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v811_spc009/hardware/qcom/audio/hal/audio_hw.c

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/*
* Copyright (C) 2013-2016 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 ATRACE_TAG ATRACE_TAG_AUDIO
/*#define LOG_NDEBUG 0*/
/*#define VERY_VERY_VERBOSE_LOGGING*/
#ifdef VERY_VERY_VERBOSE_LOGGING
#define ALOGVV ALOGV
#else
#define ALOGVV(a...) do { } while(0)
#endif
#include <errno.h>
#include <pthread.h>
#include <stdint.h>
#include <sys/time.h>
#include <stdlib.h>
#include <math.h>
#include <dlfcn.h>
#include <sys/resource.h>
#include <sys/prctl.h>
#include <limits.h>
#include <log/log.h>
#include <cutils/trace.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <cutils/atomic.h>
#include <utils/Timers.h> // systemTime
#include <hardware/audio_effect.h>
#include <hardware/audio_alsaops.h>
#include <processgroup/sched_policy.h>
#include <system/thread_defs.h>
#include <tinyalsa/asoundlib.h>
#include <audio_effects/effect_aec.h>
#include <audio_effects/effect_ns.h>
#include <audio_utils/clock.h>
#include "audio_hw.h"
#include "audio_extn.h"
#include "audio_perf.h"
#include "platform_api.h"
#include <platform.h>
#include "voice_extn.h"
#include "sound/compress_params.h"
#include "audio_extn/tfa_98xx.h"
#include "audio_extn/maxxaudio.h"
#include "audio_extn/audiozoom.h"
/* COMPRESS_OFFLOAD_FRAGMENT_SIZE must be more than 8KB and a multiple of 32KB if more than 32KB.
* COMPRESS_OFFLOAD_FRAGMENT_SIZE * COMPRESS_OFFLOAD_NUM_FRAGMENTS must be less than 8MB. */
#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (256 * 1024)
// 2 buffers causes problems with high bitrate files
#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 3
/* ToDo: Check and update a proper value in msec */
#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96
/* treat as unsigned Q1.13 */
#define APP_TYPE_GAIN_DEFAULT 0x2000
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
#define PCM_PLAYBACK_VOLUME_MAX 0x2000
#define INVALID_OUT_VOLUME -1
/* treat as unsigned Q1.13 */
#define VOIP_PLAYBACK_VOLUME_MAX 0x2000
#define RECORD_GAIN_MIN 0.0f
#define RECORD_GAIN_MAX 1.0f
#define RECORD_VOLUME_CTL_MAX 0x2000
#define PROXY_OPEN_RETRY_COUNT 100
#define PROXY_OPEN_WAIT_TIME 20
#define MIN_CHANNEL_COUNT 1
#define DEFAULT_CHANNEL_COUNT 2
#ifndef MAX_TARGET_SPECIFIC_CHANNEL_CNT
#define MAX_CHANNEL_COUNT 1
#else
#define MAX_CHANNEL_COUNT atoi(XSTR(MAX_TARGET_SPECIFIC_CHANNEL_CNT))
#define XSTR(x) STR(x)
#define STR(x) #x
#endif
#define MAX_HIFI_CHANNEL_COUNT 8
#define ULL_PERIOD_SIZE (DEFAULT_OUTPUT_SAMPLING_RATE/1000)
static unsigned int configured_low_latency_capture_period_size =
LOW_LATENCY_CAPTURE_PERIOD_SIZE;
#define MMAP_PERIOD_SIZE (DEFAULT_OUTPUT_SAMPLING_RATE/1000)
#define MMAP_PERIOD_COUNT_MIN 32
#define MMAP_PERIOD_COUNT_MAX 512
#define MMAP_PERIOD_COUNT_DEFAULT (MMAP_PERIOD_COUNT_MAX)
#define MMAP_MIN_SIZE_FRAMES_MAX 64 * 1024
/* This constant enables extended precision handling.
* TODO The flag is off until more testing is done.
*/
static const bool k_enable_extended_precision = false;
struct pcm_config pcm_config_deep_buffer = {
.channels = DEFAULT_CHANNEL_COUNT,
.rate = DEFAULT_OUTPUT_SAMPLING_RATE,
.period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE,
.period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
.start_threshold = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4,
.stop_threshold = INT_MAX,
.avail_min = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4,
};
struct pcm_config pcm_config_low_latency = {
.channels = DEFAULT_CHANNEL_COUNT,
.rate = DEFAULT_OUTPUT_SAMPLING_RATE,
.period_size = LOW_LATENCY_OUTPUT_PERIOD_SIZE,
.period_count = LOW_LATENCY_OUTPUT_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
.start_threshold = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
.stop_threshold = INT_MAX,
.avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
};
struct pcm_config pcm_config_haptics_audio = {
.channels = 1,
.rate = DEFAULT_OUTPUT_SAMPLING_RATE,
.period_size = LOW_LATENCY_OUTPUT_PERIOD_SIZE,
.period_count = LOW_LATENCY_OUTPUT_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
.start_threshold = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
.stop_threshold = INT_MAX,
.avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
};
struct pcm_config pcm_config_haptics = {
.channels = 1,
.rate = DEFAULT_OUTPUT_SAMPLING_RATE,
.period_count = 2,
.format = PCM_FORMAT_S16_LE,
.stop_threshold = INT_MAX,
.avail_min = 0,
};
static int af_period_multiplier = 4;
struct pcm_config pcm_config_rt = {
.channels = DEFAULT_CHANNEL_COUNT,
.rate = DEFAULT_OUTPUT_SAMPLING_RATE,
.period_size = ULL_PERIOD_SIZE, //1 ms
.period_count = 512, //=> buffer size is 512ms
.format = PCM_FORMAT_S16_LE,
.start_threshold = ULL_PERIOD_SIZE*8, //8ms
.stop_threshold = INT_MAX,
.silence_threshold = 0,
.silence_size = 0,
.avail_min = ULL_PERIOD_SIZE, //1 ms
};
struct pcm_config pcm_config_hdmi_multi = {
.channels = HDMI_MULTI_DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */
.rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */
.period_size = HDMI_MULTI_PERIOD_SIZE,
.period_count = HDMI_MULTI_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
.start_threshold = 0,
.stop_threshold = INT_MAX,
.avail_min = 0,
};
struct pcm_config pcm_config_mmap_playback = {
.channels = DEFAULT_CHANNEL_COUNT,
.rate = DEFAULT_OUTPUT_SAMPLING_RATE,
.period_size = MMAP_PERIOD_SIZE,
.period_count = MMAP_PERIOD_COUNT_DEFAULT,
.format = PCM_FORMAT_S16_LE,
.start_threshold = MMAP_PERIOD_SIZE*8,
.stop_threshold = INT32_MAX,
.silence_threshold = 0,
.silence_size = 0,
.avail_min = MMAP_PERIOD_SIZE, //1 ms
};
struct pcm_config pcm_config_hifi = {
.channels = DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */
.rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */
.period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE, /* change #define */
.period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT,
.format = PCM_FORMAT_S24_3LE,
.start_threshold = 0,
.stop_threshold = INT_MAX,
.avail_min = 0,
};
struct pcm_config pcm_config_audio_capture = {
.channels = DEFAULT_CHANNEL_COUNT,
.period_count = AUDIO_CAPTURE_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
.stop_threshold = INT_MAX,
.avail_min = 0,
};
struct pcm_config pcm_config_audio_capture_rt = {
.channels = DEFAULT_CHANNEL_COUNT,
.rate = DEFAULT_OUTPUT_SAMPLING_RATE,
.period_size = ULL_PERIOD_SIZE,
.period_count = 512,
.format = PCM_FORMAT_S16_LE,
.start_threshold = 0,
.stop_threshold = INT_MAX,
.silence_threshold = 0,
.silence_size = 0,
.avail_min = ULL_PERIOD_SIZE, //1 ms
};
struct pcm_config pcm_config_mmap_capture = {
.channels = DEFAULT_CHANNEL_COUNT,
.rate = DEFAULT_OUTPUT_SAMPLING_RATE,
.period_size = MMAP_PERIOD_SIZE,
.period_count = MMAP_PERIOD_COUNT_DEFAULT,
.format = PCM_FORMAT_S16_LE,
.start_threshold = 0,
.stop_threshold = INT_MAX,
.silence_threshold = 0,
.silence_size = 0,
.avail_min = MMAP_PERIOD_SIZE, //1 ms
};
struct pcm_config pcm_config_voip = {
.channels = 1,
.period_count = 2,
.format = PCM_FORMAT_S16_LE,
.stop_threshold = INT_MAX,
.avail_min = 0,
};
#define AFE_PROXY_CHANNEL_COUNT 2
#define AFE_PROXY_SAMPLING_RATE 48000
#define AFE_PROXY_PLAYBACK_PERIOD_SIZE 256
#define AFE_PROXY_PLAYBACK_PERIOD_COUNT 4
struct pcm_config pcm_config_afe_proxy_playback = {
.channels = AFE_PROXY_CHANNEL_COUNT,
.rate = AFE_PROXY_SAMPLING_RATE,
.period_size = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
.period_count = AFE_PROXY_PLAYBACK_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
.start_threshold = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
.stop_threshold = INT_MAX,
.avail_min = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
};
#define AFE_PROXY_RECORD_PERIOD_SIZE 256
#define AFE_PROXY_RECORD_PERIOD_COUNT 4
struct pcm_config pcm_config_afe_proxy_record = {
.channels = AFE_PROXY_CHANNEL_COUNT,
.rate = AFE_PROXY_SAMPLING_RATE,
.period_size = AFE_PROXY_RECORD_PERIOD_SIZE,
.period_count = AFE_PROXY_RECORD_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
.start_threshold = AFE_PROXY_RECORD_PERIOD_SIZE,
.stop_threshold = AFE_PROXY_RECORD_PERIOD_SIZE * AFE_PROXY_RECORD_PERIOD_COUNT,
.avail_min = AFE_PROXY_RECORD_PERIOD_SIZE,
};
const char * const use_case_table[AUDIO_USECASE_MAX] = {
[USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = "deep-buffer-playback",
[USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = "low-latency-playback",
[USECASE_AUDIO_PLAYBACK_WITH_HAPTICS] = "audio-with-haptics-playback",
[USECASE_AUDIO_PLAYBACK_HIFI] = "hifi-playback",
[USECASE_AUDIO_PLAYBACK_OFFLOAD] = "compress-offload-playback",
[USECASE_AUDIO_PLAYBACK_TTS] = "audio-tts-playback",
[USECASE_AUDIO_PLAYBACK_ULL] = "audio-ull-playback",
[USECASE_AUDIO_PLAYBACK_MMAP] = "mmap-playback",
[USECASE_AUDIO_RECORD] = "audio-record",
[USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record",
[USECASE_AUDIO_RECORD_MMAP] = "mmap-record",
[USECASE_AUDIO_RECORD_HIFI] = "hifi-record",
[USECASE_AUDIO_HFP_SCO] = "hfp-sco",
[USECASE_AUDIO_HFP_SCO_WB] = "hfp-sco-wb",
[USECASE_VOICE_CALL] = "voice-call",
[USECASE_VOICE2_CALL] = "voice2-call",
[USECASE_VOLTE_CALL] = "volte-call",
[USECASE_QCHAT_CALL] = "qchat-call",
[USECASE_VOWLAN_CALL] = "vowlan-call",
[USECASE_VOICEMMODE1_CALL] = "voicemmode1-call",
[USECASE_VOICEMMODE2_CALL] = "voicemmode2-call",
[USECASE_AUDIO_SPKR_CALIB_RX] = "spkr-rx-calib",
[USECASE_AUDIO_SPKR_CALIB_TX] = "spkr-vi-record",
[USECASE_AUDIO_PLAYBACK_AFE_PROXY] = "afe-proxy-playback",
[USECASE_AUDIO_RECORD_AFE_PROXY] = "afe-proxy-record",
[USECASE_INCALL_REC_UPLINK] = "incall-rec-uplink",
[USECASE_INCALL_REC_DOWNLINK] = "incall-rec-downlink",
[USECASE_INCALL_REC_UPLINK_AND_DOWNLINK] = "incall-rec-uplink-and-downlink",
[USECASE_AUDIO_PLAYBACK_VOIP] = "audio-playback-voip",
[USECASE_AUDIO_RECORD_VOIP] = "audio-record-voip",
[USECASE_INCALL_MUSIC_UPLINK] = "incall-music-uplink",
[USECASE_INCALL_MUSIC_UPLINK2] = "incall-music-uplink2",
[USECASE_AUDIO_A2DP_ABR_FEEDBACK] = "a2dp-abr-feedback",
};
#define STRING_TO_ENUM(string) { #string, string }
struct string_to_enum {
const char *name;
uint32_t value;
};
static const struct string_to_enum channels_name_to_enum_table[] = {
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
STRING_TO_ENUM(AUDIO_CHANNEL_IN_MONO),
STRING_TO_ENUM(AUDIO_CHANNEL_IN_STEREO),
STRING_TO_ENUM(AUDIO_CHANNEL_IN_FRONT_BACK),
STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_1),
STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_2),
STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_3),
STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_4),
STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_5),
STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_6),
STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_7),
STRING_TO_ENUM(AUDIO_CHANNEL_INDEX_MASK_8),
};
struct in_effect_list {
struct listnode list;
effect_handle_t handle;
};
static int set_voice_volume_l(struct audio_device *adev, float volume);
static struct audio_device *adev = NULL;
static pthread_mutex_t adev_init_lock = PTHREAD_MUTEX_INITIALIZER;
static unsigned int audio_device_ref_count;
//cache last MBDRC cal step level
static int last_known_cal_step = -1 ;
static int check_a2dp_restore_l(struct audio_device *adev, struct stream_out *out, bool restore);
static int out_set_compr_volume(struct audio_stream_out *stream, float left, float right);
static int out_set_pcm_volume(struct audio_stream_out *stream, float left, float right);
static int in_set_microphone_direction(const struct audio_stream_in *stream,
audio_microphone_direction_t dir);
static int in_set_microphone_field_dimension(const struct audio_stream_in *stream, float zoom);
static bool may_use_noirq_mode(struct audio_device *adev, audio_usecase_t uc_id,
int flags __unused)
{
int dir = 0;
switch (uc_id) {
case USECASE_AUDIO_RECORD_LOW_LATENCY:
dir = 1;
case USECASE_AUDIO_PLAYBACK_ULL:
break;
default:
return false;
}
int dev_id = platform_get_pcm_device_id(uc_id, dir == 0 ?
PCM_PLAYBACK : PCM_CAPTURE);
if (adev->adm_is_noirq_avail)
return adev->adm_is_noirq_avail(adev->adm_data,
adev->snd_card, dev_id, dir);
return false;
}
static void register_out_stream(struct stream_out *out)
{
struct audio_device *adev = out->dev;
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD)
return;
if (!adev->adm_register_output_stream)
return;
adev->adm_register_output_stream(adev->adm_data,
out->handle,
out->flags);
if (!adev->adm_set_config)
return;
if (out->realtime) {
adev->adm_set_config(adev->adm_data,
out->handle,
out->pcm, &out->config);
}
}
static void register_in_stream(struct stream_in *in)
{
struct audio_device *adev = in->dev;
if (!adev->adm_register_input_stream)
return;
adev->adm_register_input_stream(adev->adm_data,
in->capture_handle,
in->flags);
if (!adev->adm_set_config)
return;
if (in->realtime) {
adev->adm_set_config(adev->adm_data,
in->capture_handle,
in->pcm,
&in->config);
}
}
static void request_out_focus(struct stream_out *out, long ns)
{
struct audio_device *adev = out->dev;
if (adev->adm_request_focus_v2) {
adev->adm_request_focus_v2(adev->adm_data, out->handle, ns);
} else if (adev->adm_request_focus) {
adev->adm_request_focus(adev->adm_data, out->handle);
}
}
static void request_in_focus(struct stream_in *in, long ns)
{
struct audio_device *adev = in->dev;
if (adev->adm_request_focus_v2) {
adev->adm_request_focus_v2(adev->adm_data, in->capture_handle, ns);
} else if (adev->adm_request_focus) {
adev->adm_request_focus(adev->adm_data, in->capture_handle);
}
}
static void release_out_focus(struct stream_out *out, long ns __unused)
{
struct audio_device *adev = out->dev;
if (adev->adm_abandon_focus)
adev->adm_abandon_focus(adev->adm_data, out->handle);
}
static void release_in_focus(struct stream_in *in, long ns __unused)
{
struct audio_device *adev = in->dev;
if (adev->adm_abandon_focus)
adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
}
static int parse_snd_card_status(struct str_parms * parms, int * card,
card_status_t * status)
{
char value[32]={0};
char state[32]={0};
int ret = str_parms_get_str(parms, "SND_CARD_STATUS", value, sizeof(value));
if (ret < 0)
return -1;
// sscanf should be okay as value is of max length 32.
// same as sizeof state.
