_rate,
in->config.channels,
RESAMPLER_QUALITY_DEFAULT,
&in->buf_provider,
&in->resampler);
if (ret != 0) {
ret = -EINVAL;
goto err;
}
}
in->dev = ladev;
in->standby = 1;
in->device = devices;
*stream_in = &in->stream;
return 0;
err:
if (in->resampler)
release_resampler(in->resampler);
free(in);
return ret;
}
static void adev_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *stream)
{
struct tuna_stream_in *in = (struct tuna_stream_in *)stream;
int i;
in_standby(&stream->common);
for (i = 0; i < in->num_preprocessors; i++) {
free(in->preprocessors[i].channel_configs);
}
free(in->read_buf);
if (in->resampler) {
release_resampler(in->resampler);
}
if (in->proc_buf_in)
free(in->proc_buf_in);
if (in->proc_buf_out)
free(in->proc_buf_out);
if (in->ref_buf)
free(in->ref_buf);
free(stream);
return;
}
static int adev_dump(const audio_hw_device_t *device, int fd)
{
return 0;
}
static int adev_close(hw_device_t *device)
{
struct tuna_audio_device *adev = (struct tuna_audio_device *)device;
/* RIL */
ril_close(&adev->ril);
mixer_close(adev->mixer);
free(device);
return 0;
}
static uint32_t adev_get_supported_devices(const struct audio_hw_device *dev)
{
return (/* OUT */
AUDIO_DEVICE_OUT_EARPIECE |
AUDIO_DEVICE_OUT_SPEAKER |
AUDIO_DEVICE_OUT_WIRED_HEADSET |
AUDIO_DEVICE_OUT_WIRED_HEADPHONE |
AUDIO_DEVICE_OUT_AUX_DIGITAL |
AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET |
AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET |
AUDIO_DEVICE_OUT_ALL_SCO |
AUDIO_DEVICE_OUT_DEFAULT |
/* IN */
AUDIO_DEVICE_IN_COMMUNICATION |
AUDIO_DEVICE_IN_AMBIENT |
AUDIO_DEVICE_IN_BUILTIN_MIC |
AUDIO_DEVICE_IN_WIRED_HEADSET |
AUDIO_DEVICE_IN_AUX_DIGITAL |
AUDIO_DEVICE_IN_BACK_MIC |
AUDIO_DEVICE_IN_ALL_SCO |
AUDIO_DEVICE_IN_DEFAULT);
}
static int adev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
struct tuna_audio_device *adev;
int ret;
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
adev = calloc(1, sizeof(struct tuna_audio_device));
if (!adev)
return -ENOMEM;
adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_1_0;
adev->hw_device.common.module = (struct hw_module_t *) module;
adev->hw_device.common.close = adev_close;
adev->hw_device.get_supported_devices = adev_get_supported_devices;
adev->hw_device.init_check = adev_init_check;
adev->hw_device.set_voice_volume = adev_set_voice_volume;
adev->hw_device.set_master_volume = adev_set_master_volume;
adev->hw_device.set_mode = adev_set_mode;
adev->hw_device.set_mic_mute = adev_set_mic_mute;
adev->hw_device.get_mic_mute = adev_get_mic_mute;
adev->hw_device.set_parameters = adev_set_parameters;
adev->hw_device.get_parameters = adev_get_parameters;
adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
adev->hw_device.open_output_stream = adev_open_output_stream;
adev->hw_device.close_output_stream = adev_close_output_stream;
adev->hw_device.open_input_stream = adev_open_input_stream;
adev->hw_device.close_input_stream = adev_close_input_stream;
adev->hw_device.dump = adev_dump;
adev->mixer = mixer_open(0);
if (!adev->mixer) {
free(adev);
ALOGE("Unable to open the mixer, aborting.");
return -EINVAL;
}
adev->mixer_ctls.dl1_eq = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL1_EQUALIZER);
adev->mixer_ctls.mm_dl2_volume = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL2_MEDIA_PLAYBACK_VOLUME);
adev->mixer_ctls.vx_dl2_volume = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL2_VOICE_PLAYBACK_VOLUME);
adev->mixer_ctls.mm_dl1 = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL1_MIXER_MULTIMEDIA);