if (sscanf(value, "%d,%s", card, state) < 2)
return -1;
*status = !strcmp(state, "ONLINE") ? CARD_STATUS_ONLINE :
CARD_STATUS_OFFLINE;
return 0;
}
// always call with adev lock held
void send_gain_dep_calibration_l() {
if (last_known_cal_step >= 0)
platform_send_gain_dep_cal(adev->platform, last_known_cal_step);
}
__attribute__ ((visibility ("default")))
bool audio_hw_send_gain_dep_calibration(int level) {
bool ret_val = false;
ALOGV("%s: enter ... ", __func__);
pthread_mutex_lock(&adev_init_lock);
if (adev != NULL && adev->platform != NULL) {
pthread_mutex_lock(&adev->lock);
last_known_cal_step = level;
send_gain_dep_calibration_l();
pthread_mutex_unlock(&adev->lock);
} else {
ALOGE("%s: %s is NULL", __func__, adev == NULL ? "adev" : "adev->platform");
}
pthread_mutex_unlock(&adev_init_lock);
ALOGV("%s: exit with ret_val %d ", __func__, ret_val);
return ret_val;
}
#ifdef MAXXAUDIO_QDSP_ENABLED
bool audio_hw_send_ma_parameter(int stream_type, float vol, bool active)
{
bool ret = false;
ALOGV("%s: enter ...", __func__);
pthread_mutex_lock(&adev_init_lock);
if (adev != NULL && adev->platform != NULL) {
pthread_mutex_lock(&adev->lock);
ret = audio_extn_ma_set_state(adev, stream_type, vol, active);
pthread_mutex_unlock(&adev->lock);
}
pthread_mutex_unlock(&adev_init_lock);
ALOGV("%s: exit with ret %d", __func__, ret);
return ret;
}
#else
#define audio_hw_send_ma_parameter(stream_type, vol, active) (0)
#endif
__attribute__ ((visibility ("default")))
int audio_hw_get_gain_level_mapping(struct amp_db_and_gain_table *mapping_tbl,
int table_size) {
int ret_val = 0;
ALOGV("%s: enter ... ", __func__);
pthread_mutex_lock(&adev_init_lock);
if (adev == NULL) {
ALOGW("%s: adev is NULL .... ", __func__);
goto done;
}
pthread_mutex_lock(&adev->lock);
ret_val = platform_get_gain_level_mapping(mapping_tbl, table_size);
pthread_mutex_unlock(&adev->lock);
done:
pthread_mutex_unlock(&adev_init_lock);
ALOGV("%s: exit ... ", __func__);
return ret_val;
}
static bool is_supported_format(audio_format_t format)
{
switch (format) {
case AUDIO_FORMAT_MP3:
case AUDIO_FORMAT_AAC_LC:
case AUDIO_FORMAT_AAC_HE_V1:
case AUDIO_FORMAT_AAC_HE_V2:
return true;
default:
break;
}
return false;
}
static bool is_supported_24bits_audiosource(audio_source_t source)
{
switch (source) {
case AUDIO_SOURCE_UNPROCESSED:
#ifdef ENABLED_24BITS_CAMCORDER
case AUDIO_SOURCE_CAMCORDER:
#endif
return true;
default:
break;
}
return false;
}
static inline bool is_mmap_usecase(audio_usecase_t uc_id)
{
return (uc_id == USECASE_AUDIO_RECORD_AFE_PROXY) ||
(uc_id == USECASE_AUDIO_PLAYBACK_AFE_PROXY);
}
static int get_snd_codec_id(audio_format_t format)
{
int id = 0;
switch (format & AUDIO_FORMAT_MAIN_MASK) {
case AUDIO_FORMAT_MP3:
id = SND_AUDIOCODEC_MP3;
break;
case AUDIO_FORMAT_AAC:
id = SND_AUDIOCODEC_AAC;
break;
default:
ALOGE("%s: Unsupported audio format", __func__);
}
return id;
}
static int audio_ssr_status(struct audio_device *adev)
{
int ret = 0;
struct mixer_ctl *ctl;
const char *mixer_ctl_name = "Audio SSR Status";
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
ret = mixer_ctl_get_value(ctl, 0);
ALOGD("%s: value: %d", __func__, ret);
return ret;
}
static void stream_app_type_cfg_init(struct stream_app_type_cfg *cfg)
{
cfg->gain[0] = cfg->gain[1] = APP_TYPE_GAIN_DEFAULT;
}
static bool is_btsco_device(snd_device_t out_snd_device, snd_device_t in_snd_device)
{
return out_snd_device == SND_DEVICE_OUT_BT_SCO ||
out_snd_device == SND_DEVICE_OUT_BT_SCO_WB ||
in_snd_device == SND_DEVICE_IN_BT_SCO_MIC_WB_NREC ||
in_snd_device == SND_DEVICE_IN_BT_SCO_MIC_WB ||
in_snd_device == SND_DEVICE_IN_BT_SCO_MIC_NREC ||
in_snd_device == SND_DEVICE_IN_BT_SCO_MIC;
}
static bool is_a2dp_device(snd_device_t out_snd_device)
{
return out_snd_device == SND_DEVICE_OUT_BT_A2DP;
}
int enable_audio_route(struct audio_device *adev,
struct audio_usecase *usecase)
{
snd_device_t snd_device;
char mixer_path[MIXER_PATH_MAX_LENGTH];
if (usecase == NULL)
return -EINVAL;
ALOGV("%s: enter: usecase(%d)", __func__, usecase->id);
audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_BUSY);
if (usecase->type == PCM_CAPTURE) {
struct stream_in *in = usecase->stream.in;
struct audio_usecase *uinfo;
snd_device = usecase->in_snd_device;
if (in) {
if (in->enable_aec || in->enable_ec_port) {
audio_devices_t out_device = AUDIO_DEVICE_OUT_SPEAKER;
struct listnode *node;
struct audio_usecase *voip_usecase = get_usecase_from_list(adev,
USECASE_AUDIO_PLAYBACK_VOIP);
if (voip_usecase) {
out_device = voip_usecase->stream.out->devices;
} else if (adev->primary_output &&
!adev->primary_output->standby) {
out_device = adev->primary_output->devices;
} else {
list_for_each(node, &adev->usecase_list) {
uinfo = node_to_item(node, struct audio_usecase, list);
if (uinfo->type != PCM_CAPTURE) {
out_device = uinfo->stream.out->devices;
break;
}
}
}
platform_set_echo_reference(adev, true, out_device);
in->ec_opened = true;
}
}
} else
snd_device = usecase->out_snd_device;
audio_extn_utils_send_app_type_cfg(adev, usecase);
audio_extn_ma_set_device(usecase);
audio_extn_utils_send_audio_calibration(adev, usecase);
// we shouldn't truncate mixer_path
ALOGW_IF(strlcpy(mixer_path, use_case_table[usecase->id], sizeof(mixer_path))
>= sizeof(mixer_path), "%s: truncation on mixer path", __func__);
// this also appends to mixer_path
platform_add_backend_name(adev->platform, mixer_path, snd_device);
ALOGD("%s: usecase(%d) apply and update mixer path: %s", __func__, usecase->id, mixer_path);
audio_route_apply_and_update_path(adev->audio_route, mixer_path);
ALOGV("%s: exit", __func__);
return 0;
}
int disable_audio_route(struct audio_device *adev,
struct audio_usecase *usecase)
{
snd_device_t snd_device;
char mixer_path[MIXER_PATH_MAX_LENGTH];
if (usecase == NULL)
return -EINVAL;
ALOGV("%s: enter: usecase(%d)", __func__, usecase->id);
if (usecase->type == PCM_CAPTURE)
snd_device = usecase->in_snd_device;
else
snd_device = usecase->out_snd_device;
// we shouldn't truncate mixer_path
ALOGW_IF(strlcpy(mixer_path, use_case_table[usecase->id], sizeof(mixer_path))
>= sizeof(mixer_path), "%s: truncation on mixer path", __func__);
// this also appends to mixer_path
platform_add_backend_name(adev->platform, mixer_path, snd_device);
ALOGD("%s: usecase(%d) reset and update mixer path: %s", __func__, usecase->id, mixer_path);
audio_route_reset_and_update_path(adev->audio_route, mixer_path);
if (usecase->type == PCM_CAPTURE) {
struct stream_in *in = usecase->stream.in;
if (in && in->ec_opened) {
platform_set_echo_reference(in->dev, false, AUDIO_DEVICE_NONE);
in->ec_opened = false;
}
}
audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_FREE);
ALOGV("%s: exit", __func__);
return 0;
}
int enable_snd_device(struct audio_device *adev,
snd_device_t snd_device)
{
int i, num_devices = 0;
snd_device_t new_snd_devices[2];
int ret_val = -EINVAL;
if (snd_device < SND_DEVICE_MIN ||
snd_device >= SND_DEVICE_MAX) {
ALOGE("%s: Invalid sound device %d", __func__, snd_device);
goto on_error;
}
platform_send_audio_calibration(adev->platform, snd_device);
if (adev->snd_dev_ref_cnt[snd_device] >= 1) {
ALOGV("%s: snd_device(%d: %s) is already active",
__func__, snd_device, platform_get_snd_device_name(snd_device));
goto on_success;
}
/* due to the possibility of calibration overwrite between listen
and audio, notify sound trigger hal before audio calibration is sent */
audio_extn_sound_trigger_update_device_status(snd_device,
ST_EVENT_SND_DEVICE_BUSY);
if (audio_extn_spkr_prot_is_enabled())
audio_extn_spkr_prot_calib_cancel(adev);
audio_extn_dsm_feedback_enable(adev, snd_device, true);
if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
snd_device == SND_DEVICE_OUT_SPEAKER_SAFE ||
snd_device == SND_DEVICE_OUT_SPEAKER_REVERSE ||
snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
audio_extn_spkr_prot_is_enabled()) {
if (platform_get_snd_device_acdb_id(snd_device) < 0) {
goto on_error;
}
if (audio_extn_spkr_prot_start_processing(snd_device)) {
ALOGE("%s: spkr_start_processing failed", __func__);
goto on_error;
}
} else if (platform_can_split_snd_device(snd_device,
&num_devices,
new_snd_devices) == 0) {
for (i = 0; i < num_devices; i++) {
enable_snd_device(adev, new_snd_devices[i]);
}
platform_set_speaker_gain_in_combo(adev, snd_device, true);
} else {
char device_name[DEVICE_NAME_MAX_SIZE] = {0};
if (platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0 ) {
ALOGE(" %s: Invalid sound device returned", __func__);
goto on_error;
}
ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
if (is_a2dp_device(snd_device)) {
if (audio_extn_a2dp_start_playback() < 0) {
ALOGE("%s: failed to configure A2DP control path", __func__);
goto on_error;
} else {
adev->a2dp_started = true;
}
}
audio_route_apply_and_update_path(adev->audio_route, device_name);
}
on_success:
adev->snd_dev_ref_cnt[snd_device]++;
ret_val = 0;
on_error:
return ret_val;
}
int disable_snd_device(struct audio_device *adev,
snd_device_t snd_device)
{
int i, num_devices = 0;
snd_device_t new_snd_devices[2];
if (snd_device < SND_DEVICE_MIN ||
snd_device >= SND_DEVICE_MAX) {
ALOGE("%s: Invalid sound device %d", __func__, snd_device);
return -EINVAL;
}
if (adev->snd_dev_ref_cnt[snd_device] <= 0) {
ALOGE("%s: device ref cnt is already 0", __func__);
return -EINVAL;
}
audio_extn_tfa_98xx_disable_speaker(snd_device);
adev->snd_dev_ref_cnt[snd_device]--;
if (adev->snd_dev_ref_cnt[snd_device] == 0) {
audio_extn_dsm_feedback_enable(adev, snd_device, false);
if (is_a2dp_device(snd_device)) {
audio_extn_a2dp_stop_playback();
adev->a2dp_started = false;
}
if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
snd_device == SND_DEVICE_OUT_SPEAKER_SAFE ||
snd_device == SND_DEVICE_OUT_SPEAKER_REVERSE ||
snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
audio_extn_spkr_prot_is_enabled()) {
audio_extn_spkr_prot_stop_processing(snd_device);
// FIXME b/65363602: bullhead is the only Nexus with audio_extn_spkr_prot_is_enabled()
// and does not use speaker swap. As this code causes a problem with device enable ref
// counting we remove it for now.
// when speaker device is disabled, reset swap.
// will be renabled on usecase start
// platform_set_swap_channels(adev, false);
} else if (platform_can_split_snd_device(snd_device,
&num_devices,
new_snd_devices) == 0) {
for (i = 0; i < num_devices; i++) {
disable_snd_device(adev, new_snd_devices[i]);
}
platform_set_speaker_gain_in_combo(adev, snd_device, false);
} else {
char device_name[DEVICE_NAME_MAX_SIZE] = {0};
if (platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0 ) {
ALOGE(" %s: Invalid sound device returned", __func__);
return -EINVAL;
}
ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
audio_route_reset_and_update_path(adev->audio_route, device_name);
}
audio_extn_sound_trigger_update_device_status(snd_device,
ST_EVENT_SND_DEVICE_FREE);
}
return 0;
}
#ifdef DYNAMIC_ECNS_ENABLED
static int send_effect_enable_disable_mixer_ctl(struct audio_device *adev,
struct stream_in *in,
struct audio_effect_config effect_config,
unsigned int param_value)
{
char mixer_ctl_name[] = "Audio Effect";
long set_values[6];
struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get mixer ctl - %s",
__func__, mixer_ctl_name);
return -EINVAL;
}
set_values[0] = 1; //0:Rx 1:Tx
set_values[1] = in->app_type_cfg.app_type;
set_values[2] = (long)effect_config.module_id;
set_values[3] = (long)effect_config.instance_id;
set_values[4] = (long)effect_config.param_id;
set_values[5] = param_value;
mixer_ctl_set_array(ctl, set_values, ARRAY_SIZE(set_values));
return 0;
}
static int update_effect_param_ecns(struct audio_usecase *usecase,
unsigned int module_id, int effect_type,
unsigned int *param_value)
{
int ret = 0;
struct audio_effect_config other_effect_config;
struct stream_in *in = NULL;
if (!usecase)
return -EINVAL;
in = usecase->stream.in;
/* Get the effect config data of the other effect */
ret = platform_get_effect_config_data(usecase->in_snd_device,
&other_effect_config,
effect_type == EFFECT_AEC ? EFFECT_NS : EFFECT_AEC);
if (ret < 0) {
ALOGE("%s Failed to get effect params %d", __func__, ret);
return ret;
}
if (module_id == other_effect_config.module_id) {
//Same module id for AEC/NS. Values need to be combined
if (((effect_type == EFFECT_AEC) && (in->enable_ns)) ||
((effect_type == EFFECT_NS) && (in->enable_aec)))
*param_value |= other_effect_config.param_value;
}
return ret;
}
static int enable_disable_effect(struct audio_device *adev, struct stream_in *in,
int effect_type, bool enable)
{
struct audio_effect_config effect_config;
struct audio_usecase *usecase = NULL;
int ret = 0;
unsigned int param_value = 0;
if (!in) {
ALOGE("%s: Invalid input stream", __func__);
return -EINVAL;
}
ALOGD("%s: effect_type:%d enable:%d", __func__, effect_type, enable);
usecase = get_usecase_from_list(adev, in->usecase);
ret = platform_get_effect_config_data(usecase->in_snd_device,
&effect_config, effect_type);
if (ret < 0) {
ALOGE("%s Failed to get module id %d", __func__, ret);
return ret;
}
ALOGV("%s: module %d app_type %d usecase->id:%d usecase->in_snd_device:%d",
__func__, effect_config.module_id, in->app_type_cfg.app_type,
usecase->id, usecase->in_snd_device);
if (enable)
param_value = effect_config.param_value;
/*Special handling for AEC & NS effects Param values need to be
updated if module ids are same*/
if ((effect_type == EFFECT_AEC) || (effect_type == EFFECT_NS)) {
ret = update_effect_param_ecns(usecase, effect_config.module_id,
effect_type, &param_value);
if (ret < 0)
return ret;
}
ret = send_effect_enable_disable_mixer_ctl(adev, in,
effect_config, param_value);
return ret;
}
static int check_and_enable_effect(struct audio_device *adev)
{
int ret = 0;
struct listnode *node;
struct stream_in *in = NULL;
list_for_each(node, &adev->usecase_list)
{
struct audio_usecase *usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_CAPTURE && usecase->stream.in != NULL) {
in = usecase->stream.in;
if (in->standby)
continue;
if (in->enable_aec) {
ret = enable_disable_effect(adev, in, EFFECT_AEC, true);
}
if (in->enable_ns &&
in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
ret = enable_disable_effect(adev, in, EFFECT_NS, true);
}
}
}
return ret;
}
#else
#define enable_disable_effect(w, x, y, z) -ENOSYS
#define check_and_enable_effect(x) -ENOSYS
#endif
/*
legend:
uc - existing usecase
new_uc - new usecase
d1, d11, d2 - SND_DEVICE enums
a1, a2 - corresponding ANDROID device enums
B, B1, B2 - backend strings
case 1
uc->dev d1 (a1) B1
new_uc->dev d1 (a1), d2 (a2) B1, B2
resolution: disable and enable uc->dev on d1
case 2
uc->dev d1 (a1) B1
new_uc->dev d11 (a1) B1
resolution: need to switch uc since d1 and d11 are related
(e.g. speaker and voice-speaker)
use ANDROID_DEVICE_OUT enums to match devices since SND_DEVICE enums may vary
case 3
uc->dev d1 (a1) B1
new_uc->dev d2 (a2) B2
resolution: no need to switch uc
case 4
uc->dev d1 (a1) B
new_uc->dev d2 (a2) B
resolution: disable enable uc-dev on d2 since backends match
we cannot enable two streams on two different devices if they
share the same backend. e.g. if offload is on speaker device using
QUAD_MI2S backend and a low-latency stream is started on voice-handset
using the same backend, offload must also be switched to voice-handset.
case 5
uc->dev d1 (a1) B
new_uc->dev d1 (a1), d2 (a2) B
resolution: disable enable uc-dev on d2 since backends match
we cannot enable two streams on two different devices if they
share the same backend.
case 6
uc->dev d1 a1 B1
new_uc->dev d2 a1 B2
resolution: no need to switch
case 7
uc->dev d1 (a1), d2 (a2) B1, B2
new_uc->dev d1 B1
resolution: no need to switch
*/
static snd_device_t derive_playback_snd_device(struct audio_usecase *uc,
struct audio_usecase *new_uc,
snd_device_t new_snd_device)
{
audio_devices_t a1 = uc->stream.out->devices;
audio_devices_t a2 = new_uc->stream.out->devices;
snd_device_t d1 = uc->out_snd_device;
snd_device_t d2 = new_snd_device;
// Treat as a special case when a1 and a2 are not disjoint
if ((a1 != a2) && (a1 & a2)) {
snd_device_t d3[2];
int num_devices = 0;
int ret = platform_can_split_snd_device(popcount(a1) > 1 ? d1 : d2,
&num_devices,
d3);
if (ret < 0) {
if (ret != -ENOSYS) {
ALOGW("%s failed to split snd_device %d",
__func__,
popcount(a1) > 1 ? d1 : d2);
}
goto end;
}
// NB: case 7 is hypothetical and isn't a practical usecase yet.
// But if it does happen, we need to give priority to d2 if
// the combo devices active on the existing usecase share a backend.
// This is because we cannot have a usecase active on a combo device
// and a new usecase requests one device in this combo pair.
if (platform_check_backends_match(d3[0], d3[1])) {
return d2; // case 5
} else {
return d1; // case 1
}
} else {
if (platform_check_backends_match(d1, d2)) {
return d2; // case 2, 4
} else {
return d1; // case 6, 3
}
}
end:
return d2; // return whatever was calculated before.
}
static void check_and_route_playback_usecases(struct audio_device *adev,
struct audio_usecase *uc_info,
snd_device_t snd_device)
{
struct listnode *node;
struct audio_usecase *usecase;
bool switch_device[AUDIO_USECASE_MAX];
int i, num_uc_to_switch = 0;
bool force_routing = platform_check_and_set_playback_backend_cfg(adev,
uc_info,
snd_device);
/* For a2dp device reconfigure all active sessions
* with new AFE encoder format based on a2dp state
*/
if ((SND_DEVICE_OUT_BT_A2DP == snd_device ||
SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device ||
SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP == snd_device) &&
audio_extn_a2dp_is_force_device_switch()) {
force_routing = true;
}
/*
* This function is to make sure that all the usecases that are active on
* the hardware codec backend are always routed to any one device that is
* handled by the hardware codec.
* For example, if low-latency and deep-buffer usecases are currently active
* on speaker and out_set_parameters(headset) is received on low-latency
* output, then we have to make sure deep-buffer is also switched to headset,
* because of the limitation that both the devices cannot be enabled
* at the same time as they share the same backend.
*/
/* Disable all the usecases on the shared backend other than the
specified usecase */
for (i = 0; i < AUDIO_USECASE_MAX; i++)
switch_device[i] = false;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_CAPTURE || usecase == uc_info)
continue;
if (force_routing ||
(usecase->out_snd_device != snd_device &&
(usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND ||
usecase->devices & (AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_USB_HEADSET)) &&
platform_check_backends_match(snd_device, usecase->out_snd_device))) {
ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
__func__, use_case_table[usecase->id],
platform_get_snd_device_name(usecase->out_snd_device));
disable_audio_route(adev, usecase);
switch_device[usecase->id] = true;
num_uc_to_switch++;
}
}
if (num_uc_to_switch) {
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
disable_snd_device(adev, usecase->out_snd_device);
}
}
snd_device_t d_device;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
d_device = derive_playback_snd_device(usecase, uc_info,
snd_device);
enable_snd_device(adev, d_device);
/* Update the out_snd_device before enabling the audio route */
usecase->out_snd_device = d_device;
}
}
/* Re-route all the usecases on the shared backend other than the
specified usecase to new snd devices */
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id] ) {
enable_audio_route(adev, usecase);
if (usecase->stream.out && usecase->id == USECASE_AUDIO_PLAYBACK_VOIP) {
struct stream_out *out = usecase->stream.out;
audio_extn_utils_send_app_type_gain(out->dev,
out->app_type_cfg.app_type,
&out->app_type_cfg.gain[0]);
}
}
}
}
}
static void check_and_route_capture_usecases(struct audio_device *adev,
struct audio_usecase *uc_info,
snd_device_t snd_device)
{
struct listnode *node;
struct audio_usecase *usecase;
bool switch_device[AUDIO_USECASE_MAX];
int i, num_uc_to_switch = 0;
platform_check_and_set_capture_backend_cfg(adev, uc_info, snd_device);
/*
* This function is to make sure that all the active capture usecases
* are always routed to the same input sound device.
* For example, if audio-record and voice-call usecases are currently
* active on speaker(rx) and speaker-mic (tx) and out_set_parameters(earpiece)
* is received for voice call then we have to make sure that audio-record
* usecase is also switched to earpiece i.e. voice-dmic-ef,
* because of the limitation that two devices cannot be enabled
* at the same time if they share the same backend.
*/
for (i = 0; i < AUDIO_USECASE_MAX; i++)
switch_device[i] = false;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type != PCM_PLAYBACK &&
usecase != uc_info &&
usecase->in_snd_device != snd_device &&
((uc_info->type == VOICE_CALL &&
usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL) ||
platform_check_backends_match(snd_device,\
usecase->in_snd_device)) &&
(usecase->id != USECASE_AUDIO_SPKR_CALIB_TX)) {
ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
__func__, use_case_table[usecase->id],
platform_get_snd_device_name(usecase->in_snd_device));
disable_audio_route(adev, usecase);
switch_device[usecase->id] = true;
num_uc_to_switch++;
}
}
if (num_uc_to_switch) {
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
disable_snd_device(adev, usecase->in_snd_device);
}
}
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (switch_device[usecase->id]) {
enable_snd_device(adev, snd_device);
}
}
/* Re-route all the usecases on the shared backend other than the
specified usecase to new snd devices */
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
/* Update the in_snd_device only before enabling the audio route */
if (switch_device[usecase->id] ) {
usecase->in_snd_device = snd_device;
enable_audio_route(adev, usecase);
}
}
}
}
/* must be called with hw device mutex locked */
static int read_hdmi_channel_masks(struct stream_out *out)
{
int ret = 0;
int channels = platform_edid_get_max_channels(out->dev->platform);
switch (channels) {
/*
* Do not handle stereo output in Multi-channel cases
* Stereo case is handled in normal playback path
*/
case 6:
ALOGV("%s: HDMI supports 5.1", __func__);
out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
break;
case 8:
ALOGV("%s: HDMI supports 5.1 and 7.1 channels", __func__);
out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_7POINT1;
break;
default:
ALOGE("HDMI does not support multi channel playback");
ret = -ENOSYS;
break;
}
return ret;
}
static ssize_t read_usb_sup_sample_rates(bool is_playback,
uint32_t *supported_sample_rates,
uint32_t max_rates)
{
ssize_t count = audio_extn_usb_sup_sample_rates(is_playback,
supported_sample_rates,
max_rates);
#if !LOG_NDEBUG
for (ssize_t i=0; i<count; i++) {
ALOGV("%s %s %d", __func__, is_playback ? "P" : "C",
supported_sample_rates[i]);
}
#endif
return count;
}
static int read_usb_sup_channel_masks(bool is_playback,
audio_channel_mask_t *supported_channel_masks,
uint32_t max_masks)
{
int channels = audio_extn_usb_get_max_channels(is_playback);
int channel_count;
uint32_t num_masks = 0;
if (channels > MAX_HIFI_CHANNEL_COUNT) {
channels = MAX_HIFI_CHANNEL_COUNT;
}
if (is_playback) {
// start from 2 channels as framework currently doesn't support mono.
if (channels >= FCC_2) {
supported_channel_masks[num_masks++] = audio_channel_out_mask_from_count(FCC_2);
}
for (channel_count = FCC_2;
channel_count <= channels && num_masks < max_masks;
++channel_count) {
supported_channel_masks[num_masks++] =
audio_channel_mask_for_index_assignment_from_count(channel_count);
}
} else {
// For capture we report all supported channel masks from 1 channel up.
channel_count = MIN_CHANNEL_COUNT;
// audio_channel_in_mask_from_count() does the right conversion to either positional or
// indexed mask
for ( ; channel_count <= channels && num_masks < max_masks; channel_count++) {
audio_channel_mask_t mask = AUDIO_CHANNEL_NONE;
if (channel_count <= FCC_2) {
mask = audio_channel_in_mask_from_count(channel_count);
supported_channel_masks[num_masks++] = mask;
}
const audio_channel_mask_t index_mask =
audio_channel_mask_for_index_assignment_from_count(channel_count);
if (mask != index_mask && num_masks < max_masks) { // ensure index mask added.
supported_channel_masks[num_masks++] = index_mask;
}
}
}
#ifdef NDEBUG
for (size_t i = 0; i < num_masks; ++i) {
ALOGV("%s: %s supported ch %d supported_channel_masks[%zu] %08x num_masks %d", __func__,
is_playback ? "P" : "C", channels, i, supported_channel_masks[i], num_masks);
}
#endif
return num_masks;
}
static int read_usb_sup_formats(bool is_playback __unused,
audio_format_t *supported_formats,
uint32_t max_formats __unused)
{
int bitwidth = audio_extn_usb_get_max_bit_width(is_playback);
switch (bitwidth) {
case 24:
// XXX : usb.c returns 24 for s24 and s24_le?
supported_formats[0] = AUDIO_FORMAT_PCM_24_BIT_PACKED;
break;
case 32:
supported_formats[0] = AUDIO_FORMAT_PCM_32_BIT;
break;
case 16:
default :
supported_formats[0] = AUDIO_FORMAT_PCM_16_BIT;
break;
}
ALOGV("%s: %s supported format %d", __func__,
is_playback ? "P" : "C", bitwidth);
return 1;
}
static int read_usb_sup_params_and_compare(bool is_playback,
audio_format_t *format,
audio_format_t *supported_formats,
uint32_t max_formats,
audio_channel_mask_t *mask,
audio_channel_mask_t *supported_channel_masks,
uint32_t max_masks,
uint32_t *rate,
uint32_t *supported_sample_rates,
uint32_t max_rates) {
int ret = 0;
int num_formats;
int num_masks;
int num_rates;
int i;
num_formats = read_usb_sup_formats(is_playback, supported_formats,
max_formats);
num_masks = read_usb_sup_channel_masks(is_playback, supported_channel_masks,
max_masks);
num_rates = read_usb_sup_sample_rates(is_playback,
supported_sample_rates, max_rates);
#define LUT(table, len, what, dflt) \
for (i=0; i<len && (table[i] != what); i++); \
if (i==len) { ret |= (what == dflt ? 0 : -1); what=table[0]; }
LUT(supported_formats, num_formats, *format, AUDIO_FORMAT_DEFAULT);
LUT(supported_channel_masks, num_masks, *mask, AUDIO_CHANNEL_NONE);
LUT(supported_sample_rates, num_rates, *rate, 0);
#undef LUT
return ret < 0 ? -EINVAL : 0; // HACK TBD
}
static bool is_usb_ready(struct audio_device *adev, bool is_playback)
{
// Check if usb is ready.
// The usb device may have been removed quickly after insertion and hence
// no longer available. This will show up as empty channel masks, or rates.
pthread_mutex_lock(&adev->lock);
uint32_t supported_sample_rate;
// we consider usb ready if we can fetch at least one sample rate.
const bool ready = read_usb_sup_sample_rates(
is_playback, &supported_sample_rate, 1 /* max_rates */) > 0;
pthread_mutex_unlock(&adev->lock);
return ready;
}
static audio_usecase_t get_voice_usecase_id_from_list(struct audio_device *adev)
{
struct audio_usecase *usecase;
struct listnode *node;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == VOICE_CALL) {
ALOGV("%s: usecase id %d", __func__, usecase->id);
return usecase->id;
}
}
return USECASE_INVALID;
}
struct audio_usecase *get_usecase_from_list(struct audio_device *adev,
audio_usecase_t uc_id)
{
struct audio_usecase *usecase;
struct listnode *node;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->id == uc_id)
return usecase;
}
return NULL;
}
static bool force_device_switch(struct audio_usecase *usecase)
{
if (usecase->type == PCM_CAPTURE || usecase->stream.out == NULL) {
return false;
}
// Force all A2DP output devices to reconfigure for proper AFE encode format
// Also handle a case where in earlier A2DP start failed as A2DP stream was
// in suspended state, hence try to trigger a retry when we again get a routing request.
if ((usecase->stream.out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) &&
audio_extn_a2dp_is_force_device_switch()) {
ALOGD("%s: Force A2DP device switch to update new encoder config", __func__);
return true;
}
return false;
}
struct stream_in *adev_get_active_input(const struct audio_device *adev)
{
struct listnode *node;
struct stream_in *last_active_in = NULL;
/* Get last added active input.
* TODO: We may use a priority mechanism to pick highest priority active source */
list_for_each(node, &adev->usecase_list)
{
struct audio_usecase *usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_CAPTURE && usecase->stream.in != NULL) {
last_active_in = usecase->stream.in;
}
}
return last_active_in;
}
struct stream_in *get_voice_communication_input(const struct audio_device *adev)
{
struct listnode *node;
/* First check active inputs with voice communication source and then
* any input if audio mode is in communication */
list_for_each(node, &adev->usecase_list)
{
struct audio_usecase *usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_CAPTURE && usecase->stream.in != NULL &&
usecase->stream.in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) {
return usecase->stream.in;
}
}
if (adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
return adev_get_active_input(adev);
}
return NULL;
}
/*
* Aligned with policy.h
*/
static inline int source_priority(int inputSource)
{
switch (inputSource) {
case AUDIO_SOURCE_VOICE_COMMUNICATION:
return 9;
case AUDIO_SOURCE_CAMCORDER:
return 8;
case AUDIO_SOURCE_VOICE_PERFORMANCE:
return 7;
case AUDIO_SOURCE_UNPROCESSED:
return 6;
case AUDIO_SOURCE_MIC:
return 5;
case AUDIO_SOURCE_ECHO_REFERENCE:
return 4;
case AUDIO_SOURCE_FM_TUNER:
return 3;
case AUDIO_SOURCE_VOICE_RECOGNITION:
return 2;
case AUDIO_SOURCE_HOTWORD:
return 1;
default:
break;
}
return 0;
}
static struct stream_in *get_priority_input(struct audio_device *adev)
{
struct listnode *node;
struct audio_usecase *usecase;
int last_priority = 0, priority;
struct stream_in *priority_in = NULL;
struct stream_in *in;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_CAPTURE) {
in = usecase->stream.in;
if (!in)
continue;
priority = source_priority(in->source);
if (priority > last_priority) {
last_priority = priority;
priority_in = in;
}
}
}
return priority_in;
}
int select_devices_with_force_switch(struct audio_device *adev,
audio_usecase_t uc_id,
bool force_switch)
{
snd_device_t out_snd_device = SND_DEVICE_NONE;
snd_device_t in_snd_device = SND_DEVICE_NONE;
struct audio_usecase *usecase = NULL;
struct audio_usecase *vc_usecase = NULL;
struct audio_usecase *hfp_usecase = NULL;
audio_usecase_t hfp_ucid;
struct listnode *node;
int status = 0;
struct audio_usecase *voip_usecase = get_usecase_from_list(adev,
USECASE_AUDIO_PLAYBACK_VOIP);
usecase = get_usecase_from_list(adev, uc_id);
if (usecase == NULL) {
ALOGE("%s: Could not find the usecase(%d)", __func__, uc_id);
return -EINVAL;
}
if ((usecase->type == VOICE_CALL) ||
(usecase->type == PCM_HFP_CALL)) {
out_snd_device = platform_get_output_snd_device(adev->platform,
usecase->stream.out->devices);
in_snd_device = platform_get_input_snd_device(adev->platform,
NULL,
usecase->stream.out->devices);
usecase->devices = usecase->stream.out->devices;
} else {
/*
* If the voice call is active, use the sound devices of voice call usecase
* so that it would not result any device switch. All the usecases will
* be switched to new device when select_devices() is called for voice call
* usecase. This is to avoid switching devices for voice call when
* check_and_route_playback_usecases() is called below.
*/
if (voice_is_in_call(adev)) {
vc_usecase = get_usecase_from_list(adev,
get_voice_usecase_id_from_list(adev));
if ((vc_usecase != NULL) &&
((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
(vc_usecase->devices == AUDIO_DEVICE_OUT_HEARING_AID) ||
(usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL))) {
in_snd_device = vc_usecase->in_snd_device;
out_snd_device = vc_usecase->out_snd_device;
}
} else if (audio_extn_hfp_is_active(adev)) {
hfp_ucid = audio_extn_hfp_get_usecase();
hfp_usecase = get_usecase_from_list(adev, hfp_ucid);
if (hfp_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
in_snd_device = hfp_usecase->in_snd_device;
out_snd_device = hfp_usecase->out_snd_device;
}
}
if (usecase->type == PCM_PLAYBACK) {
usecase->devices = usecase->stream.out->devices;
in_snd_device = SND_DEVICE_NONE;
if (out_snd_device == SND_DEVICE_NONE) {
struct stream_out *voip_out = adev->primary_output;
struct stream_in *voip_in = get_voice_communication_input(adev);
out_snd_device = platform_get_output_snd_device(adev->platform,
usecase->stream.out->devices);
if (voip_usecase)
voip_out = voip_usecase->stream.out;
if (usecase->stream.out == voip_out && voip_in != NULL) {
select_devices(adev, voip_in->usecase);
}
}
} else if (usecase->type == PCM_CAPTURE) {
usecase->devices = usecase->stream.in->device;
out_snd_device = SND_DEVICE_NONE;
if (in_snd_device == SND_DEVICE_NONE) {
audio_devices_t out_device = AUDIO_DEVICE_NONE;
struct stream_in *voip_in = get_voice_communication_input(adev);
struct stream_in *priority_in = NULL;
if (voip_in != NULL) {
struct audio_usecase *voip_usecase = get_usecase_from_list(adev,
USECASE_AUDIO_PLAYBACK_VOIP);
usecase->stream.in->enable_ec_port = false;
if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) {
out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX;
} else if (voip_usecase) {
out_device = voip_usecase->stream.out->devices;
} else if (adev->primary_output &&
!adev->primary_output->standby) {
out_device = adev->primary_output->devices;
} else {
/* forcing speaker o/p device to get matching i/p pair
in case o/p is not routed from same primary HAL */
out_device = AUDIO_DEVICE_OUT_SPEAKER;
}
priority_in = voip_in;
} else {
/* get the input with the highest priority source*/
priority_in = get_priority_input(adev);
if (!priority_in)
priority_in = usecase->stream.in;
}
in_snd_device = platform_get_input_snd_device(adev->platform,
priority_in,
out_device);
}
}
}
if (out_snd_device == usecase->out_snd_device &&
in_snd_device == usecase->in_snd_device) {
if (!force_device_switch(usecase) && !force_switch)
return 0;
}
if (is_a2dp_device(out_snd_device) && !audio_extn_a2dp_is_ready()) {
ALOGD("SCO/A2DP is selected but they are not connected/ready hence dont route");
return 0;
}
if ((out_snd_device == SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP ||
out_snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP) &&
(!audio_extn_a2dp_is_ready())) {
ALOGW("%s: A2DP profile is not ready, routing to speaker only", __func__);
if (out_snd_device == SND_DEVICE_OUT_SPEAKER_SAFE_AND_BT_A2DP)
out_snd_device = SND_DEVICE_OUT_SPEAKER_SAFE;
else
out_snd_device = SND_DEVICE_OUT_SPEAKER;
}
if (usecase->id == USECASE_INCALL_MUSIC_UPLINK ||
usecase->id == USECASE_INCALL_MUSIC_UPLINK2) {
out_snd_device = SND_DEVICE_OUT_VOICE_MUSIC_TX;
}
if (out_snd_device != SND_DEVICE_NONE &&
out_snd_device != adev->last_logged_snd_device[uc_id][0]) {
ALOGD("%s: changing use case %s output device from(%d: %s, acdb %d) to (%d: %s, acdb %d)",
__func__,
use_case_table[uc_id],
adev->last_logged_snd_device[uc_id][0],
platform_get_snd_device_name(adev->last_logged_snd_device[uc_id][0]),
adev->last_logged_snd_device[uc_id][0] != SND_DEVICE_NONE ?
platform_get_snd_device_acdb_id(adev->last_logged_snd_device[uc_id][0]) :
-1,
out_snd_device,
platform_get_snd_device_name(out_snd_device),
platform_get_snd_device_acdb_id(out_snd_device));
adev->last_logged_snd_device[uc_id][0] = out_snd_device;
}
if (in_snd_device != SND_DEVICE_NONE &&
in_snd_device != adev->last_logged_snd_device[uc_id][1]) {
ALOGD("%s: changing use case %s input device from(%d: %s, acdb %d) to (%d: %s, acdb %d)",
__func__,
use_case_table[uc_id],
adev->last_logged_snd_device[uc_id][1],
platform_get_snd_device_name(adev->last_logged_snd_device[uc_id][1]),
adev->last_logged_snd_device[uc_id][1] != SND_DEVICE_NONE ?
platform_get_snd_device_acdb_id(adev->last_logged_snd_device[uc_id][1]) :
-1,
in_snd_device,
platform_get_snd_device_name(in_snd_device),
platform_get_snd_device_acdb_id(in_snd_device));
adev->last_logged_snd_device[uc_id][1] = in_snd_device;
}
/*
* Limitation: While in call, to do a device switch we need to disable
* and enable both RX and TX devices though one of them is same as current
* device.
*/
if ((usecase->type == VOICE_CALL) &&
(usecase->in_snd_device != SND_DEVICE_NONE) &&
(usecase->out_snd_device != SND_DEVICE_NONE)) {
status = platform_switch_voice_call_device_pre(adev->platform);
/* Disable sidetone only if voice call already exists */
if (voice_is_call_state_active(adev))
voice_set_sidetone(adev, usecase->out_snd_device, false);
}
/* Disable current sound devices */
if (usecase->out_snd_device != SND_DEVICE_NONE) {
disable_audio_route(adev, usecase);
disable_snd_device(adev, usecase->out_snd_device);
}
if (usecase->in_snd_device != SND_DEVICE_NONE) {
disable_audio_route(adev, usecase);
disable_snd_device(adev, usecase->in_snd_device);
}
/* Applicable only on the targets that has external modem.
* New device information should be sent to modem before enabling
* the devices to reduce in-call device switch time.
*/
if ((usecase->type == VOICE_CALL) &&
(usecase->in_snd_device != SND_DEVICE_NONE) &&
(usecase->out_snd_device != SND_DEVICE_NONE)) {
status = platform_switch_voice_call_enable_device_config(adev->platform,
out_snd_device,
in_snd_device);
}
/* Enable new sound devices */
if (out_snd_device != SND_DEVICE_NONE) {
if ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
(usecase->devices & (AUDIO_DEVICE_OUT_USB_DEVICE|AUDIO_DEVICE_OUT_USB_HEADSET)) ||
(usecase->devices & AUDIO_DEVICE_OUT_ALL_A2DP))
check_and_route_playback_usecases(adev, usecase, out_snd_device);
enable_snd_device(adev, out_snd_device);
}
if (in_snd_device != SND_DEVICE_NONE) {
check_and_route_capture_usecases(adev, usecase, in_snd_device);
enable_snd_device(adev, in_snd_device);
}
if (usecase->type == VOICE_CALL)
status = platform_switch_voice_call_device_post(adev->platform,
out_snd_device,
in_snd_device);
usecase->in_snd_device = in_snd_device;
usecase->out_snd_device = out_snd_device;
audio_extn_tfa_98xx_set_mode();
enable_audio_route(adev, usecase);
/* If input stream is already running the effect needs to be
applied on the new input device that's being enabled here. */
if (in_snd_device != SND_DEVICE_NONE)
check_and_enable_effect(adev);
/* Applicable only on the targets that has external modem.
* Enable device command should be sent to modem only after
* enabling voice call mixer controls
*/
if (usecase->type == VOICE_CALL) {
status = platform_switch_voice_call_usecase_route_post(adev->platform,
out_snd_device,
in_snd_device);
/* Enable sidetone only if voice call already exists */
if (voice_is_call_state_active(adev))
voice_set_sidetone(adev, out_snd_device, true);
}
if (usecase->type != PCM_CAPTURE && voip_usecase) {
struct stream_out *voip_out = voip_usecase->stream.out;
audio_extn_utils_send_app_type_gain(adev,
voip_out->app_type_cfg.app_type,
&voip_out->app_type_cfg.gain[0]);
}
return status;
}
int select_devices(struct audio_device *adev,
audio_usecase_t uc_id)
{
return select_devices_with_force_switch(adev, uc_id, false);
}
static int stop_input_stream(struct stream_in *in)
{
int i, ret = 0;
struct audio_usecase *uc_info;
struct audio_device *adev = in->dev;
struct stream_in *priority_in = NULL;
ALOGV("%s: enter: usecase(%d: %s)", __func__,
in->usecase, use_case_table[in->usecase]);
uc_info = get_usecase_from_list(adev, in->usecase);
if (uc_info == NULL) {
ALOGE("%s: Could not find the usecase (%d) in the list",
__func__, in->usecase);
return -EINVAL;
}
priority_in = get_priority_input(adev);
/* Close in-call recording streams */
voice_check_and_stop_incall_rec_usecase(adev, in);
/* 1. Disable stream specific mixer controls */
disable_audio_route(adev, uc_info);
/* 2. Disable the tx device */
disable_snd_device(adev, uc_info->in_snd_device);
list_remove(&uc_info->list);
free(uc_info);
if (priority_in == in) {
priority_in = get_priority_input(adev);
if (priority_in)
select_devices(adev, priority_in->usecase);
}
ALOGV("%s: exit: status(%d)", __func__, ret);
return ret;
}
int start_input_stream(struct stream_in *in)
{
/* 1. Enable output device and stream routing controls */
int ret = 0;
struct audio_usecase *uc_info;
struct audio_device *adev = in->dev;
ALOGV("%s: enter: usecase(%d)", __func__, in->usecase);
if (audio_extn_tfa_98xx_is_supported() && !audio_ssr_status(adev))
return -EIO;
if (in->card_status == CARD_STATUS_OFFLINE ||
adev->card_status == CARD_STATUS_OFFLINE) {
ALOGW("in->card_status or adev->card_status offline, try again");
ret = -EAGAIN;
goto error_config;
}
/* Check if source matches incall recording usecase criteria */
ret = voice_check_and_set_incall_rec_usecase(adev, in);
if (ret)
goto error_config;
else
ALOGV("%s: usecase(%d)", __func__, in->usecase);
in->pcm_device_id = platform_get_pcm_device_id(in->usecase, PCM_CAPTURE);
if (in->pcm_device_id < 0) {
ALOGE("%s: Could not find PCM device id for the usecase(%d)",
__func__, in->usecase);
ret = -EINVAL;
goto error_config;
}
uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
uc_info->id = in->usecase;
uc_info->type = PCM_CAPTURE;
uc_info->stream.in = in;
uc_info->devices = in->device;
uc_info->in_snd_device = SND_DEVICE_NONE;
uc_info->out_snd_device = SND_DEVICE_NONE;
list_add_tail(&adev->usecase_list, &uc_info->list);
audio_streaming_hint_start();
audio_extn_perf_lock_acquire();
select_devices(adev, in->usecase);
if (in->usecase == USECASE_AUDIO_RECORD_MMAP) {
if (in->pcm == NULL || !pcm_is_ready(in->pcm)) {
ALOGE("%s: pcm stream not ready", __func__);
goto error_open;
}
ret = pcm_start(in->pcm);
if (ret < 0) {
ALOGE("%s: MMAP pcm_start failed ret %d", __func__, ret);
goto error_open;
}
} else {
unsigned int flags = PCM_IN | PCM_MONOTONIC;
unsigned int pcm_open_retry_count = 0;
if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
flags |= PCM_MMAP | PCM_NOIRQ;
pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
} else if (in->realtime) {
flags |= PCM_MMAP | PCM_NOIRQ;
}
ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d",
__func__, adev->snd_card, in->pcm_device_id, in->config.channels);
while (1) {
in->pcm = pcm_open(adev->snd_card, in->pcm_device_id,
flags, &in->config);
if (in->pcm == NULL || !pcm_is_ready(in->pcm)) {
ALOGE("%s: %s", __func__, pcm_get_error(in->pcm));
if (in->pcm != NULL) {
pcm_close(in->pcm);
in->pcm = NULL;
}
if (pcm_open_retry_count-- == 0) {
ret = -EIO;
goto error_open;
}
usleep(PROXY_OPEN_WAIT_TIME * 1000);
continue;
}
break;
}
ALOGV("%s: pcm_prepare", __func__);
ret = pcm_prepare(in->pcm);
if (ret < 0) {
ALOGE("%s: pcm_prepare returned %d", __func__, ret);
pcm_close(in->pcm);
in->pcm = NULL;
goto error_open;
}
if (in->realtime) {
ret = pcm_start(in->pcm);
if (ret < 0) {
ALOGE("%s: RT pcm_start failed ret %d", __func__, ret);
pcm_close(in->pcm);
in->pcm = NULL;
goto error_open;
}
}
}
register_in_stream(in);
check_and_enable_effect(adev);
audio_extn_audiozoom_set_microphone_direction(in, in->zoom);
audio_extn_audiozoom_set_microphone_field_dimension(in, in->direction);
audio_streaming_hint_end();
audio_extn_perf_lock_release();
ALOGV("%s: exit", __func__);
return 0;
error_open:
stop_input_stream(in);
audio_streaming_hint_end();
audio_extn_perf_lock_release();
error_config:
ALOGW("%s: exit: status(%d)", __func__, ret);
return ret;
}
void lock_input_stream(struct stream_in *in)
{
pthread_mutex_lock(&in->pre_lock);
pthread_mutex_lock(&in->lock);
pthread_mutex_unlock(&in->pre_lock);
}
void lock_output_stream(struct stream_out *out)
{
pthread_mutex_lock(&out->pre_lock);
pthread_mutex_lock(&out->lock);
pthread_mutex_unlock(&out->pre_lock);
}
/* must be called with out->lock locked */
static int send_offload_cmd_l(struct stream_out* out, int command)
{
struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd));
ALOGVV("%s %d", __func__, command);
cmd->cmd = command;
list_add_tail(&out->offload_cmd_list, &cmd->node);
pthread_cond_signal(&out->offload_cond);
return 0;
}
/* must be called iwth out->lock locked */
static void stop_compressed_output_l(struct stream_out *out)
{
out->offload_state = OFFLOAD_STATE_IDLE;
out->playback_started = 0;
out->send_new_metadata = 1;
if (out->compr != NULL) {
compress_stop(out->compr);
while (out->offload_thread_blocked) {
pthread_cond_wait(&out->cond, &out->lock);
}
}
}
static void *offload_thread_loop(void *context)
{
struct stream_out *out = (struct stream_out *) context;
struct listnode *item;
setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
set_sched_policy(0, SP_FOREGROUND);
prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0);
ALOGV("%s", __func__);
lock_output_stream(out);
out->offload_state = OFFLOAD_STATE_IDLE;
out->playback_started = 0;
for (;;) {
struct offload_cmd *cmd = NULL;
stream_callback_event_t event;
bool send_callback = false;
ALOGVV("%s offload_cmd_list %d out->offload_state %d",
__func__, list_empty(&out->offload_cmd_list),
out->offload_state);
if (list_empty(&out->offload_cmd_list)) {
ALOGV("%s SLEEPING", __func__);
pthread_cond_wait(&out->offload_cond, &out->lock);
ALOGV("%s RUNNING", __func__);
continue;
}
item = list_head(&out->offload_cmd_list);
cmd = node_to_item(item, struct offload_cmd, node);
list_remove(item);
ALOGVV("%s STATE %d CMD %d out->compr %p",
__func__, out->offload_state, cmd->cmd, out->compr);
if (cmd->cmd == OFFLOAD_CMD_EXIT) {
free(cmd);
break;
}
if (out->compr == NULL) {
ALOGE("%s: Compress handle is NULL", __func__);
free(cmd);
pthread_cond_signal(&out->cond);
continue;
}
out->offload_thread_blocked = true;
pthread_mutex_unlock(&out->lock);
send_callback = false;
switch (cmd->cmd) {
case OFFLOAD_CMD_WAIT_FOR_BUFFER:
compress_wait(out->compr, -1);
send_callback = true;
event = STREAM_CBK_EVENT_WRITE_READY;
break;
case OFFLOAD_CMD_PARTIAL_DRAIN:
compress_next_track(out->compr);
compress_partial_drain(out->compr);
send_callback = true;
event = STREAM_CBK_EVENT_DRAIN_READY;
/* Resend the metadata for next iteration */
out->send_new_metadata = 1;
break;
case OFFLOAD_CMD_DRAIN:
compress_drain(out->compr);
send_callback = true;
event = STREAM_CBK_EVENT_DRAIN_READY;
break;
case OFFLOAD_CMD_ERROR:
send_callback = true;
event = STREAM_CBK_EVENT_ERROR;
break;
default:
ALOGE("%s unknown command received: %d", __func__, cmd->cmd);
break;
}
lock_output_stream(out);
out->offload_thread_blocked = false;
pthread_cond_signal(&out->cond);
if (send_callback) {
ALOGVV("%s: sending offload_callback event %d", __func__, event);
out->offload_callback(event, NULL, out->offload_cookie);
}
free(cmd);
}
pthread_cond_signal(&out->cond);
while (!list_empty(&out->offload_cmd_list)) {
item = list_head(&out->offload_cmd_list);
list_remove(item);
free(node_to_item(item, struct offload_cmd, node));
}
pthread_mutex_unlock(&out->lock);
return NULL;
}
static int create_offload_callback_thread(struct stream_out *out)
{
pthread_cond_init(&out->offload_cond, (const pthread_condattr_t *) NULL);
list_init(&out->offload_cmd_list);
pthread_create(&out->offload_thread, (const pthread_attr_t *) NULL,
offload_thread_loop, out);
return 0;
}
static int destroy_offload_callback_thread(struct stream_out *out)
{
lock_output_stream(out);
stop_compressed_output_l(out);
send_offload_cmd_l(out, OFFLOAD_CMD_EXIT);
pthread_mutex_unlock(&out->lock);
pthread_join(out->offload_thread, (void **) NULL);
pthread_cond_destroy(&out->offload_cond);
return 0;
}
static bool allow_hdmi_channel_config(struct audio_device *adev)
{
struct listnode *node;
struct audio_usecase *usecase;
bool ret = true;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
/*
* If voice call is already existing, do not proceed further to avoid
* disabling/enabling both RX and TX devices, CSD calls, etc.