adev->mixer_ctls.vx_dl1 = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL1_MIXER_VOICE);
adev->mixer_ctls.tones_dl1 = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL1_MIXER_TONES);
adev->mixer_ctls.mm_dl2 = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL2_MIXER_MULTIMEDIA);
adev->mixer_ctls.vx_dl2 = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL2_MIXER_VOICE);
adev->mixer_ctls.tones_dl2 = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL2_MIXER_TONES);
adev->mixer_ctls.dl2_mono = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL2_MONO_MIXER);
adev->mixer_ctls.dl1_headset = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL1_PDM_SWITCH);
adev->mixer_ctls.dl1_bt = mixer_get_ctl_by_name(adev->mixer,
MIXER_DL1_BT_VX_SWITCH);
adev->mixer_ctls.earpiece_enable = mixer_get_ctl_by_name(adev->mixer,
MIXER_EARPHONE_ENABLE_SWITCH);
adev->mixer_ctls.left_capture = mixer_get_ctl_by_name(adev->mixer,
MIXER_ANALOG_LEFT_CAPTURE_ROUTE);
adev->mixer_ctls.right_capture = mixer_get_ctl_by_name(adev->mixer,
MIXER_ANALOG_RIGHT_CAPTURE_ROUTE);
adev->mixer_ctls.amic_ul_volume = mixer_get_ctl_by_name(adev->mixer,
MIXER_AMIC_UL_VOLUME);
adev->mixer_ctls.voice_ul_volume = mixer_get_ctl_by_name(adev->mixer,
MIXER_AUDUL_VOICE_UL_VOLUME);
adev->mixer_ctls.sidetone_capture = mixer_get_ctl_by_name(adev->mixer,
MIXER_SIDETONE_MIXER_CAPTURE);
adev->mixer_ctls.headset_volume = mixer_get_ctl_by_name(adev->mixer,
MIXER_HEADSET_PLAYBACK_VOLUME);
adev->mixer_ctls.speaker_volume = mixer_get_ctl_by_name(adev->mixer,
MIXER_HANDSFREE_PLAYBACK_VOLUME);
adev->mixer_ctls.earpiece_volume = mixer_get_ctl_by_name(adev->mixer,
MIXER_EARPHONE_PLAYBACK_VOLUME);
if (!adev->mixer_ctls.dl1_eq || !adev->mixer_ctls.vx_dl2_volume ||
!adev->mixer_ctls.mm_dl2_volume || !adev->mixer_ctls.mm_dl1 ||
!adev->mixer_ctls.vx_dl1 || !adev->mixer_ctls.tones_dl1 ||
!adev->mixer_ctls.mm_dl2 || !adev->mixer_ctls.vx_dl2 ||
!adev->mixer_ctls.tones_dl2 ||!adev->mixer_ctls.dl2_mono ||
!adev->mixer_ctls.dl1_headset || !adev->mixer_ctls.dl1_bt ||
!adev->mixer_ctls.earpiece_enable || !adev->mixer_ctls.left_capture ||
!adev->mixer_ctls.right_capture || !adev->mixer_ctls.amic_ul_volume ||
!adev->mixer_ctls.voice_ul_volume || !adev->mixer_ctls.sidetone_capture ||
!adev->mixer_ctls.headset_volume || !adev->mixer_ctls.speaker_volume ||
!adev->mixer_ctls.earpiece_volume) {
mixer_close(adev->mixer);
free(adev);
ALOGE("Unable to locate all mixer controls, aborting.");
return -EINVAL;
}
/* Set the default route before the PCM stream is opened */
pthread_mutex_lock(&adev->lock);
set_route_by_array(adev->mixer, defaults, 1);
adev->mode = AUDIO_MODE_NORMAL;
adev->devices = AUDIO_DEVICE_OUT_SPEAKER | AUDIO_DEVICE_IN_BUILTIN_MIC;
select_output_device(adev);
adev->pcm_modem_dl = NULL;
adev->pcm_modem_ul = NULL;
adev->voice_volume = 1.0f;
adev->tty_mode = TTY_MODE_OFF;
adev->device_is_toro = is_device_toro();
adev->bluetooth_nrec = true;
adev->wb_amr = 0;
/* RIL */
ril_open(&adev->ril);
pthread_mutex_unlock(&adev->lock);
/* register callback for wideband AMR setting */
ril_register_set_wb_amr_callback(audio_set_wb_amr_callback, (void *)adev);
*device = &adev->hw_device.common;
return 0;
}
static struct hw_module_methods_t hal_module_methods = {
.open = adev_open,
};
struct audio_module HAL_MODULE_INFO_SYM = {
.common = {
.tag = HARDWARE_MODULE_TAG,
.module_api_version = AUDIO_MODULE_API_VERSION_0_1,
.hal_api_version = HARDWARE_HAL_API_VERSION,
.id = AUDIO_HARDWARE_MODULE_ID,
.name = "Tuna audio HW HAL",
.author = "The Android Open Source Project",
.methods = &hal_module_methods,
},
};