* Once the voice call done, the HDMI channels can be configured to
* max channels of remaining use cases.
*/
if (usecase->id == USECASE_VOICE_CALL) {
ALOGV("%s: voice call is active, no change in HDMI channels",
__func__);
ret = false;
break;
} else if (usecase->id == USECASE_AUDIO_PLAYBACK_HIFI) {
ALOGV("%s: hifi playback is active, "
"no change in HDMI channels", __func__);
ret = false;
break;
}
}
}
return ret;
}
static int check_and_set_hdmi_channels(struct audio_device *adev,
unsigned int channels)
{
struct listnode *node;
struct audio_usecase *usecase;
/* Check if change in HDMI channel config is allowed */
if (!allow_hdmi_channel_config(adev))
return 0;
if (channels == adev->cur_hdmi_channels) {
ALOGV("%s: Requested channels are same as current", __func__);
return 0;
}
platform_set_hdmi_channels(adev->platform, channels);
adev->cur_hdmi_channels = channels;
/*
* Deroute all the playback streams routed to HDMI so that
* the back end is deactivated. Note that backend will not
* be deactivated if any one stream is connected to it.
*/
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_PLAYBACK &&
usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
disable_audio_route(adev, usecase);
}
}
/*
* Enable all the streams disabled above. Now the HDMI backend
* will be activated with new channel configuration
*/
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_PLAYBACK &&
usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
enable_audio_route(adev, usecase);
}
}
return 0;
}
static int check_and_set_usb_service_interval(struct audio_device *adev,
struct audio_usecase *uc_info,
bool min)
{
struct listnode *node;
struct audio_usecase *usecase;
bool switch_usecases = false;
bool reconfig = false;
if ((uc_info->id != USECASE_AUDIO_PLAYBACK_MMAP) &&
(uc_info->id != USECASE_AUDIO_PLAYBACK_ULL))
return -1;
/* set if the valid usecase do not already exist */
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_PLAYBACK &&
(audio_is_usb_out_device(usecase->devices & AUDIO_DEVICE_OUT_ALL_USB))) {
switch (usecase->id) {
case USECASE_AUDIO_PLAYBACK_MMAP:
case USECASE_AUDIO_PLAYBACK_ULL:
// cannot reconfig while mmap/ull is present.
return -1;
default:
switch_usecases = true;
break;
}
}
if (switch_usecases)
break;
}
/*
* client can try to set service interval in start_output_stream
* to min or to 0 (i.e reset) in stop_output_stream .
*/
unsigned long service_interval =
audio_extn_usb_find_service_interval(min, true /*playback*/);
int ret = platform_set_usb_service_interval(adev->platform,
true /*playback*/,
service_interval,
&reconfig);
/* no change or not supported or no active usecases */
if (ret || !reconfig || !switch_usecases)
return -1;
return 0;
#undef VALID_USECASE
}
static int stop_output_stream(struct stream_out *out)
{
int i, ret = 0;
struct audio_usecase *uc_info;
struct audio_device *adev = out->dev;
ALOGV("%s: enter: usecase(%d: %s)", __func__,
out->usecase, use_case_table[out->usecase]);
uc_info = get_usecase_from_list(adev, out->usecase);
if (uc_info == NULL) {
ALOGE("%s: Could not find the usecase (%d) in the list",
__func__, out->usecase);
return -EINVAL;
}
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
if (adev->visualizer_stop_output != NULL)
adev->visualizer_stop_output(out->handle, out->pcm_device_id);
if (adev->offload_effects_stop_output != NULL)
adev->offload_effects_stop_output(out->handle, out->pcm_device_id);
} else if (out->usecase == USECASE_AUDIO_PLAYBACK_ULL ||
out->usecase == USECASE_AUDIO_PLAYBACK_MMAP) {
audio_low_latency_hint_end();
}
if (out->usecase == USECASE_INCALL_MUSIC_UPLINK ||
out->usecase == USECASE_INCALL_MUSIC_UPLINK2) {
voice_set_device_mute_flag(adev, false);
}
/* 1. Get and set stream specific mixer controls */
disable_audio_route(adev, uc_info);
/* 2. Disable the rx device */
disable_snd_device(adev, uc_info->out_snd_device);
list_remove(&uc_info->list);
audio_extn_extspk_update(adev->extspk);
/* Must be called after removing the usecase from list */
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
check_and_set_hdmi_channels(adev, DEFAULT_HDMI_OUT_CHANNELS);
else if (audio_is_usb_out_device(out->devices & AUDIO_DEVICE_OUT_ALL_USB)) {
ret = check_and_set_usb_service_interval(adev, uc_info, false /*min*/);
if (ret == 0) {
/* default service interval was successfully updated,
reopen USB backend with new service interval */
check_and_route_playback_usecases(adev, uc_info, uc_info->out_snd_device);
}
ret = 0;
}
/* 1) media + voip output routing to handset must route media back to
speaker when voip stops.
2) trigger voip input to reroute when voip output changes to
hearing aid. */
if (out->usecase == USECASE_AUDIO_PLAYBACK_VOIP ||
out->devices & AUDIO_DEVICE_OUT_SPEAKER_SAFE) {
struct listnode *node;
struct audio_usecase *usecase;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if ((usecase->type == PCM_CAPTURE &&
usecase->id != USECASE_AUDIO_RECORD_VOIP)
|| usecase == uc_info)
continue;
ALOGD("%s: select_devices at usecase(%d: %s) after removing the usecase(%d: %s)",
__func__, usecase->id, use_case_table[usecase->id],
out->usecase, use_case_table[out->usecase]);
select_devices(adev, usecase->id);
}
}
free(uc_info);
ALOGV("%s: exit: status(%d)", __func__, ret);
return ret;
}
struct pcm* pcm_open_prepare_helper(unsigned int snd_card, unsigned int pcm_device_id,
unsigned int flags, unsigned int pcm_open_retry_count,
struct pcm_config *config)
{
struct pcm* pcm = NULL;
while (1) {
pcm = pcm_open(snd_card, pcm_device_id, flags, config);
if (pcm == NULL || !pcm_is_ready(pcm)) {
ALOGE("%s: %s", __func__, pcm_get_error(pcm));
if (pcm != NULL) {
pcm_close(pcm);
pcm = NULL;
}
if (pcm_open_retry_count-- == 0)
return NULL;
usleep(PROXY_OPEN_WAIT_TIME * 1000);
continue;
}
break;
}
if (pcm_is_ready(pcm)) {
int ret = pcm_prepare(pcm);
if (ret < 0) {
ALOGE("%s: pcm_prepare returned %d", __func__, ret);
pcm_close(pcm);
pcm = NULL;
}
}
return pcm;
}
int start_output_stream(struct stream_out *out)
{
int ret = 0;
struct audio_usecase *uc_info;
struct audio_device *adev = out->dev;
bool a2dp_combo = false;
ALOGV("%s: enter: usecase(%d: %s) %s devices(%#x)",
__func__, out->usecase, use_case_table[out->usecase],
out->usecase == USECASE_AUDIO_PLAYBACK_WITH_HAPTICS ? "(with haptics)" : "",
out->devices);
if (out->card_status == CARD_STATUS_OFFLINE ||
adev->card_status == CARD_STATUS_OFFLINE) {
ALOGW("out->card_status or adev->card_status offline, try again");
ret = -EAGAIN;
goto error_config;
}
//Update incall music usecase to reflect correct voice session
if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
ret = voice_extn_check_and_set_incall_music_usecase(adev, out);
if (ret != 0) {
ALOGE("%s: Incall music delivery usecase cannot be set error:%d",
__func__, ret);
goto error_config;
}
}
if (out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
if (out->devices & (AUDIO_DEVICE_OUT_SPEAKER | AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
a2dp_combo = true;
} else if (!audio_extn_a2dp_is_ready() &&
!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
ALOGE("%s: A2DP profile is not ready, return error", __func__);
ret = -EAGAIN;
goto error_config;
}
}
out->pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
if (out->pcm_device_id < 0) {
ALOGE("%s: Invalid PCM device id(%d) for the usecase(%d)",
__func__, out->pcm_device_id, out->usecase);
ret = -EINVAL;
goto error_config;
}
uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
uc_info->id = out->usecase;
uc_info->type = PCM_PLAYBACK;
uc_info->stream.out = out;
uc_info->devices = out->devices;
uc_info->in_snd_device = SND_DEVICE_NONE;
uc_info->out_snd_device = SND_DEVICE_NONE;
/* This must be called before adding this usecase to the list */
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
check_and_set_hdmi_channels(adev, out->config.channels);
else if (audio_is_usb_out_device(out->devices & AUDIO_DEVICE_OUT_ALL_USB)) {
check_and_set_usb_service_interval(adev, uc_info, true /*min*/);
/* USB backend is not reopened immediately.
This is eventually done as part of select_devices */
}
list_add_tail(&adev->usecase_list, &uc_info->list);
audio_streaming_hint_start();
audio_extn_perf_lock_acquire();
if (!(out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) ||
audio_extn_a2dp_is_ready()) {
select_devices(adev, out->usecase);
}
if ((out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) &&
(!audio_extn_a2dp_is_ready() || !adev->a2dp_started)) {
if (a2dp_combo) {
audio_devices_t dev = out->devices;
if (dev & AUDIO_DEVICE_OUT_SPEAKER_SAFE)
out->devices = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
else
out->devices = AUDIO_DEVICE_OUT_SPEAKER;
select_devices(adev, out->usecase);
out->devices = dev;
} else if (!audio_extn_a2dp_is_ready()) {
check_a2dp_restore_l(adev, out, false);
} else {
ALOGE("%s: A2DP is not started, return error", __func__);
ret = -EINVAL;
goto error_open;
}
}
audio_extn_extspk_update(adev->extspk);
if (out->usecase == USECASE_INCALL_MUSIC_UPLINK ||
out->usecase == USECASE_INCALL_MUSIC_UPLINK2) {
voice_set_device_mute_flag(adev, true);
}
ALOGV("%s: Opening PCM device card_id(%d) device_id(%d) format(%#x)",
__func__, adev->snd_card, out->pcm_device_id, out->config.format);
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
out->pcm = NULL;
out->compr = compress_open(adev->snd_card, out->pcm_device_id,
COMPRESS_IN, &out->compr_config);
if (out->compr && !is_compress_ready(out->compr)) {
ALOGE("%s: %s", __func__, compress_get_error(out->compr));
compress_close(out->compr);
out->compr = NULL;
ret = -EIO;
goto error_open;
}
if (out->offload_callback)
compress_nonblock(out->compr, out->non_blocking);
if (adev->visualizer_start_output != NULL) {
int capture_device_id =
platform_get_pcm_device_id(USECASE_AUDIO_RECORD_AFE_PROXY,
PCM_CAPTURE);
adev->visualizer_start_output(out->handle, out->pcm_device_id,
adev->snd_card, capture_device_id);
}
if (adev->offload_effects_start_output != NULL)
adev->offload_effects_start_output(out->handle, out->pcm_device_id);
} else if (out->usecase == USECASE_AUDIO_PLAYBACK_MMAP) {
if (out->pcm == NULL || !pcm_is_ready(out->pcm)) {
ALOGE("%s: pcm stream not ready", __func__);
goto error_open;
}
ret = pcm_start(out->pcm);
if (ret < 0) {
ALOGE("%s: MMAP pcm_start failed ret %d", __func__, ret);
goto error_open;
}
} else {
unsigned int flags = PCM_OUT | PCM_MONOTONIC;
unsigned int pcm_open_retry_count = 0;
if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
flags |= PCM_MMAP | PCM_NOIRQ;
pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
} else if (out->realtime) {
flags |= PCM_MMAP | PCM_NOIRQ;
}
out->pcm = pcm_open_prepare_helper(adev->snd_card, out->pcm_device_id,
flags, pcm_open_retry_count,
&(out->config));
if (out->pcm == NULL) {
ret = -EIO;
goto error_open;
}
if (out->usecase == USECASE_AUDIO_PLAYBACK_WITH_HAPTICS) {
if (adev->haptic_pcm != NULL) {
pcm_close(adev->haptic_pcm);
adev->haptic_pcm = NULL;
}
adev->haptic_pcm = pcm_open_prepare_helper(adev->snd_card,
adev->haptic_pcm_device_id,
flags, pcm_open_retry_count,
&(adev->haptics_config));
// failure to open haptics pcm shouldnt stop audio,
// so do not close audio pcm in case of error
}
if (out->realtime) {
ret = pcm_start(out->pcm);
if (ret < 0) {
ALOGE("%s: RT pcm_start failed ret %d", __func__, ret);
pcm_close(out->pcm);
out->pcm = NULL;
goto error_open;
}
}
if ((out->usecase == USECASE_AUDIO_PLAYBACK_LOW_LATENCY
|| out->usecase == USECASE_AUDIO_PLAYBACK_DEEP_BUFFER
|| out->usecase == USECASE_AUDIO_PLAYBACK_ULL)) {
out_set_pcm_volume(&out->stream, out->volume_l, out->volume_r);
}
}
register_out_stream(out);
audio_streaming_hint_end();
audio_extn_perf_lock_release();
audio_extn_tfa_98xx_enable_speaker();
if (out->usecase == USECASE_AUDIO_PLAYBACK_ULL ||
out->usecase == USECASE_AUDIO_PLAYBACK_MMAP) {
audio_low_latency_hint_start();
}
// consider a scenario where on pause lower layers are tear down.
// so on resume, swap mixer control need to be sent only when
// backend is active, hence rather than sending from enable device
// sending it from start of stream
platform_set_swap_channels(adev, true);
ALOGV("%s: exit", __func__);
return 0;
error_open:
if (adev->haptic_pcm) {
pcm_close(adev->haptic_pcm);
adev->haptic_pcm = NULL;
}
audio_streaming_hint_end();
audio_extn_perf_lock_release();
stop_output_stream(out);
error_config:
return ret;
}
static int check_input_parameters(uint32_t sample_rate,
audio_format_t format,
int channel_count, bool is_usb_hifi)
{
if ((format != AUDIO_FORMAT_PCM_16_BIT) &&
(format != AUDIO_FORMAT_PCM_8_24_BIT) &&
(format != AUDIO_FORMAT_PCM_24_BIT_PACKED) &&
!(is_usb_hifi && (format == AUDIO_FORMAT_PCM_32_BIT))) {
ALOGE("%s: unsupported AUDIO FORMAT (%d) ", __func__, format);
return -EINVAL;
}
int max_channel_count = is_usb_hifi ? MAX_HIFI_CHANNEL_COUNT : MAX_CHANNEL_COUNT;
if ((channel_count < MIN_CHANNEL_COUNT) || (channel_count > max_channel_count)) {
ALOGE("%s: unsupported channel count (%d) passed Min / Max (%d / %d)", __func__,
channel_count, MIN_CHANNEL_COUNT, max_channel_count);
return -EINVAL;
}
switch (sample_rate) {
case 8000:
case 11025:
case 12000:
case 16000:
case 22050:
case 24000:
case 32000:
case 44100:
case 48000:
case 96000:
break;
default:
ALOGE("%s: unsupported (%d) samplerate passed ", __func__, sample_rate);
return -EINVAL;
}
return 0;
}
/** Add a value in a list if not already present.
* @return true if value was successfully inserted or already present,
* false if the list is full and does not contain the value.
*/
static bool register_uint(uint32_t value, uint32_t* list, size_t list_length) {
for (size_t i = 0; i < list_length; i++) {
if (list[i] == value) return true; // value is already present
if (list[i] == 0) { // no values in this slot
list[i] = value;
return true; // value inserted
}
}
return false; // could not insert value
}
/** Add channel_mask in supported_channel_masks if not already present.
* @return true if channel_mask was successfully inserted or already present,
* false if supported_channel_masks is full and does not contain channel_mask.
*/
static void register_channel_mask(audio_channel_mask_t channel_mask,
audio_channel_mask_t supported_channel_masks[static MAX_SUPPORTED_CHANNEL_MASKS]) {
ALOGE_IF(!register_uint(channel_mask, supported_channel_masks, MAX_SUPPORTED_CHANNEL_MASKS),
"%s: stream can not declare supporting its channel_mask %x", __func__, channel_mask);
}
/** Add format in supported_formats if not already present.
* @return true if format was successfully inserted or already present,
* false if supported_formats is full and does not contain format.
*/
static void register_format(audio_format_t format,
audio_format_t supported_formats[static MAX_SUPPORTED_FORMATS]) {
ALOGE_IF(!register_uint(format, supported_formats, MAX_SUPPORTED_FORMATS),
"%s: stream can not declare supporting its format %x", __func__, format);
}
/** Add sample_rate in supported_sample_rates if not already present.
* @return true if sample_rate was successfully inserted or already present,
* false if supported_sample_rates is full and does not contain sample_rate.
*/
static void register_sample_rate(uint32_t sample_rate,
uint32_t supported_sample_rates[static MAX_SUPPORTED_SAMPLE_RATES]) {
ALOGE_IF(!register_uint(sample_rate, supported_sample_rates, MAX_SUPPORTED_SAMPLE_RATES),
"%s: stream can not declare supporting its sample rate %x", __func__, sample_rate);
}
static size_t get_stream_buffer_size(size_t duration_ms,
uint32_t sample_rate,
audio_format_t format,
int channel_count,
bool is_low_latency)
{
// Compute target frames based on time or period size.
size_t target_frames = is_low_latency
? configured_low_latency_capture_period_size // record only
: (sample_rate * duration_ms) / 1000;
// Round up to a multiple of 16 frames in case sizing for the MixerThread.
if (!is_low_latency) { // low latency flag set for record only
target_frames = (target_frames + 0xf) & ~0xf;
}
// Buffer size is the target frames multiplied by the frame size in bytes.
const size_t frame_size = channel_count * audio_bytes_per_sample(format);
const size_t buffer_size = target_frames * frame_size;
return buffer_size;
}
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
return out->sample_rate;
}
static int out_set_sample_rate(struct audio_stream *stream __unused, uint32_t rate __unused)
{
return -ENOSYS;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
return out->compr_config.fragment_size;
}
return out->config.period_size * out->af_period_multiplier *
audio_stream_out_frame_size((const struct audio_stream_out *)stream);
}
static uint32_t out_get_channels(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
return out->channel_mask;
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
return out->format;
}
static int out_set_format(struct audio_stream *stream __unused, audio_format_t format __unused)
{
return -ENOSYS;
}
/* must be called with out->lock locked */
static int out_standby_l(struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
bool do_stop = true;
if (!out->standby) {
if (adev->adm_deregister_stream)
adev->adm_deregister_stream(adev->adm_data, out->handle);
pthread_mutex_lock(&adev->lock);
out->standby = true;
if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
if (out->pcm) {
pcm_close(out->pcm);
out->pcm = NULL;
if (out->usecase == USECASE_AUDIO_PLAYBACK_WITH_HAPTICS) {
if (adev->haptic_pcm) {
pcm_close(adev->haptic_pcm);
adev->haptic_pcm = NULL;
}
if (adev->haptic_buffer != NULL) {
free(adev->haptic_buffer);
adev->haptic_buffer = NULL;
adev->haptic_buffer_size = 0;
}
}
}
if (out->usecase == USECASE_AUDIO_PLAYBACK_MMAP) {
do_stop = out->playback_started;
out->playback_started = false;
if (out->mmap_shared_memory_fd >= 0) {
ALOGV("%s: closing mmap_shared_memory_fd = %d",
__func__, out->mmap_shared_memory_fd);
close(out->mmap_shared_memory_fd);
out->mmap_shared_memory_fd = -1;
}
}
} else {
stop_compressed_output_l(out);
out->gapless_mdata.encoder_delay = 0;
out->gapless_mdata.encoder_padding = 0;
if (out->compr != NULL) {
compress_close(out->compr);
out->compr = NULL;
}
}
if (do_stop) {
stop_output_stream(out);
}
pthread_mutex_unlock(&adev->lock);
}
return 0;
}
static int out_standby(struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
ALOGV("%s: enter: usecase(%d: %s)", __func__,
out->usecase, use_case_table[out->usecase]);
lock_output_stream(out);
out_standby_l(stream);
pthread_mutex_unlock(&out->lock);
ALOGV("%s: exit", __func__);
return 0;
}
static int out_on_error(struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
bool do_standby = false;
lock_output_stream(out);
if (!out->standby) {
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
stop_compressed_output_l(out);
send_offload_cmd_l(out, OFFLOAD_CMD_ERROR);
} else
do_standby = true;
}
pthread_mutex_unlock(&out->lock);
if (do_standby)
return out_standby(&out->stream.common);
return 0;
}
static int out_dump(const struct audio_stream *stream, int fd)
{
struct stream_out *out = (struct stream_out *)stream;
// We try to get the lock for consistency,
// but it isn't necessary for these variables.
// If we're not in standby, we may be blocked on a write.
const bool locked = (pthread_mutex_trylock(&out->lock) == 0);
dprintf(fd, " Standby: %s\n", out->standby ? "yes" : "no");
dprintf(fd, " Frames written: %lld\n", (long long)out->written);
char buffer[256]; // for statistics formatting
simple_stats_to_string(&out->fifo_underruns, buffer, sizeof(buffer));
dprintf(fd, " Fifo frame underruns: %s\n", buffer);
if (out->start_latency_ms.n > 0) {
simple_stats_to_string(&out->start_latency_ms, buffer, sizeof(buffer));
dprintf(fd, " Start latency ms: %s\n", buffer);
}
if (locked) {
pthread_mutex_unlock(&out->lock);
}
// dump error info
(void)error_log_dump(
out->error_log, fd, " " /* prefix */, 0 /* lines */, 0 /* limit_ns */);
return 0;
}
static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms)
{
int ret = 0;
char value[32];
struct compr_gapless_mdata tmp_mdata;
if (!out || !parms) {
return -EINVAL;
}
ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value));
if (ret >= 0) {
tmp_mdata.encoder_delay = atoi(value); //whats a good limit check?
} else {
return -EINVAL;
}
ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value));
if (ret >= 0) {
tmp_mdata.encoder_padding = atoi(value);
} else {
return -EINVAL;
}
out->gapless_mdata = tmp_mdata;
out->send_new_metadata = 1;
ALOGV("%s new encoder delay %u and padding %u", __func__,
out->gapless_mdata.encoder_delay, out->gapless_mdata.encoder_padding);
return 0;
}
static bool output_drives_call(struct audio_device *adev, struct stream_out *out)
{
return out == adev->primary_output || out == adev->voice_tx_output;
}
static int get_alive_usb_card(struct str_parms* parms) {
int card;
if ((str_parms_get_int(parms, "card", &card) >= 0) &&
!audio_extn_usb_alive(card)) {
return card;
}
return -ENODEV;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
struct audio_usecase *usecase;
struct listnode *node;
struct str_parms *parms;
char value[32];
int ret, val = 0;
bool select_new_device = false;
int status = 0;
bool bypass_a2dp = false;
bool forced_speaker_fallback = false;
ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s",
__func__, out->usecase, use_case_table[out->usecase], kvpairs);
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);
lock_output_stream(out);
if (val == AUDIO_DEVICE_NONE &&
audio_is_usb_out_device(out->devices)) {
val = AUDIO_DEVICE_OUT_SPEAKER;
forced_speaker_fallback = true;
}
pthread_mutex_lock(&adev->lock);
/*
* When HDMI cable is unplugged the music playback is paused and
* the policy manager sends routing=0. But the audioflinger
* continues to write data until standby time (3sec).
* As the HDMI core is turned off, the write gets blocked.
* Avoid this by routing audio to speaker until standby.
*/
if (out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL &&
val == AUDIO_DEVICE_NONE) {
val = AUDIO_DEVICE_OUT_SPEAKER;
forced_speaker_fallback = true;
}
/*
* When A2DP is disconnected the
* music playback is paused and the policy manager sends routing=0
* But the audioflingercontinues to write data until standby time
* (3sec). As BT is turned off, the write gets blocked.
* Avoid this by routing audio to speaker until standby.
*/
if ((out->devices & AUDIO_DEVICE_OUT_BLUETOOTH_A2DP) &&
(val == AUDIO_DEVICE_NONE) &&
!audio_extn_a2dp_is_ready() &&
!adev->bt_sco_on) {
val = AUDIO_DEVICE_OUT_SPEAKER;
forced_speaker_fallback = true;
}
/* To avoid a2dp to sco overlapping / BT device improper state
* check with BT lib about a2dp streaming support before routing
*/
if (val & AUDIO_DEVICE_OUT_ALL_A2DP) {
if (!audio_extn_a2dp_is_ready()) {
if (val & (AUDIO_DEVICE_OUT_SPEAKER | AUDIO_DEVICE_OUT_SPEAKER_SAFE)) {
//combo usecase just by pass a2dp
ALOGW("%s: A2DP profile is not ready,routing to speaker only", __func__);
bypass_a2dp = true;
} else {
ALOGE("%s: A2DP profile is not ready,ignoring routing request", __func__);
/* update device to a2dp and don't route as BT returned error
* However it is still possible a2dp routing called because
* of current active device disconnection (like wired headset)
*/
out->devices = val;
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&adev->lock);
status = -ENOSYS;
goto routing_fail;
}
}
}
audio_devices_t new_dev = val;
// Workaround: If routing to an non existing usb device, fail gracefully
// The routing request will otherwise block during 10 second
int card;
if (audio_is_usb_out_device(new_dev) &&
(card = get_alive_usb_card(parms)) >= 0) {
ALOGW("out_set_parameters() ignoring rerouting to non existing USB card %d", card);
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&out->lock);
status = -ENOSYS;
goto routing_fail;
}
/*
* select_devices() call below switches all the usecases on the same
* backend to the new device. Refer to check_and_route_playback_usecases() in
* the select_devices(). But how do we undo this?
*
* For example, music playback is active on headset (deep-buffer usecase)
* and if we go to ringtones and select a ringtone, low-latency usecase
* will be started on headset+speaker. As we can't enable headset+speaker
* and headset devices at the same time, select_devices() switches the music
* playback to headset+speaker while starting low-lateny usecase for ringtone.
* So when the ringtone playback is completed, how do we undo the same?
*
* We are relying on the out_set_parameters() call on deep-buffer output,
* once the ringtone playback is ended.
* NOTE: We should not check if the current devices are same as new devices.
* Because select_devices() must be called to switch back the music
* playback to headset.
*/
if (new_dev != AUDIO_DEVICE_NONE) {
bool same_dev = out->devices == new_dev;
out->devices = new_dev;
if (output_drives_call(adev, out)) {
if (!voice_is_call_state_active(adev)) {
if (adev->mode == AUDIO_MODE_IN_CALL) {
adev->current_call_output = out;
ret = voice_start_call(adev);
}
} else {
adev->current_call_output = out;
voice_update_devices_for_all_voice_usecases(adev);
}
}
if (!out->standby) {
int volume_delay_us = 0;
if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
pthread_mutex_lock(&out->compr_mute_lock);
if (out->a2dp_compress_mute &&
(!(new_dev & AUDIO_DEVICE_OUT_ALL_A2DP) ||
audio_extn_a2dp_is_ready())) {
out->a2dp_compress_mute = false;
}
float volume_l = out->volume_l;
float volume_r = out->volume_r;
if (out->a2dp_compress_mute || forced_speaker_fallback) {
volume_l = 0.0;
volume_r = 0.0;
}
if (volume_l != out->applied_volume_l || volume_r != out->applied_volume_r)
volume_delay_us = COMPRESS_OFFLOAD_PLAYBACK_LATENCY * 2000;
out_set_compr_volume(&out->stream, volume_l, volume_r);
pthread_mutex_unlock(&out->compr_mute_lock);
} else if (out->usecase == USECASE_AUDIO_PLAYBACK_LOW_LATENCY ||
out->usecase == USECASE_AUDIO_PLAYBACK_DEEP_BUFFER ||
out->usecase == USECASE_AUDIO_PLAYBACK_ULL) {
float volume_l = out->volume_l;
float volume_r = out->volume_r;
if (forced_speaker_fallback) {
volume_l = 0.0;
volume_r = 0.0;
}
if (volume_l != out->applied_volume_l || volume_r != out->applied_volume_r)
volume_delay_us = (int)platform_render_latency(out) * 2;
out_set_pcm_volume(&out->stream, volume_l, volume_r);
}
if (volume_delay_us > 0)
usleep(volume_delay_us * 2);
if (!same_dev) {
ALOGV("update routing change");
// inform adm before actual routing to prevent glitches.
if (adev->adm_on_routing_change) {
adev->adm_on_routing_change(adev->adm_data,
out->handle);
}
}
if (!bypass_a2dp) {
select_devices(adev, out->usecase);
} else {
if (new_dev & AUDIO_DEVICE_OUT_SPEAKER_SAFE)
out->devices = AUDIO_DEVICE_OUT_SPEAKER_SAFE;
else
out->devices = AUDIO_DEVICE_OUT_SPEAKER;
select_devices(adev, out->usecase);
out->devices = new_dev;
}
audio_extn_tfa_98xx_update();
// on device switch force swap, lower functions will make sure
// to check if swap is allowed or not.
if (!same_dev)
platform_set_swap_channels(adev, true);
}
}
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&out->lock);
/*handles device and call state changes*/
audio_extn_extspk_update(adev->extspk);
}
routing_fail:
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
parse_compress_metadata(out, parms);
}
str_parms_destroy(parms);
ALOGV("%s: exit: code(%d)", __func__, status);
return status;
}
static bool stream_get_parameter_channels(struct str_parms *query,
struct str_parms *reply,
audio_channel_mask_t *supported_channel_masks) {
int ret = -1;
char value[ARRAY_SIZE(channels_name_to_enum_table) * 32 /* max channel name size */];
bool first = true;
size_t i, j;
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
ret = 0;
value[0] = '\0';
i = 0;
while (supported_channel_masks[i] != 0) {
for (j = 0; j < ARRAY_SIZE(channels_name_to_enum_table); j++) {
if (channels_name_to_enum_table[j].value == supported_channel_masks[i]) {
if (!first) {
strcat(value, "|");
}
strcat(value, channels_name_to_enum_table[j].name);
first = false;
break;
}
}
i++;
}
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value);
}
return ret >= 0;
}
static bool stream_get_parameter_formats(struct str_parms *query,
struct str_parms *reply,
audio_format_t *supported_formats) {
int ret = -1;
char value[256];
int i;
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
ret = 0;
value[0] = '\0';
switch (supported_formats[0]) {
case AUDIO_FORMAT_PCM_16_BIT:
strcat(value, "AUDIO_FORMAT_PCM_16_BIT");
break;
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
strcat(value, "AUDIO_FORMAT_PCM_24_BIT_PACKED");
break;
case AUDIO_FORMAT_PCM_32_BIT:
strcat(value, "AUDIO_FORMAT_PCM_32_BIT");
break;
default:
ALOGE("%s: unsupported format %#x", __func__,
supported_formats[0]);
break;
}
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
}
return ret >= 0;
}
static bool stream_get_parameter_rates(struct str_parms *query,
struct str_parms *reply,
uint32_t *supported_sample_rates) {
int i;
char value[256];
int ret = -1;
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
ret = 0;
value[0] = '\0';
i=0;
int cursor = 0;
while (supported_sample_rates[i]) {
int avail = sizeof(value) - cursor;
ret = snprintf(value + cursor, avail, "%s%d",
cursor > 0 ? "|" : "",
supported_sample_rates[i]);
if (ret < 0 || ret >= avail) {
// if cursor is at the last element of the array
// overwrite with \0 is duplicate work as
// snprintf already put a \0 in place.
// else
// we had space to write the '|' at value[cursor]
// (which will be overwritten) or no space to fill
// the first element (=> cursor == 0)
value[cursor] = '\0';
break;
}
cursor += ret;
++i;
}
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES,
value);
}
return ret >= 0;
}
static char* out_get_parameters(const struct audio_stream *stream, const char *keys)
{
struct stream_out *out = (struct stream_out *)stream;
struct str_parms *query = str_parms_create_str(keys);
char *str;
struct str_parms *reply = str_parms_create();
bool replied = false;
ALOGV("%s: enter: keys - %s", __func__, keys);
replied |= stream_get_parameter_channels(query, reply,
&out->supported_channel_masks[0]);
replied |= stream_get_parameter_formats(query, reply,
&out->supported_formats[0]);
replied |= stream_get_parameter_rates(query, reply,
&out->supported_sample_rates[0]);
if (replied) {
str = str_parms_to_str(reply);
} else {
str = strdup("");
}
str_parms_destroy(query);
str_parms_destroy(reply);
ALOGV("%s: exit: returns - %s", __func__, str);
return str;
}
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
uint32_t hw_delay, period_ms;
struct stream_out *out = (struct stream_out *)stream;
uint32_t latency;
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD)
return COMPRESS_OFFLOAD_PLAYBACK_LATENCY;
else if ((out->realtime) ||
(out->usecase == USECASE_AUDIO_PLAYBACK_MMAP)) {
// since the buffer won't be filled up faster than realtime,
// return a smaller number
period_ms = (out->af_period_multiplier * out->config.period_size *
1000) / (out->config.rate);
hw_delay = platform_render_latency(out)/1000;
return period_ms + hw_delay;
}
latency = (out->config.period_count * out->config.period_size * 1000) /
(out->config.rate);
if (AUDIO_DEVICE_OUT_ALL_A2DP & out->devices)
latency += audio_extn_a2dp_get_encoder_latency();
return latency;
}
static int out_set_compr_volume(struct audio_stream_out *stream, float left,
float right)
{
struct stream_out *out = (struct stream_out *)stream;
int volume[2];
char mixer_ctl_name[128];
struct audio_device *adev = out->dev;
struct mixer_ctl *ctl;
int pcm_device_id = platform_get_pcm_device_id(out->usecase,
PCM_PLAYBACK);
if (left == out->applied_volume_l && right == out->applied_volume_r)
return 0;
snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
"Compress Playback %d Volume", pcm_device_id);
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
return -EINVAL;
}
ALOGV("%s: ctl for mixer cmd - %s, left %f, right %f",
__func__, mixer_ctl_name, left, right);
volume[0] = (int)(left * COMPRESS_PLAYBACK_VOLUME_MAX);
volume[1] = (int)(right * COMPRESS_PLAYBACK_VOLUME_MAX);
mixer_ctl_set_array(ctl, volume, sizeof(volume) / sizeof(volume[0]));
out->applied_volume_l = left;
out->applied_volume_r = right;
return 0;
}
static int out_set_pcm_volume(struct audio_stream_out *stream, float left,
float right)
{
struct stream_out *out = (struct stream_out *)stream;
if (left == out->applied_volume_l && right == out->applied_volume_r)
return 0;
/* Volume control for pcm playback */
if (left != right) {
return -EINVAL;
} else {
char mixer_ctl_name[128];
struct audio_device *adev = out->dev;
struct mixer_ctl *ctl;
int pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "Playback %d Volume", pcm_device_id);
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGE("%s : Could not get ctl for mixer cmd - %s", __func__, mixer_ctl_name);
return -EINVAL;
}
int volume = (int) (left * PCM_PLAYBACK_VOLUME_MAX);
int ret = mixer_ctl_set_value(ctl, 0, volume);
if (ret < 0) {
ALOGE("%s: Could not set ctl, error:%d ", __func__, ret);
return -EINVAL;
}
ALOGV("%s : Pcm set volume value %d left %f", __func__, volume, left);
out->applied_volume_l = left;
out->applied_volume_r = right;
return 0;
}
}
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
struct stream_out *out = (struct stream_out *)stream;
int ret = 0;
if (out->usecase == USECASE_AUDIO_PLAYBACK_HIFI) {
/* only take left channel into account: the API is for stereo anyway */
out->muted = (left == 0.0f);
return 0;
} else if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
pthread_mutex_lock(&out->compr_mute_lock);
ALOGV("%s: compress mute %d", __func__, out->a2dp_compress_mute);
if (!out->a2dp_compress_mute)
ret = out_set_compr_volume(stream, left, right);
out->volume_l = left;
out->volume_r = right;
pthread_mutex_unlock(&out->compr_mute_lock);
return ret;
} else if (out->usecase == USECASE_AUDIO_PLAYBACK_VOIP) {
out->app_type_cfg.gain[0] = (int)(left * VOIP_PLAYBACK_VOLUME_MAX);
out->app_type_cfg.gain[1] = (int)(right * VOIP_PLAYBACK_VOLUME_MAX);
if (!out->standby) {
// if in standby, cached volume will be sent after stream is opened
audio_extn_utils_send_app_type_gain(out->dev,
out->app_type_cfg.app_type,
&out->app_type_cfg.gain[0]);
}
return 0;
}
return -ENOSYS;
}
// note: this call is safe only if the stream_cb is
// removed first in close_output_stream (as is done now).
static void out_snd_mon_cb(void * stream, struct str_parms * parms)
{
if (!stream || !parms)
return;
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
card_status_t status;
int card;
if (parse_snd_card_status(parms, &card, &status) < 0)
return;
pthread_mutex_lock(&adev->lock);
bool valid_cb = (card == adev->snd_card);
pthread_mutex_unlock(&adev->lock);
if (!valid_cb)
return;
lock_output_stream(out);
if (out->card_status != status)
out->card_status = status;
pthread_mutex_unlock(&out->lock);
ALOGW("out_snd_mon_cb for card %d usecase %s, status %s", card,
use_case_table[out->usecase],
status == CARD_STATUS_OFFLINE ? "offline" : "online");
if (status == CARD_STATUS_OFFLINE)
out_on_error(stream);
return;
}
#ifdef NO_AUDIO_OUT
static ssize_t out_write_for_no_output(struct audio_stream_out *stream,
const void *buffer __unused, size_t bytes)
{
struct stream_out *out = (struct stream_out *)stream;
/* No Output device supported other than BT for playback.
* Sleep for the amount of buffer duration
*/
lock_output_stream(out);
usleep(bytes * 1000000 / audio_stream_out_frame_size(
(const struct audio_stream_out *)&out->stream) /
out_get_sample_rate(&out->stream.common));
pthread_mutex_unlock(&out->lock);
return bytes;
}
#endif
static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
size_t bytes)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
ssize_t ret = 0;
int error_code = ERROR_CODE_STANDBY;
lock_output_stream(out);
// this is always nonzero
const size_t frame_size = audio_stream_out_frame_size(stream);
const size_t frames = bytes / frame_size;
if (out->usecase == USECASE_AUDIO_PLAYBACK_MMAP) {
error_code = ERROR_CODE_WRITE;
goto exit;
}
if ((out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) &&
(audio_extn_a2dp_is_suspended())) {
if (!(out->devices & (AUDIO_DEVICE_OUT_SPEAKER | AUDIO_DEVICE_OUT_SPEAKER_SAFE))) {
if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
ret = -EIO;
goto exit;
}
}
}
const bool was_in_standby = out->standby;
if (out->standby) {
out->standby = false;
const int64_t startNs = systemTime(SYSTEM_TIME_MONOTONIC);
pthread_mutex_lock(&adev->lock);
ret = start_output_stream(out);
/* ToDo: If use case is compress offload should return 0 */
if (ret != 0) {
out->standby = true;
pthread_mutex_unlock(&adev->lock);
goto exit;
}
// after standby always force set last known cal step
// dont change level anywhere except at the audio_hw_send_gain_dep_calibration
ALOGD("%s: retry previous failed cal level set", __func__);
send_gain_dep_calibration_l();
pthread_mutex_unlock(&adev->lock);
// log startup time in ms.
simple_stats_log(
&out->start_latency_ms, (systemTime(SYSTEM_TIME_MONOTONIC) - startNs) * 1e-6);
out->last_fifo_valid = false; // we're coming out of standby, last_fifo isn't valid.
}
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
ALOGVV("%s: writing buffer (%zu bytes) to compress device", __func__, bytes);
if (out->send_new_metadata) {
ALOGVV("send new gapless metadata");
compress_set_gapless_metadata(out->compr, &out->gapless_mdata);
out->send_new_metadata = 0;
}
unsigned int avail;
struct timespec tstamp;
ret = compress_get_hpointer(out->compr, &avail, &tstamp);
/* Do not limit write size if the available frames count is unknown */
if (ret != 0) {
avail = bytes;
}
if (avail == 0) {
ret = 0;
} else {
// check for compressed format underrun, essentially an empty buffer check
// for a lack of better measurement.
if (!was_in_standby && avail == out->kernel_buffer_size) {
ALOGW("%s: compressed buffer empty (underrun)", __func__);
simple_stats_log(&out->fifo_underruns, 1.); // Note: log one frame for compressed.
}
if (avail > bytes) {
avail = bytes;
}
ret = compress_write(out->compr, buffer, avail);
ALOGVV("%s: writing buffer (%d bytes) to compress device returned %zd",
__func__, avail, ret);
}
if (ret >= 0 && ret < (ssize_t)bytes) {
send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
}
if (ret > 0 && !out->playback_started) {
compress_start(out->compr);
out->playback_started = 1;
out->offload_state = OFFLOAD_STATE_PLAYING;
}
if (ret < 0) {
error_log_log(out->error_log, ERROR_CODE_WRITE, audio_utils_get_real_time_ns());
} else {
out->written += ret; // accumulate bytes written for offload.
}
pthread_mutex_unlock(&out->lock);
// TODO: consider logging offload pcm
return ret;
} else {
error_code = ERROR_CODE_WRITE;
if (out->pcm) {
size_t bytes_to_write = bytes;
if (out->muted)
memset((void *)buffer, 0, bytes);
// FIXME: this can be removed once audio flinger mixer supports mono output
if (out->usecase == USECASE_AUDIO_PLAYBACK_VOIP ||
out->usecase == USECASE_INCALL_MUSIC_UPLINK ||
out->usecase == USECASE_INCALL_MUSIC_UPLINK2) {
size_t channel_count = audio_channel_count_from_out_mask(out->channel_mask);
int16_t *src = (int16_t *)buffer;
int16_t *dst = (int16_t *)buffer;
LOG_ALWAYS_FATAL_IF(out->config.channels != 1 || channel_count != 2 ||
out->format != AUDIO_FORMAT_PCM_16_BIT,
"out_write called for VOIP use case with wrong properties");
for (size_t i = 0; i < frames ; i++, dst++, src += 2) {
*dst = (int16_t)(((int32_t)src[0] + (int32_t)src[1]) >> 1);
}
bytes_to_write /= 2;
}
// Note: since out_get_presentation_position() is called alternating with out_write()
// by AudioFlinger, we can check underruns using the prior timestamp read.
// (Alternately we could check if the buffer is empty using pcm_get_htimestamp().
if (out->last_fifo_valid) {
// compute drain to see if there is an underrun.
const int64_t current_ns = systemTime(SYSTEM_TIME_MONOTONIC); // sys call
const int64_t frames_by_time =
(current_ns - out->last_fifo_time_ns) * out->config.rate / NANOS_PER_SECOND;
const int64_t underrun = frames_by_time - out->last_fifo_frames_remaining;
if (underrun > 0) {
simple_stats_log(&out->fifo_underruns, underrun);
ALOGW("%s: underrun(%lld) "
"frames_by_time(%lld) > out->last_fifo_frames_remaining(%lld)",
__func__,
(long long)out->fifo_underruns.n,
(long long)frames_by_time,
(long long)out->last_fifo_frames_remaining);
}
out->last_fifo_valid = false; // we're writing below, mark fifo info as stale.
}
long ns = (frames * (int64_t) NANOS_PER_SECOND) / out->config.rate;
request_out_focus(out, ns);
bool use_mmap = is_mmap_usecase(out->usecase) || out->realtime;
if (use_mmap) {
ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes_to_write);
} else {
if (out->usecase == USECASE_AUDIO_PLAYBACK_WITH_HAPTICS) {
size_t channel_count = audio_channel_count_from_out_mask(out->channel_mask);
size_t bytes_per_sample = audio_bytes_per_sample(out->format);
size_t frame_size = channel_count * bytes_per_sample;
size_t frame_count = bytes_to_write / frame_size;
bool force_haptic_path =
property_get_bool("vendor.audio.test_haptic", false);
// extract Haptics data from Audio buffer
bool alloc_haptic_buffer = false;
int haptic_channel_count = adev->haptics_config.channels;
size_t haptic_frame_size = bytes_per_sample * haptic_channel_count;
size_t audio_frame_size = frame_size - haptic_frame_size;
size_t total_haptic_buffer_size = frame_count * haptic_frame_size;
if (adev->haptic_buffer == NULL) {
alloc_haptic_buffer = true;
} else if (adev->haptic_buffer_size < total_haptic_buffer_size) {
free(adev->haptic_buffer);
adev->haptic_buffer_size = 0;
alloc_haptic_buffer = true;
}
if (alloc_haptic_buffer) {
adev->haptic_buffer = (uint8_t *)calloc(1, total_haptic_buffer_size);
adev->haptic_buffer_size = total_haptic_buffer_size;
}
size_t src_index = 0, aud_index = 0, hap_index = 0;
uint8_t *audio_buffer = (uint8_t *)buffer;
uint8_t *haptic_buffer = adev->haptic_buffer;
// This is required for testing only. This works for stereo data only.
// One channel is fed to audio stream and other to haptic stream for testing.
if (force_haptic_path) {
audio_frame_size = haptic_frame_size = bytes_per_sample;
}
for (size_t i = 0; i < frame_count; i++) {
for (size_t j = 0; j < audio_frame_size; j++)
audio_buffer[aud_index++] = audio_buffer[src_index++];
for (size_t j = 0; j < haptic_frame_size; j++)
haptic_buffer[hap_index++] = audio_buffer[src_index++];
}
// This is required for testing only.
// Discard haptic channel data.
if (force_haptic_path) {
src_index += haptic_frame_size;
}
// write to audio pipeline
ret = pcm_write(out->pcm,
(void *)audio_buffer,
frame_count * audio_frame_size);
// write to haptics pipeline
if (adev->haptic_pcm)
ret = pcm_write(adev->haptic_pcm,
(void *)adev->haptic_buffer,
frame_count * haptic_frame_size);
} else {
ret = pcm_write(out->pcm, (void *)buffer, bytes_to_write);
}
}
release_out_focus(out, ns);
} else {
LOG_ALWAYS_FATAL("out->pcm is NULL after starting output stream");
}
}
exit:
// For PCM we always consume the buffer and return #bytes regardless of ret.
if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
out->written += frames;
}
long long sleeptime_us = 0;
if (ret != 0) {
error_log_log(out->error_log, error_code, audio_utils_get_real_time_ns());
if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) {
ALOGE_IF(out->pcm != NULL,
"%s: error %zd - %s", __func__, ret, pcm_get_error(out->pcm));
sleeptime_us = frames * 1000000LL / out_get_sample_rate(&out->stream.common);
// usleep not guaranteed for values over 1 second but we don't limit here.
}
}
pthread_mutex_unlock(&out->lock);
if (ret != 0) {
out_on_error(&out->stream.common);
if (sleeptime_us != 0)
usleep(sleeptime_us);
}
return bytes;
}
static int out_get_render_position(const struct audio_stream_out *stream,
uint32_t *dsp_frames)
{
struct stream_out *out = (struct stream_out *)stream;
*dsp_frames = 0;
if ((out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) && (dsp_frames != NULL)) {
lock_output_stream(out);
if (out->compr != NULL) {
unsigned long frames = 0;
// TODO: check return value
compress_get_tstamp(out->compr, &frames, &out->sample_rate);
*dsp_frames = (uint32_t)frames;
ALOGVV("%s rendered frames %d sample_rate %d",
__func__, *dsp_frames, out->sample_rate);
}
pthread_mutex_unlock(&out->lock);
return 0;
} else
return -ENODATA;
}
static int out_add_audio_effect(const struct audio_stream *stream __unused,
effect_handle_t effect __unused)
{
return 0;
}
static int out_remove_audio_effect(const struct audio_stream *stream __unused,
effect_handle_t effect __unused)
{
return 0;
}
static int out_get_next_write_timestamp(const struct audio_stream_out *stream __unused,
int64_t *timestamp __unused)
{
return -ENOSYS;
}
static int out_get_presentation_position(const struct audio_stream_out *stream,
uint64_t *frames, struct timespec *timestamp)
{
struct stream_out *out = (struct stream_out *)stream;
int ret = -ENODATA;
unsigned long dsp_frames;
lock_output_stream(out);
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
if (out->compr != NULL) {
// TODO: check return value
compress_get_tstamp(out->compr, &dsp_frames,
&out->sample_rate);
// Adjustment accounts for A2DP encoder latency with offload usecases
// Note: Encoder latency is returned in ms.
if (AUDIO_DEVICE_OUT_ALL_A2DP & out->devices) {
unsigned long offset =
(audio_extn_a2dp_get_encoder_latency() * out->sample_rate / 1000);
dsp_frames = (dsp_frames > offset) ? (dsp_frames - offset) : 0;
}
ALOGVV("%s rendered frames %ld sample_rate %d",
__func__, dsp_frames, out->sample_rate);
*frames = dsp_frames;
ret = 0;
/* this is the best we can do */
clock_gettime(CLOCK_MONOTONIC, timestamp);
}
} else {
if (out->pcm) {
unsigned int avail;
if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
// pcm_get_htimestamp() computes the available frames by comparing
// the alsa driver hw_ptr and the appl_ptr levels.
// In underrun, the hw_ptr may keep running and report an excessively
// large number available number.
if (avail > out->kernel_buffer_size) {
ALOGW("%s: avail:%u > kernel_buffer_size:%zu clamping!",
__func__, avail, out->kernel_buffer_size);
avail = out->kernel_buffer_size;
out->last_fifo_frames_remaining = 0;
} else {
out->last_fifo_frames_remaining = out->kernel_buffer_size - avail;
}
out->last_fifo_valid = true;
out->last_fifo_time_ns = audio_utils_ns_from_timespec(timestamp);
int64_t signed_frames = out->written - out->last_fifo_frames_remaining;
ALOGVV("%s: frames:%lld avail:%u kernel_buffer_size:%zu",
__func__, (long long)signed_frames, avail, out->kernel_buffer_size);
// This adjustment accounts for buffering after app processor.
// It is based on estimated DSP latency per use case, rather than exact.
signed_frames -=
(platform_render_latency(out) * out->sample_rate / 1000000LL);
// Adjustment accounts for A2DP encoder latency with non-offload usecases
// Note: Encoder latency is returned in ms, while platform_render_latency in us.
if (AUDIO_DEVICE_OUT_ALL_A2DP & out->devices) {
signed_frames -=
(audio_extn_a2dp_get_encoder_latency() * out->sample_rate / 1000);
}
// It would be unusual for this value to be negative, but check just in case ...
if (signed_frames >= 0) {
*frames = signed_frames;
ret = 0;
}
}
}
}
pthread_mutex_unlock(&out->lock);
return ret;
}
static int out_set_callback(struct audio_stream_out *stream,
stream_callback_t callback, void *cookie)
{
struct stream_out *out = (struct stream_out *)stream;
ALOGV("%s", __func__);
lock_output_stream(out);
out->offload_callback = callback;
out->offload_cookie = cookie;
pthread_mutex_unlock(&out->lock);
return 0;
}
static int out_pause(struct audio_stream_out* stream)
{
struct stream_out *out = (struct stream_out *)stream;
int status = -ENOSYS;
ALOGV("%s", __func__);
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
status = -ENODATA;
lock_output_stream(out);
if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) {
status = compress_pause(out->compr);
out->offload_state = OFFLOAD_STATE_PAUSED;
}
pthread_mutex_unlock(&out->lock);
}
return status;
}
static int out_resume(struct audio_stream_out* stream)
{
struct stream_out *out = (struct stream_out *)stream;
int status = -ENOSYS;
ALOGV("%s", __func__);
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
status = -ENODATA;
lock_output_stream(out);
if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) {
status = compress_resume(out->compr);
out->offload_state = OFFLOAD_STATE_PLAYING;
}
pthread_mutex_unlock(&out->lock);
}
return status;
}
static int out_drain(struct audio_stream_out* stream, audio_drain_type_t type )
{
struct stream_out *out = (struct stream_out *)stream;
int status = -ENOSYS;
ALOGV("%s", __func__);
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
lock_output_stream(out);
if (type == AUDIO_DRAIN_EARLY_NOTIFY)
status = send_offload_cmd_l(out, OFFLOAD_CMD_PARTIAL_DRAIN);
else
status = send_offload_cmd_l(out, OFFLOAD_CMD_DRAIN);
pthread_mutex_unlock(&out->lock);
}
return status;
}
static int out_flush(struct audio_stream_out* stream)
{
struct stream_out *out = (struct stream_out *)stream;
ALOGV("%s", __func__);
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
lock_output_stream(out);
stop_compressed_output_l(out);
pthread_mutex_unlock(&out->lock);
return 0;
}
return -ENOSYS;
}
static int out_stop(const struct audio_stream_out* stream)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
int ret = -ENOSYS;
ALOGV("%s", __func__);
pthread_mutex_lock(&adev->lock);
if (out->usecase == USECASE_AUDIO_PLAYBACK_MMAP && !out->standby &&
out->playback_started && out->pcm != NULL) {
pcm_stop(out->pcm);
ret = stop_output_stream(out);
out->playback_started = false;
}
pthread_mutex_unlock(&adev->lock);
return ret;
}
static int out_start(const struct audio_stream_out* stream)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
int ret = -ENOSYS;
ALOGV("%s", __func__);
pthread_mutex_lock(&adev->lock);
if (out->usecase == USECASE_AUDIO_PLAYBACK_MMAP && !out->standby &&
!out->playback_started && out->pcm != NULL) {
ret = start_output_stream(out);
if (ret == 0) {
out->playback_started = true;
}
}
pthread_mutex_unlock(&adev->lock);
return ret;
}
/*
* Modify config->period_count based on min_size_frames
*/
static void adjust_mmap_period_count(struct pcm_config *config, int32_t min_size_frames)
{
int periodCountRequested = (min_size_frames + config->period_size - 1)
/ config->period_size;
int periodCount = MMAP_PERIOD_COUNT_MIN;
ALOGV("%s original config.period_size = %d config.period_count = %d",
__func__, config->period_size, config->period_count);
while (periodCount < periodCountRequested && (periodCount * 2) < MMAP_PERIOD_COUNT_MAX) {
periodCount *= 2;
}
config->period_count = periodCount;
ALOGV("%s requested config.period_count = %d", __func__, config->period_count);
}
// Read offset for the positional timestamp from a persistent vendor property.
// This is to workaround apparent inaccuracies in the timing information that
// is used by the AAudio timing model. The inaccuracies can cause glitches.
static int64_t get_mmap_out_time_offset() {
const int32_t kDefaultOffsetMicros = 0;
int32_t mmap_time_offset_micros = property_get_int32(
"persist.audio.out_mmap_delay_micros", kDefaultOffsetMicros);
ALOGI("mmap_time_offset_micros = %d for output", mmap_time_offset_micros);
return mmap_time_offset_micros * (int64_t)1000;
}
static int out_create_mmap_buffer(const struct audio_stream_out *stream,
int32_t min_size_frames,
struct audio_mmap_buffer_info *info)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
int ret = 0;
unsigned int offset1;
unsigned int frames1;
const char *step = "";
uint32_t mmap_size;
uint32_t buffer_size;
ALOGV("%s", __func__);
lock_output_stream(out);
pthread_mutex_lock(&adev->lock);
if (info == NULL || min_size_frames <= 0 || min_size_frames > MMAP_MIN_SIZE_FRAMES_MAX) {
ALOGE("%s: info = %p, min_size_frames = %d", __func__, info, min_size_frames);
ret = -EINVAL;
goto exit;
}
if (out->usecase != USECASE_AUDIO_PLAYBACK_MMAP || !out->standby) {
ALOGE("%s: usecase = %d, standby = %d", __func__, out->usecase, out->standby);
ret = -ENOSYS;
goto exit;
}
out->pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
if (out->pcm_device_id < 0) {
ALOGE("%s: Invalid PCM device id(%d) for the usecase(%d)",
__func__, out->pcm_device_id, out->usecase);
ret = -EINVAL;
goto exit;
}
adjust_mmap_period_count(&out->config, min_size_frames);
ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d",
__func__, adev->snd_card, out->pcm_device_id, out->config.channels);
out->pcm = pcm_open(adev->snd_card, out->pcm_device_id,
(PCM_OUT | PCM_MMAP | PCM_NOIRQ | PCM_MONOTONIC), &out->config);
if (out->pcm == NULL || !pcm_is_ready(out->pcm)) {
step = "open";
ret = -ENODEV;
goto exit;
}
ret = pcm_mmap_begin(out->pcm, &info->shared_memory_address, &offset1, &frames1);
if (ret < 0) {
step = "begin";
goto exit;
}
info->buffer_size_frames = pcm_get_buffer_size(out->pcm);
buffer_size = pcm_frames_to_bytes(out->pcm, info->buffer_size_frames);
info->burst_size_frames = out->config.period_size;
ret = platform_get_mmap_data_fd(adev->platform,
out->pcm_device_id, 0 /*playback*/,
&info->shared_memory_fd,
&mmap_size);
if (ret < 0) {
// Fall back to non exclusive mode
info->shared_memory_fd = pcm_get_poll_fd(out->pcm);
} else {
out->mmap_shared_memory_fd = info->shared_memory_fd; // for closing later
ALOGV("%s: opened mmap_shared_memory_fd = %d", __func__, out->mmap_shared_memory_fd);
if (mmap_size < buffer_size) {
step = "mmap";
goto exit;
}
// FIXME: indicate exclusive mode support by returning a negative buffer size
info->buffer_size_frames *= -1;
}
memset(info->shared_memory_address, 0, buffer_size);
ret = pcm_mmap_commit(out->pcm, 0, MMAP_PERIOD_SIZE);
if (ret < 0) {
step = "commit";
goto exit;
}
out->mmap_time_offset_nanos = get_mmap_out_time_offset();
out->standby = false;
ret = 0;
ALOGV("%s: got mmap buffer address %p info->buffer_size_frames %d",
__func__, info->shared_memory_address, info->buffer_size_frames);
exit:
if (ret != 0) {
if (out->pcm == NULL) {
ALOGE("%s: %s - %d", __func__, step, ret);
} else {
ALOGE("%s: %s %s", __func__, step, pcm_get_error(out->pcm));
pcm_close(out->pcm);
out->pcm = NULL;
}
}
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&out->lock);
return ret;
}
static int out_get_mmap_position(const struct audio_stream_out *stream,
struct audio_mmap_position *position)
{
int ret = 0;
struct stream_out *out = (struct stream_out *)stream;
ALOGVV("%s", __func__);
if (position == NULL) {
return -EINVAL;
}
lock_output_stream(out);
if (out->usecase != USECASE_AUDIO_PLAYBACK_MMAP ||
out->pcm == NULL) {
ret = -ENOSYS;
goto exit;
}
struct timespec ts = { 0, 0 };
ret = pcm_mmap_get_hw_ptr(out->pcm, (unsigned int *)&position->position_frames, &ts);
if (ret < 0) {
ALOGE("%s: %s", __func__, pcm_get_error(out->pcm));
goto exit;
}
position->time_nanoseconds = audio_utils_ns_from_timespec(&ts)
+ out->mmap_time_offset_nanos;
exit:
pthread_mutex_unlock(&out->lock);
return ret;
}
/** audio_stream_in implementation **/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
return in->config.rate;
}
static int in_set_sample_rate(struct audio_stream *stream __unused, uint32_t rate __unused)
{
return -ENOSYS;
}
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
return in->config.period_size * in->af_period_multiplier *
audio_stream_in_frame_size((const struct audio_stream_in *)stream);
}
static uint32_t in_get_channels(const struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
return in->channel_mask;
}
static audio_format_t in_get_format(const struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
return in->format;
}
static int in_set_format(struct audio_stream *stream __unused, audio_format_t format __unused)
{
return -ENOSYS;
}
static int in_standby(struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
int status = 0;
bool do_stop = true;
ALOGV("%s: enter", __func__);
lock_input_stream(in);
if (!in->standby && (in->flags & AUDIO_INPUT_FLAG_HW_HOTWORD)) {
ALOGV("%s: sound trigger pcm stop lab", __func__);
audio_extn_sound_trigger_stop_lab(in);
in->standby = true;
}
if (!in->standby) {
if (adev->adm_deregister_stream)
adev->adm_deregister_stream(adev->adm_data, in->capture_handle);
pthread_mutex_lock(&adev->lock);
in->standby = true;
if (in->usecase == USECASE_AUDIO_RECORD_MMAP) {
do_stop = in->capture_started;
in->capture_started = false;
if (in->mmap_shared_memory_fd >= 0) {
ALOGV("%s: closing mmap_shared_memory_fd = %d",
__func__, in->mmap_shared_memory_fd);
close(in->mmap_shared_memory_fd);
in->mmap_shared_memory_fd = -1;
}
}
if (in->pcm) {
pcm_close(in->pcm);
in->pcm = NULL;
}
if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION)
adev->enable_voicerx = false;
if (do_stop) {
status = stop_input_stream(in);
}
pthread_mutex_unlock(&adev->lock);
}
pthread_mutex_unlock(&in->lock);
ALOGV("%s: exit: status(%d)", __func__, status);
return status;
}
static int in_dump(const struct audio_stream *stream, int fd)
{
struct stream_in *in = (struct stream_in *)stream;
// We try to get the lock for consistency,
// but it isn't necessary for these variables.
// If we're not in standby, we may be blocked on a read.
const bool locked = (pthread_mutex_trylock(&in->lock) == 0);
dprintf(fd, " Standby: %s\n", in->standby ? "yes" : "no");
dprintf(fd, " Frames read: %lld\n", (long long)in->frames_read);
dprintf(fd, " Frames muted: %lld\n", (long long)in->frames_muted);
char buffer[256]; // for statistics formatting
if (in->start_latency_ms.n > 0) {
simple_stats_to_string(&in->start_latency_ms, buffer, sizeof(buffer));
dprintf(fd, " Start latency ms: %s\n", buffer);
}
if (locked) {
pthread_mutex_unlock(&in->lock);
}
// dump error info
(void)error_log_dump(
in->error_log, fd, " " /* prefix */, 0 /* lines */, 0 /* limit_ns */);
return 0;
}
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
struct str_parms *parms;
char *str;
char value[32];
int ret, val = 0;
int status = 0;
ALOGV("%s: enter: kvpairs=%s", __func__, kvpairs);
parms = str_parms_create_str(kvpairs);
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value));
lock_input_stream(in);
pthread_mutex_lock(&adev->lock);
if (ret >= 0) {
val = atoi(value);
/* no audio source uses val == 0 */
if ((in->source != val) && (val != 0)) {
in->source = val;
}
}
ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
if (((int)in->device != val) && (val != 0) && audio_is_input_device(val) ) {
// Workaround: If routing to an non existing usb device, fail gracefully
// The routing request will otherwise block during 10 second
int card;
if (audio_is_usb_in_device(val) &&
(card = get_alive_usb_card(parms)) >= 0) {
ALOGW("in_set_parameters() ignoring rerouting to non existing USB card %d", card);
status = -ENOSYS;
} else {
in->device = val;
/* If recording is in progress, change the tx device to new device */
if (!in->standby) {
ALOGV("update input routing change");
// inform adm before actual routing to prevent glitches.
if (adev->adm_on_routing_change) {
adev->adm_on_routing_change(adev->adm_data,
in->capture_handle);
}
select_devices(adev, in->usecase);
}
}
}
}
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&in->lock);
str_parms_destroy(parms);
ALOGV("%s: exit: status(%d)", __func__, status);
return status;
}
static char* in_get_parameters(const struct audio_stream *stream,
const char *keys)
{
struct stream_in *in = (struct stream_in *)stream;
struct str_parms *query = str_parms_create_str(keys);
char *str;
struct str_parms *reply = str_parms_create();
bool replied = false;
ALOGV("%s: enter: keys - %s", __func__, keys);
replied |= stream_get_parameter_channels(query, reply,
&in->supported_channel_masks[0]);
replied |= stream_get_parameter_formats(query, reply,
&in->supported_formats[0]);
replied |= stream_get_parameter_rates(query, reply,
&in->supported_sample_rates[0]);
if (replied) {
str = str_parms_to_str(reply);
} else {
str = strdup("");
}
str_parms_destroy(query);
str_parms_destroy(reply);
ALOGV("%s: exit: returns - %s", __func__, str);
return str;
}
static int in_set_gain(struct audio_stream_in *stream, float gain)
{
struct stream_in *in = (struct stream_in *)stream;
char mixer_ctl_name[128];
struct mixer_ctl *ctl;
int ctl_value;
ALOGV("%s: gain %f", __func__, gain);
if (stream == NULL)
return -EINVAL;
/* in_set_gain() only used to silence MMAP capture for now */
if (in->usecase != USECASE_AUDIO_RECORD_MMAP)
return -ENOSYS;
snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "Capture %d Volume", in->pcm_device_id);
ctl = mixer_get_ctl_by_name(in->dev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGW("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
return -ENOSYS;
}
if (gain < RECORD_GAIN_MIN)
gain = RECORD_GAIN_MIN;
else if (gain > RECORD_GAIN_MAX)
gain = RECORD_GAIN_MAX;
ctl_value = (int)(RECORD_VOLUME_CTL_MAX * gain);
mixer_ctl_set_value(ctl, 0, ctl_value);
return 0;
}
static void in_snd_mon_cb(void * stream, struct str_parms * parms)
{
if (!stream || !parms)
return;
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
card_status_t status;
int card;
if (parse_snd_card_status(parms, &card, &status) < 0)
return;
pthread_mutex_lock(&adev->lock);
bool valid_cb = (card == adev->snd_card);
pthread_mutex_unlock(&adev->lock);
if (!valid_cb)
return;
lock_input_stream(in);
if (in->card_status != status)
in->card_status = status;
pthread_mutex_unlock(&in->lock);
ALOGW("in_snd_mon_cb for card %d usecase %s, status %s", card,
use_case_table[in->usecase],
status == CARD_STATUS_OFFLINE ? "offline" : "online");
// a better solution would be to report error back to AF and let
// it put the stream to standby
if (status == CARD_STATUS_OFFLINE)
in_standby(&in->stream.common);
return;
}
static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
size_t bytes)
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
int i, ret = -1;
int *int_buf_stream = NULL;
int error_code = ERROR_CODE_STANDBY; // initial errors are considered coming out of standby.
lock_input_stream(in);
const size_t frame_size = audio_stream_in_frame_size(stream);
const size_t frames = bytes / frame_size;
if (in->flags & AUDIO_INPUT_FLAG_HW_HOTWORD) {
ALOGVV(" %s: reading on st session bytes=%zu", __func__, bytes);
/* Read from sound trigger HAL */
audio_extn_sound_trigger_read(in, buffer, bytes);
pthread_mutex_unlock(&in->lock);
return bytes;
}
if (in->usecase == USECASE_AUDIO_RECORD_MMAP) {
ret = -ENOSYS;
goto exit;
}
if (in->standby) {
const int64_t startNs = systemTime(SYSTEM_TIME_MONOTONIC);
pthread_mutex_lock(&adev->lock);
ret = start_input_stream(in);
pthread_mutex_unlock(&adev->lock);
if (ret != 0) {
goto exit;
}
in->standby = 0;
// log startup time in ms.
simple_stats_log(
&in->start_latency_ms, (systemTime(SYSTEM_TIME_MONOTONIC) - startNs) * 1e-6);
}
// errors that occur here are read errors.
error_code = ERROR_CODE_READ;
//what's the duration requested by the client?
long ns = pcm_bytes_to_frames(in->pcm, bytes)*1000000000LL/
in->config.rate;
request_in_focus(in, ns);
bool use_mmap = is_mmap_usecase(in->usecase) || in->realtime;
if (in->pcm) {
if (use_mmap) {
ret = pcm_mmap_read(in->pcm, buffer, bytes);
} else {
ret = pcm_read(in->pcm, buffer, bytes);
}
if (ret < 0) {
ALOGE("Failed to read w/err %s", strerror(errno));
ret = -errno;
}
if (!ret && bytes > 0 && (in->format == AUDIO_FORMAT_PCM_8_24_BIT)) {
if (bytes % 4 == 0) {
/* data from DSP comes in 24_8 format, convert it to 8_24 */
int_buf_stream = buffer;
for (size_t itt=0; itt < bytes/4 ; itt++) {
int_buf_stream[itt] >>= 8;
}
} else {
ALOGE("%s: !!! something wrong !!! ... data not 32 bit aligned ", __func__);
ret = -EINVAL;
goto exit;
}
}
}
release_in_focus(in, ns);
/*
* Instead of writing zeroes here, we could trust the hardware
* to always provide zeroes when muted.
* No need to acquire adev->lock to read mic_muted here as we don't change its state.
*/
if (ret == 0 && adev->mic_muted &&
!voice_is_in_call_rec_stream(in) &&
in->usecase != USECASE_AUDIO_RECORD_AFE_PROXY) {
memset(buffer, 0, bytes);
in->frames_muted += frames;
}
exit:
pthread_mutex_unlock(&in->lock);
if (ret != 0) {
error_log_log(in->error_log, error_code, audio_utils_get_real_time_ns());
in_standby(&in->stream.common);
ALOGV("%s: read failed - sleeping for buffer duration", __func__);
usleep(frames * 1000000LL / in_get_sample_rate(&in->stream.common));
memset(buffer, 0, bytes); // clear return data
in->frames_muted += frames;
}
if (bytes > 0) {
in->frames_read += frames;
}
return bytes;
}
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream __unused)
{
return 0;
}
static int in_get_capture_position(const struct audio_stream_in *stream,
int64_t *frames, int64_t *time)
{
if (stream == NULL || frames == NULL || time == NULL) {
return -EINVAL;
}
struct stream_in *in = (struct stream_in *)stream;
int ret = -ENOSYS;
lock_input_stream(in);
// note: ST sessions do not close the alsa pcm driver synchronously
// on standby. Therefore, we may return an error even though the
// pcm stream is still opened.
if (in->standby) {
ALOGE_IF(in->pcm != NULL && !(in->flags & AUDIO_INPUT_FLAG_HW_HOTWORD),
"%s stream in standby but pcm not NULL for non ST session", __func__);
goto exit;
}
if (in->pcm) {
struct timespec timestamp;
unsigned int avail;
if (pcm_get_htimestamp(in->pcm, &avail, &timestamp) == 0) {
*frames = in->frames_read + avail;
*time = timestamp.tv_sec * 1000000000LL + timestamp.tv_nsec
- platform_capture_latency(in) * 1000LL;
ret = 0;
}
}
exit:
pthread_mutex_unlock(&in->lock);
return ret;
}
static int in_update_effect_list(bool add, effect_handle_t effect,
struct listnode *head)
{
struct listnode *node;
struct in_effect_list *elist = NULL;
struct in_effect_list *target = NULL;
int ret = 0;
if (!head)
return ret;
list_for_each(node, head) {
elist = node_to_item(node, struct in_effect_list, list);
if (elist->handle == effect) {
target = elist;
break;
}
}
if (add) {
if (target) {
ALOGD("effect %p already exist", effect);
return ret;
}
target = (struct in_effect_list *)
calloc(1, sizeof(struct in_effect_list));
if (!target) {
ALOGE("%s:fail to allocate memory", __func__);
return -ENOMEM;
}
target->handle = effect;
list_add_tail(head, &target->list);
} else {
if (target) {
list_remove(&target->list);
free(target);
}
}
return ret;
}
static int add_remove_audio_effect(const struct audio_stream *stream,
effect_handle_t effect,
bool enable)
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
int status = 0;
effect_descriptor_t desc;
status = (*effect)->get_descriptor(effect, &desc);
ALOGV("%s: status %d in->standby %d enable:%d", __func__, status, in->standby, enable);
if (status != 0)
return status;
lock_input_stream(in);
pthread_mutex_lock(&in->dev->lock);
if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
in->source == AUDIO_SOURCE_VOICE_RECOGNITION ||
adev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
(memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0)) {
in_update_effect_list(enable, effect, &in->aec_list);
enable = !list_empty(&in->aec_list);
if (enable == in->enable_aec)
goto exit;
in->enable_aec = enable;
ALOGD("AEC enable %d", enable);
if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
adev->mode == AUDIO_MODE_IN_COMMUNICATION) {
adev->enable_voicerx = enable;
struct audio_usecase *usecase;
struct listnode *node;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_PLAYBACK)
select_devices(adev, usecase->id);
}
}
if (!in->standby
&& enable_disable_effect(in->dev, in, EFFECT_AEC, enable) == -ENOSYS)
select_devices(in->dev, in->usecase);
}
if (memcmp(&desc.type, FX_IID_NS, sizeof(effect_uuid_t)) == 0) {
in_update_effect_list(enable, effect, &in->ns_list);
enable = !list_empty(&in->ns_list);
if (enable == in->enable_ns)
goto exit;
in->enable_ns = enable;
ALOGD("NS enable %d", enable);
if (!in->standby) {
if (in->source != AUDIO_SOURCE_VOICE_COMMUNICATION
|| enable_disable_effect(in->dev, in, EFFECT_NS, enable) == -ENOSYS)
select_devices(in->dev, in->usecase);
}
}
exit:
pthread_mutex_unlock(&in->dev->lock);
pthread_mutex_unlock(&in->lock);
return 0;
}
static int in_add_audio_effect(const struct audio_stream *stream,
effect_handle_t effect)
{
ALOGV("%s: effect %p", __func__, effect);
return add_remove_audio_effect(stream, effect, true);
}
static int in_remove_audio_effect(const struct audio_stream *stream,
effect_handle_t effect)
{
ALOGV("%s: effect %p", __func__, effect);
return add_remove_audio_effect(stream, effect, false);
}
static int in_stop(const struct audio_stream_in* stream)
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
int ret = -ENOSYS;
ALOGV("%s", __func__);
pthread_mutex_lock(&adev->lock);
if (in->usecase == USECASE_AUDIO_RECORD_MMAP && !in->standby &&
in->capture_started && in->pcm != NULL) {
pcm_stop(in->pcm);
ret = stop_input_stream(in);
in->capture_started = false;
}
pthread_mutex_unlock(&adev->lock);
return ret;
}
static int in_start(const struct audio_stream_in* stream)
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
int ret = -ENOSYS;
ALOGV("%s in %p", __func__, in);
pthread_mutex_lock(&adev->lock);
if (in->usecase == USECASE_AUDIO_RECORD_MMAP && !in->standby &&
!in->capture_started && in->pcm != NULL) {
if (!in->capture_started) {
ret = start_input_stream(in);
if (ret == 0) {
in->capture_started = true;
}
}
}
pthread_mutex_unlock(&adev->lock);
return ret;
}
// Read offset for the positional timestamp from a persistent vendor property.
// This is to workaround apparent inaccuracies in the timing information that
// is used by the AAudio timing model. The inaccuracies can cause glitches.
static int64_t in_get_mmap_time_offset() {
const int32_t kDefaultOffsetMicros = 0;
int32_t mmap_time_offset_micros = property_get_int32(
"persist.audio.in_mmap_delay_micros", kDefaultOffsetMicros);
ALOGI("in_get_mmap_time_offset set to %d micros", mmap_time_offset_micros);
return mmap_time_offset_micros * (int64_t)1000;
}
static int in_create_mmap_buffer(const struct audio_stream_in *stream,
int32_t min_size_frames,
struct audio_mmap_buffer_info *info)
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
int ret = 0;
unsigned int offset1;
unsigned int frames1;
const char *step = "";
uint32_t mmap_size;
uint32_t buffer_size;
lock_input_stream(in);
pthread_mutex_lock(&adev->lock);
ALOGV("%s in %p", __func__, in);
if (info == NULL || min_size_frames <= 0 || min_size_frames > MMAP_MIN_SIZE_FRAMES_MAX) {
ALOGE("%s invalid argument info %p min_size_frames %d", __func__, info, min_size_frames);
ret = -EINVAL;
goto exit;
}
if (in->usecase != USECASE_AUDIO_RECORD_MMAP || !in->standby) {
ALOGE("%s: usecase = %d, standby = %d", __func__, in->usecase, in->standby);
ALOGV("%s in %p", __func__, in);
ret = -ENOSYS;
goto exit;
}
in->pcm_device_id = platform_get_pcm_device_id(in->usecase, PCM_CAPTURE);
if (in->pcm_device_id < 0) {
ALOGE("%s: Invalid PCM device id(%d) for the usecase(%d)",
__func__, in->pcm_device_id, in->usecase);
ret = -EINVAL;
goto exit;
}
adjust_mmap_period_count(&in->config, min_size_frames);
ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d",
__func__, adev->snd_card, in->pcm_device_id, in->config.channels);
in->pcm = pcm_open(adev->snd_card, in->pcm_device_id,
(PCM_IN | PCM_MMAP | PCM_NOIRQ | PCM_MONOTONIC), &in->config);
if (in->pcm == NULL || !pcm_is_ready(in->pcm)) {
step = "open";
ret = -ENODEV;
goto exit;
}
ret = pcm_mmap_begin(in->pcm, &info->shared_memory_address, &offset1, &frames1);
if (ret < 0) {
step = "begin";
goto exit;
}
info->buffer_size_frames = pcm_get_buffer_size(in->pcm);
buffer_size = pcm_frames_to_bytes(in->pcm, info->buffer_size_frames);
info->burst_size_frames = in->config.period_size;
ret = platform_get_mmap_data_fd(adev->platform,
in->pcm_device_id, 1 /*capture*/,
&info->shared_memory_fd,
&mmap_size);
if (ret < 0) {
// Fall back to non exclusive mode
info->shared_memory_fd = pcm_get_poll_fd(in->pcm);
} else {
in->mmap_shared_memory_fd = info->shared_memory_fd; // for closing later
ALOGV("%s: opened mmap_shared_memory_fd = %d", __func__, in->mmap_shared_memory_fd);
if (mmap_size < buffer_size) {
step = "mmap";
goto exit;
}
// FIXME: indicate exclusive mode support by returning a negative buffer size
info->buffer_size_frames *= -1;
}
memset(info->shared_memory_address, 0, buffer_size);
ret = pcm_mmap_commit(in->pcm, 0, MMAP_PERIOD_SIZE);
if (ret < 0) {
step = "commit";
goto exit;
}
in->mmap_time_offset_nanos = in_get_mmap_time_offset();
in->standby = false;
ret = 0;
ALOGV("%s: got mmap buffer address %p info->buffer_size_frames %d",
__func__, info->shared_memory_address, info->buffer_size_frames);
exit:
if (ret != 0) {
if (in->pcm == NULL) {
ALOGE("%s: %s - %d", __func__, step, ret);
} else {
ALOGE("%s: %s %s", __func__, step, pcm_get_error(in->pcm));
pcm_close(in->pcm);
in->pcm = NULL;
}
}
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&in->lock);
return ret;
}
static int in_get_mmap_position(const struct audio_stream_in *stream,
struct audio_mmap_position *position)
{
int ret = 0;
struct stream_in *in = (struct stream_in *)stream;
ALOGVV("%s", __func__);
if (position == NULL) {
return -EINVAL;
}
lock_input_stream(in);
if (in->usecase != USECASE_AUDIO_RECORD_MMAP ||
in->pcm == NULL) {
ret = -ENOSYS;
goto exit;
}
struct timespec ts = { 0, 0 };
ret = pcm_mmap_get_hw_ptr(in->pcm, (unsigned int *)&position->position_frames, &ts);
if (ret < 0) {
ALOGE("%s: %s", __func__, pcm_get_error(in->pcm));
goto exit;
}
position->time_nanoseconds = audio_utils_ns_from_timespec(&ts)
+ in->mmap_time_offset_nanos;
exit:
pthread_mutex_unlock(&in->lock);
return ret;
}
static int in_get_active_microphones(const struct audio_stream_in *stream,
struct audio_microphone_characteristic_t *mic_array,
size_t *mic_count) {
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
ALOGVV("%s", __func__);
lock_input_stream(in);
pthread_mutex_lock(&adev->lock);
int ret = platform_get_active_microphones(adev->platform,
audio_channel_count_from_in_mask(in->channel_mask),
in->usecase, mic_array, mic_count);
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&in->lock);
return ret;
}
static int adev_get_microphones(const struct audio_hw_device *dev,
struct audio_microphone_characteristic_t *mic_array,
size_t *mic_count) {
struct audio_device *adev = (struct audio_device *)dev;
ALOGVV("%s", __func__);
pthread_mutex_lock(&adev->lock);
int ret = platform_get_microphones(adev->platform, mic_array, mic_count);
pthread_mutex_unlock(&adev->lock);
return ret;
}
static int in_set_microphone_direction(const struct audio_stream_in *stream,
audio_microphone_direction_t dir) {
struct stream_in *in = (struct stream_in *)stream;
ALOGVV("%s: standby %d source %d dir %d", __func__, in->standby, in->source, dir);
in->direction = dir;
if (in->standby)
return 0;
return audio_extn_audiozoom_set_microphone_direction(in, dir);
}
static int in_set_microphone_field_dimension(const struct audio_stream_in *stream, float zoom) {
struct stream_in *in = (struct stream_in *)stream;
ALOGVV("%s: standby %d source %d zoom %f", __func__, in->standby, in->source, zoom);
if (zoom > 1.0 || zoom < -1.0)
return -EINVAL;
in->zoom = zoom;
if (in->standby)
return 0;
return audio_extn_audiozoom_set_microphone_field_dimension(in, zoom);
}
static void in_update_sink_metadata(struct audio_stream_in *stream,
const struct sink_metadata *sink_metadata) {
if (stream == NULL
|| sink_metadata == NULL
|| sink_metadata->tracks == NULL) {
return;
}
int error = 0;
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
audio_devices_t device = AUDIO_DEVICE_NONE;
if (sink_metadata->track_count != 0)
device = sink_metadata->tracks->dest_device;
lock_input_stream(in);
pthread_mutex_lock(&adev->lock);
ALOGV("%s: in->usecase: %d, device: %x", __func__, in->usecase, device);
if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY
&& device != AUDIO_DEVICE_NONE
&& adev->voice_tx_output != NULL) {
/* Use the rx device from afe-proxy record to route voice call because
there is no routing if tx device is on primary hal and rx device
is on other hal during voice call. */
adev->voice_tx_output->devices = device;
if (!voice_is_call_state_active(adev)) {
if (adev->mode == AUDIO_MODE_IN_CALL) {
adev->current_call_output = adev->voice_tx_output;
error = voice_start_call(adev);
if (error != 0)
ALOGE("%s: start voice call failed %d", __func__, error);
}
} else {
adev->current_call_output = adev->voice_tx_output;
voice_update_devices_for_all_voice_usecases(adev);
}
}
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&in->lock);
}
static int check_and_set_gapless_mode(struct audio_device *adev)
{
bool gapless_enabled = false;
const char *mixer_ctl_name = "Compress Gapless Playback";
struct mixer_ctl *ctl;
ALOGV("%s:", __func__);
gapless_enabled = property_get_bool("vendor.audio.offload.gapless.enabled", false);
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
return -EINVAL;
}
if (mixer_ctl_set_value(ctl, 0, gapless_enabled) < 0) {
ALOGE("%s: Could not set gapless mode %d",
__func__, gapless_enabled);
return -EINVAL;
}
return 0;
}
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,
const char *address __unused)
{
struct audio_device *adev = (struct audio_device *)dev;
struct stream_out *out;
int i, ret = 0;
bool is_hdmi = devices & AUDIO_DEVICE_OUT_AUX_DIGITAL;
bool is_usb_dev = audio_is_usb_out_device(devices) &&
(devices != AUDIO_DEVICE_OUT_USB_ACCESSORY);
bool force_haptic_path =
property_get_bool("vendor.audio.test_haptic", false);
if (is_usb_dev && !is_usb_ready(adev, true /* is_playback */)) {
return -ENOSYS;
}
ALOGV("%s: enter: format(%#x) sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)",
__func__, config->format, config->sample_rate, config->channel_mask, devices, flags);
*stream_out = NULL;
out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
pthread_mutex_init(&out->compr_mute_lock, (const pthread_mutexattr_t *) NULL);
if (devices == AUDIO_DEVICE_NONE)
devices = AUDIO_DEVICE_OUT_SPEAKER;
out->flags = flags;
out->devices = devices;
out->dev = adev;
out->handle = handle;
out->a2dp_compress_mute = false;
out->mmap_shared_memory_fd = -1; // not open
/* Init use case and pcm_config */
if ((is_hdmi || is_usb_dev) &&
(audio_is_linear_pcm(config->format) || config->format == AUDIO_FORMAT_DEFAULT) &&
(flags == AUDIO_OUTPUT_FLAG_NONE ||
(flags & AUDIO_OUTPUT_FLAG_DIRECT) != 0)) {
audio_format_t req_format = config->format;
audio_channel_mask_t req_channel_mask = config->channel_mask;
uint32_t req_sample_rate = config->sample_rate;
pthread_mutex_lock(&adev->lock);
if (is_hdmi) {
ret = read_hdmi_channel_masks(out);
if (config->sample_rate == 0)
config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
if (config->channel_mask == AUDIO_CHANNEL_NONE)
config->channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
if (config->format == AUDIO_FORMAT_DEFAULT)
config->format = AUDIO_FORMAT_PCM_16_BIT;
} else if (is_usb_dev) {
ret = read_usb_sup_params_and_compare(true /*is_playback*/,
&config->format,
&out->supported_formats[0],
MAX_SUPPORTED_FORMATS,
&config->channel_mask,
&out->supported_channel_masks[0],
MAX_SUPPORTED_CHANNEL_MASKS,
&config->sample_rate,
&out->supported_sample_rates[0],
MAX_SUPPORTED_SAMPLE_RATES);
ALOGV("plugged dev USB ret %d", ret);
}
pthread_mutex_unlock(&adev->lock);
if (ret != 0) {
// For MMAP NO IRQ, allow conversions in ADSP
if (is_hdmi || (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) == 0)
goto error_open;
if (req_sample_rate != 0 && config->sample_rate != req_sample_rate)
config->sample_rate = req_sample_rate;
if (req_channel_mask != AUDIO_CHANNEL_NONE && config->channel_mask != req_channel_mask)
config->channel_mask = req_channel_mask;
if (req_format != AUDIO_FORMAT_DEFAULT && config->format != req_format)
config->format = req_format;
}
out->sample_rate = config->sample_rate;
out->channel_mask = config->channel_mask;
out->format = config->format;
if (is_hdmi) {
out->usecase = USECASE_AUDIO_PLAYBACK_HIFI;
out->config = pcm_config_hdmi_multi;
} else if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
out->usecase = USECASE_AUDIO_PLAYBACK_MMAP;
out->config = pcm_config_mmap_playback;
out->stream.start = out_start;
out->stream.stop = out_stop;
out->stream.create_mmap_buffer = out_create_mmap_buffer;
out->stream.get_mmap_position = out_get_mmap_position;
} else {
out->usecase = USECASE_AUDIO_PLAYBACK_HIFI;
out->config = pcm_config_hifi;
}
out->config.rate = out->sample_rate;
out->config.channels = audio_channel_count_from_out_mask(out->channel_mask);
if (is_hdmi) {
out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels *
audio_bytes_per_sample(out->format));
}
out->config.format = pcm_format_from_audio_format(out->format);
} else if (flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
pthread_mutex_lock(&adev->lock);
bool offline = (adev->card_status == CARD_STATUS_OFFLINE);
pthread_mutex_unlock(&adev->lock);
// reject offload during card offline to allow
// fallback to s/w paths
if (offline) {
ret = -ENODEV;
goto error_open;
}
if (config->offload_info.version != AUDIO_INFO_INITIALIZER.version ||
config->offload_info.size != AUDIO_INFO_INITIALIZER.size) {
ALOGE("%s: Unsupported Offload information", __func__);
ret = -EINVAL;
goto error_open;
}
if (!is_supported_format(config->offload_info.format)) {
ALOGE("%s: Unsupported audio format", __func__);
ret = -EINVAL;
goto error_open;
}
out->sample_rate = config->offload_info.sample_rate;
if (config->offload_info.channel_mask != AUDIO_CHANNEL_NONE)
out->channel_mask = config->offload_info.channel_mask;
else if (config->channel_mask != AUDIO_CHANNEL_NONE)
out->channel_mask = config->channel_mask;
else
out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
out->format = config->offload_info.format;
out->compr_config.codec = (struct snd_codec *)
calloc(1, sizeof(struct snd_codec));
out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD;
out->stream.set_callback = out_set_callback;
out->stream.pause = out_pause;
out->stream.resume = out_resume;
out->stream.drain = out_drain;
out->stream.flush = out_flush;
out->compr_config.codec->id =
get_snd_codec_id(config->offload_info.format);
out->compr_config.fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE;
out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
out->compr_config.codec->sample_rate = out->sample_rate;
out->compr_config.codec->bit_rate =
config->offload_info.bit_rate;
out->compr_config.codec->ch_in =
audio_channel_count_from_out_mask(out->channel_mask);
out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
out->non_blocking = 1;
out->send_new_metadata = 1;
check_and_set_gapless_mode(adev);
create_offload_callback_thread(out);
ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
__func__, config->offload_info.version,
config->offload_info.bit_rate);
} else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
switch (config->sample_rate) {
case 0:
out->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
break;
case 8000:
case 16000:
case 48000:
out->sample_rate = config->sample_rate;
break;
default:
ALOGE("%s: Unsupported sampling rate %d for Incall Music", __func__,
config->sample_rate);
config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
ret = -EINVAL;
goto error_open;
}
//FIXME: add support for MONO stream configuration when audioflinger mixer supports it
switch (config->channel_mask) {
case AUDIO_CHANNEL_NONE:
case AUDIO_CHANNEL_OUT_STEREO:
out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
break;
default:
ALOGE("%s: Unsupported channel mask %#x for Incall Music", __func__,
config->channel_mask);
config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
ret = -EINVAL;
goto error_open;
}
switch (config->format) {
case AUDIO_FORMAT_DEFAULT:
case AUDIO_FORMAT_PCM_16_BIT:
out->format = AUDIO_FORMAT_PCM_16_BIT;
break;
default:
ALOGE("%s: Unsupported format %#x for Incall Music", __func__,
config->format);
config->format = AUDIO_FORMAT_PCM_16_BIT;
ret = -EINVAL;
goto error_open;
}
voice_extn_check_and_set_incall_music_usecase(adev, out);
} else if (out->devices == AUDIO_DEVICE_OUT_TELEPHONY_TX) {
switch (config->sample_rate) {
case 0:
out->sample_rate = AFE_PROXY_SAMPLING_RATE;
break;
case 8000:
case 16000:
case 48000:
out->sample_rate = config->sample_rate;
break;
default:
ALOGE("%s: Unsupported sampling rate %d for Telephony TX", __func__,
config->sample_rate);
config->sample_rate = AFE_PROXY_SAMPLING_RATE;
ret = -EINVAL;
break;
}
//FIXME: add support for MONO stream configuration when audioflinger mixer supports it
switch (config->channel_mask) {
case AUDIO_CHANNEL_NONE:
out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
break;
case AUDIO_CHANNEL_OUT_STEREO:
out->channel_mask = config->channel_mask;
break;
default:
ALOGE("%s: Unsupported channel mask %#x for Telephony TX", __func__,
config->channel_mask);
config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
ret = -EINVAL;
break;
}
switch (config->format) {
case AUDIO_FORMAT_DEFAULT:
out->format = AUDIO_FORMAT_PCM_16_BIT;
break;
case AUDIO_FORMAT_PCM_16_BIT:
out->format = config->format;
break;
default:
ALOGE("%s: Unsupported format %#x for Telephony TX", __func__,
config->format);
config->format = AUDIO_FORMAT_PCM_16_BIT;
ret = -EINVAL;
break;
}
if (ret != 0)
goto error_open;
out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY;
out->config = pcm_config_afe_proxy_playback;
out->config.rate = out->sample_rate;
out->config.channels =
audio_channel_count_from_out_mask(out->channel_mask);
out->config.format = pcm_format_from_audio_format(out->format);
adev->voice_tx_output = out;
} else if (flags == AUDIO_OUTPUT_FLAG_VOIP_RX) {
switch (config->sample_rate) {
case 0:
out->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
break;
case 8000:
case 16000:
case 32000:
case 48000:
out->sample_rate = config->sample_rate;
break;
default:
ALOGE("%s: Unsupported sampling rate %d for Voip RX", __func__,
config->sample_rate);
config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
ret = -EINVAL;
break;
}
//FIXME: add support for MONO stream configuration when audioflinger mixer supports it
switch (config->channel_mask) {
case AUDIO_CHANNEL_NONE:
out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
break;
case AUDIO_CHANNEL_OUT_STEREO:
out->channel_mask = config->channel_mask;
break;
default:
ALOGE("%s: Unsupported channel mask %#x for Voip RX", __func__,
config->channel_mask);
config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
ret = -EINVAL;
break;
}
switch (config->format) {
case AUDIO_FORMAT_DEFAULT:
out->format = AUDIO_FORMAT_PCM_16_BIT;
break;
case AUDIO_FORMAT_PCM_16_BIT:
out->format = config->format;
break;
default:
ALOGE("%s: Unsupported format %#x for Voip RX", __func__,
config->format);
config->format = AUDIO_FORMAT_PCM_16_BIT;
ret = -EINVAL;
break;
}
if (ret != 0)
goto error_open;
uint32_t buffer_size, frame_size;
out->usecase = USECASE_AUDIO_PLAYBACK_VOIP;
out->config = pcm_config_voip;
out->config.rate = out->sample_rate;
out->config.format = pcm_format_from_audio_format(out->format);
buffer_size = get_stream_buffer_size(VOIP_PLAYBACK_PERIOD_DURATION_MSEC,
out->sample_rate,
out->format,
out->config.channels,
false /*is_low_latency*/);
frame_size = audio_bytes_per_sample(out->format) * out->config.channels;
out->config.period_size = buffer_size / frame_size;
out->config.period_count = VOIP_PLAYBACK_PERIOD_COUNT;
out->af_period_multiplier = 1;
} else {
if (flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
out->config = pcm_config_deep_buffer;
} else if (flags & AUDIO_OUTPUT_FLAG_TTS) {
out->usecase = USECASE_AUDIO_PLAYBACK_TTS;
out->config = pcm_config_deep_buffer;
} else if (flags & AUDIO_OUTPUT_FLAG_RAW) {
out->usecase = USECASE_AUDIO_PLAYBACK_ULL;
out->realtime = may_use_noirq_mode(adev, USECASE_AUDIO_PLAYBACK_ULL, out->flags);
out->config = out->realtime ? pcm_config_rt : pcm_config_low_latency;
} else if (flags & AUDIO_OUTPUT_FLAG_MMAP_NOIRQ) {
out->usecase = USECASE_AUDIO_PLAYBACK_MMAP;
out->config = pcm_config_mmap_playback;
out->stream.start = out_start;
out->stream.stop = out_stop;
out->stream.create_mmap_buffer = out_create_mmap_buffer;
out->stream.get_mmap_position = out_get_mmap_position;
} else {
if (config->channel_mask & AUDIO_CHANNEL_HAPTIC_ALL) {
out->usecase = USECASE_AUDIO_PLAYBACK_WITH_HAPTICS;
adev->haptic_pcm_device_id = platform_get_haptics_pcm_device_id();
if (adev->haptic_pcm_device_id < 0) {
ALOGE("%s: Invalid Haptics pcm device id(%d) for the usecase(%d)",
__func__, adev->haptic_pcm_device_id, out->usecase);
ret = -ENOSYS;
goto error_open;
}
out->config = pcm_config_haptics_audio;
if (force_haptic_path)
adev->haptics_config = pcm_config_haptics_audio;
else
adev->haptics_config = pcm_config_haptics;
} else {
out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
out->config = pcm_config_low_latency;
}
}
if (out->usecase == USECASE_AUDIO_PLAYBACK_LOW_LATENCY ||
out->usecase == USECASE_AUDIO_PLAYBACK_DEEP_BUFFER ||
out->usecase == USECASE_AUDIO_PLAYBACK_ULL) {
out->volume_l = 1.0;
out->volume_r = 1.0;
}
if (config->sample_rate == 0) {
out->sample_rate = out->config.rate;
} else {
out->sample_rate = config->sample_rate;
}
if (config->channel_mask == AUDIO_CHANNEL_NONE) {
out->channel_mask = audio_channel_out_mask_from_count(out->config.channels);
} else {
out->channel_mask = config->channel_mask;
}
if (config->format == AUDIO_FORMAT_DEFAULT)
out->format = audio_format_from_pcm_format(out->config.format);
else if (!audio_is_linear_pcm(config->format)) {
config->format = AUDIO_FORMAT_PCM_16_BIT;
ret = -EINVAL;
goto error_open;
} else {
out->format = config->format;
}
out->config.rate = out->sample_rate;
if (config->channel_mask & AUDIO_CHANNEL_HAPTIC_ALL) {
out->config.channels =
audio_channel_count_from_out_mask(out->channel_mask &
~AUDIO_CHANNEL_HAPTIC_ALL);
if (force_haptic_path) {
out->config.channels = 1;
adev->haptics_config.channels = 1;
} else {
adev->haptics_config.channels =
audio_channel_count_from_out_mask(out->channel_mask &
AUDIO_CHANNEL_HAPTIC_ALL);
}
} else {
out->config.channels =
audio_channel_count_from_out_mask(out->channel_mask);
}
if (out->format != audio_format_from_pcm_format(out->config.format)) {
out->config.format = pcm_format_from_audio_format(out->format);
}
}
if ((config->sample_rate != 0 && config->sample_rate != out->sample_rate) ||
(config->format != AUDIO_FORMAT_DEFAULT && config->format != out->format) ||
(config->channel_mask != AUDIO_CHANNEL_NONE && config->channel_mask != out->channel_mask)) {
ALOGI("%s: Unsupported output config. sample_rate:%u format:%#x channel_mask:%#x",
__func__, config->sample_rate, config->format, config->channel_mask);
config->sample_rate = out->sample_rate;
config->format = out->format;
config->channel_mask = out->channel_mask;
ret = -EINVAL;
goto error_open;
}
ALOGV("%s: Usecase(%s) config->format %#x out->config.format %#x\n",
__func__, use_case_table[out->usecase], config->format, out->config.format);
if (flags & AUDIO_OUTPUT_FLAG_PRIMARY) {
if (adev->primary_output == NULL)
adev->primary_output = out;
else {
ALOGE("%s: Primary output is already opened", __func__);
ret = -EEXIST;
goto error_open;
}
}
/* Check if this usecase is already existing */
pthread_mutex_lock(&adev->lock);
if (get_usecase_from_list(adev, out->usecase) != NULL) {
ALOGE("%s: Usecase (%d) is already present", __func__, out->usecase);
pthread_mutex_unlock(&adev->lock);
ret = -EEXIST;
goto error_open;
}
pthread_mutex_unlock(&adev->lock);
out->applied_volume_l = INVALID_OUT_VOLUME;
out->applied_volume_r = INVALID_OUT_VOLUME;
out->stream.common.get_sample_rate = out_get_sample_rate;
out->stream.common.set_sample_rate = out_set_sample_rate;
out->stream.common.get_buffer_size = out_get_buffer_size;
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.get_latency = out_get_latency;
out->stream.set_volume = out_set_volume;
#ifdef NO_AUDIO_OUT
out->stream.write = out_write_for_no_output;
#else
out->stream.write = out_write;
#endif
out->stream.get_render_position = out_get_render_position;
out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
out->stream.get_presentation_position = out_get_presentation_position;
if (out->realtime)
out->af_period_multiplier = af_period_multiplier;
else
out->af_period_multiplier = 1;
out->kernel_buffer_size = out->config.period_size * out->config.period_count;
out->standby = 1;
/* out->muted = false; by calloc() */
/* out->written = 0; by calloc() */
pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL);
pthread_mutex_init(&out->pre_lock, (const pthread_mutexattr_t *) NULL);
pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL);
config->format = out->stream.common.get_format(&out->stream.common);
config->channel_mask = out->stream.common.get_channels(&out->stream.common);
config->sample_rate = out->stream.common.get_sample_rate(&out->stream.common);
register_format(out->format, out->supported_formats);
register_channel_mask(out->channel_mask, out->supported_channel_masks);
register_sample_rate(out->sample_rate, out->supported_sample_rates);
out->error_log = error_log_create(
ERROR_LOG_ENTRIES,
1000000000 /* aggregate consecutive identical errors within one second in ns */);
/*
By locking output stream before registering, we allow the callback
to update stream's state only after stream's initial state is set to
adev state.
*/
lock_output_stream(out);
audio_extn_snd_mon_register_listener(out, out_snd_mon_cb);
pthread_mutex_lock(&adev->lock);
out->card_status = adev->card_status;
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&out->lock);
stream_app_type_cfg_init(&out->app_type_cfg);
*stream_out = &out->stream;
ALOGV("%s: exit", __func__);
return 0;
error_open:
free(out);
*stream_out = NULL;
ALOGW("%s: exit: ret %d", __func__, ret);
return ret;
}
static void adev_close_output_stream(struct audio_hw_device *dev __unused,
struct audio_stream_out *stream)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
ALOGV("%s: enter", __func__);
// must deregister from sndmonitor first to prevent races
// between the callback and close_stream
audio_extn_snd_mon_unregister_listener(out);
out_standby(&stream->common);
if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
destroy_offload_callback_thread(out);
if (out->compr_config.codec != NULL)
free(out->compr_config.codec);
}
out->a2dp_compress_mute = false;
if (adev->voice_tx_output == out)
adev->voice_tx_output = NULL;
error_log_destroy(out->error_log);
out->error_log = NULL;
pthread_cond_destroy(&out->cond);
pthread_mutex_destroy(&out->pre_lock);
pthread_mutex_destroy(&out->lock);
free(stream);
ALOGV("%s: exit", __func__);
}
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
struct audio_device *adev = (struct audio_device *)dev;
struct str_parms *parms;
char *str;
char value[32];
int val;
int ret;
int status = 0;
bool a2dp_reconfig = false;
ALOGV("%s: enter: %s", __func__, kvpairs);
pthread_mutex_lock(&adev->lock);
parms = str_parms_create_str(kvpairs);
status = voice_set_parameters(adev, parms);
if (status != 0) {
goto done;
}
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value));
if (ret >= 0) {
/* When set to false, HAL should disable EC and NS */
if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
adev->bluetooth_nrec = true;
else
adev->bluetooth_nrec = false;
}
ret = str_parms_get_str(parms, "screen_state", value, sizeof(value));
if (ret >= 0) {
if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
adev->screen_off = false;
else
adev->screen_off = true;
}
ret = str_parms_get_int(parms, "rotation", &val);
if (ret >= 0) {
bool reverse_speakers = false;
int camera_rotation = CAMERA_ROTATION_LANDSCAPE;
switch (val) {
// FIXME: note that the code below assumes that the speakers are in the correct placement
// relative to the user when the device is rotated 90deg from its default rotation. This
// assumption is device-specific, not platform-specific like this code.
case 270:
reverse_speakers = true;
camera_rotation = CAMERA_ROTATION_INVERT_LANDSCAPE;
break;
case 0:
case 180:
camera_rotation = CAMERA_ROTATION_PORTRAIT;
break;
case 90:
camera_rotation = CAMERA_ROTATION_LANDSCAPE;
break;
default:
ALOGE("%s: unexpected rotation of %d", __func__, val);
status = -EINVAL;
}
if (status == 0) {
// check and set swap
// - check if orientation changed and speaker active
// - set rotation and cache the rotation value
adev->camera_orientation =
(adev->camera_orientation & ~CAMERA_ROTATION_MASK) | camera_rotation;
#ifndef MAXXAUDIO_QDSP_ENABLED
platform_check_and_set_swap_lr_channels(adev, reverse_speakers);
#endif
}
}
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_SCO_WB, value, sizeof(value));
if (ret >= 0) {
adev->bt_wb_speech_enabled = !strcmp(value, AUDIO_PARAMETER_VALUE_ON);
}
ret = str_parms_get_str(parms, "BT_SCO", value, sizeof(value));
if (ret >= 0) {
if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
adev->bt_sco_on = true;
else
adev->bt_sco_on = false;
}
ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value, sizeof(value));
if (ret >= 0) {
audio_devices_t device = (audio_devices_t)strtoul(value, NULL, 10);
if (audio_is_usb_out_device(device)) {
ret = str_parms_get_str(parms, "card", value, sizeof(value));
if (ret >= 0) {
const int card = atoi(value);
audio_extn_usb_add_device(device, card);
}
} else if (audio_is_usb_in_device(device)) {
ret = str_parms_get_str(parms, "card", value, sizeof(value));
if (ret >= 0) {
const int card = atoi(value);
audio_extn_usb_add_device(device, card);
}
}
}
ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value, sizeof(value));
if (ret >= 0) {
audio_devices_t device = (audio_devices_t)strtoul(value, NULL, 10);
if (audio_is_usb_out_device(device)) {
ret = str_parms_get_str(parms, "card", value, sizeof(value));
if (ret >= 0) {
const int card = atoi(value);
audio_extn_usb_remove_device(device, card);
}
} else if (audio_is_usb_in_device(device)) {
ret = str_parms_get_str(parms, "card", value, sizeof(value));
if (ret >= 0) {
const int card = atoi(value);
audio_extn_usb_remove_device(device, card);
}
}
}
audio_extn_hfp_set_parameters(adev, parms);
audio_extn_ma_set_parameters(adev, parms);
status = audio_extn_a2dp_set_parameters(parms, &a2dp_reconfig);
if (status >= 0 && a2dp_reconfig) {
struct audio_usecase *usecase;
struct listnode *node;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if ((usecase->type == PCM_PLAYBACK) &&
(usecase->devices & AUDIO_DEVICE_OUT_ALL_A2DP)) {
ALOGD("%s: reconfigure A2DP... forcing device switch", __func__);
pthread_mutex_unlock(&adev->lock);
lock_output_stream(usecase->stream.out);
pthread_mutex_lock(&adev->lock);
audio_extn_a2dp_set_handoff_mode(true);
// force device switch to reconfigure encoder
select_devices(adev, usecase->id);
audio_extn_a2dp_set_handoff_mode(false);
pthread_mutex_unlock(&usecase->stream.out->lock);
break;
}
}
}
//FIXME: to be replaced by proper video capture properties API
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_CAMERA_FACING, value, sizeof(value));
if (ret >= 0) {
int camera_facing = CAMERA_FACING_BACK;
if (strcmp(value, AUDIO_PARAMETER_VALUE_FRONT) == 0)
camera_facing = CAMERA_FACING_FRONT;
else if (strcmp(value, AUDIO_PARAMETER_VALUE_BACK) == 0)
camera_facing = CAMERA_FACING_BACK;
else {
ALOGW("%s: invalid camera facing value: %s", __func__, value);
goto done;
}
adev->camera_orientation =
(adev->camera_orientation & ~CAMERA_FACING_MASK) | camera_facing;
struct audio_usecase *usecase;
struct listnode *node;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
struct stream_in *in = usecase->stream.in;
if (usecase->type == PCM_CAPTURE && in != NULL &&
in->source == AUDIO_SOURCE_CAMCORDER && !in->standby) {
select_devices(adev, in->usecase);
}
}
}
done:
str_parms_destroy(parms);
pthread_mutex_unlock(&adev->lock);
ALOGV("%s: exit with code(%d)", __func__, status);
return status;
}
static char* adev_get_parameters(const struct audio_hw_device *dev,
const char *keys)
{
struct audio_device *adev = (struct audio_device *)dev;
struct str_parms *reply = str_parms_create();
struct str_parms *query = str_parms_create_str(keys);
char *str;
pthread_mutex_lock(&adev->lock);
voice_get_parameters(adev, query, reply);
audio_extn_a2dp_get_parameters(query, reply);
str = str_parms_to_str(reply);
str_parms_destroy(query);
str_parms_destroy(reply);
pthread_mutex_unlock(&adev->lock);
ALOGV("%s: exit: returns - %s", __func__, str);
return str;
}
static int adev_init_check(const struct audio_hw_device *dev __unused)
{
return 0;
}
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
int ret;
struct audio_device *adev = (struct audio_device *)dev;
audio_extn_extspk_set_voice_vol(adev->extspk, volume);
pthread_mutex_lock(&adev->lock);
ret = voice_set_volume(adev, volume);
pthread_mutex_unlock(&adev->lock);
return ret;
}
static int adev_set_master_volume(struct audio_hw_device *dev __unused, float volume __unused)
{
return -ENOSYS;
}
static int adev_get_master_volume(struct audio_hw_device *dev __unused,
float *volume __unused)
{
return -ENOSYS;
}
static int adev_set_master_mute(struct audio_hw_device *dev __unused, bool muted __unused)
{
return -ENOSYS;
}
static int adev_get_master_mute(struct audio_hw_device *dev __unused, bool *muted __unused)
{
return -ENOSYS;
}
static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
struct audio_device *adev = (struct audio_device *)dev;
pthread_mutex_lock(&adev->lock);
if (adev->mode != mode) {
ALOGD("%s: mode %d", __func__, (int)mode);
adev->mode = mode;
if ((mode == AUDIO_MODE_NORMAL || mode == AUDIO_MODE_IN_COMMUNICATION) &&
voice_is_in_call(adev)) {
voice_stop_call(adev);
adev->current_call_output = NULL;
/*
* After stopping the call, it must check if any active capture
* activity device needs to be re-selected.
*/
struct audio_usecase *usecase;
struct listnode *node;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_CAPTURE && usecase->stream.in != NULL) {
select_devices_with_force_switch(adev, usecase->id, true);
}
}
}
}
pthread_mutex_unlock(&adev->lock);
audio_extn_extspk_set_mode(adev->extspk, mode);
return 0;
}
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
int ret;
struct audio_device *adev = (struct audio_device *)dev;
ALOGD("%s: state %d", __func__, (int)state);
pthread_mutex_lock(&adev->lock);
if (audio_extn_tfa_98xx_is_supported() && adev->enable_hfp) {
ret = audio_extn_hfp_set_mic_mute(adev, state);
} else {
ret = voice_set_mic_mute(adev, state);
}
adev->mic_muted = state;
pthread_mutex_unlock(&adev->lock);
return ret;
}
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
*state = voice_get_mic_mute((struct audio_device *)dev);
return 0;
}
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev __unused,
const struct audio_config *config)
{
int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
/* Don't know if USB HIFI in this context so use true to be conservative */
if (check_input_parameters(config->sample_rate, config->format, channel_count,
true /*is_usb_hifi */) != 0)
return 0;
return get_stream_buffer_size(AUDIO_CAPTURE_PERIOD_DURATION_MSEC,
config->sample_rate, config->format,
channel_count,
false /* is_low_latency: since we don't know, be conservative */);
}
static bool adev_input_allow_hifi_record(struct audio_device *adev,
audio_devices_t devices,
audio_input_flags_t flags,
audio_source_t source) {
const bool allowed = true;
if (!audio_is_usb_in_device(devices))
return !allowed;
switch (flags) {
case AUDIO_INPUT_FLAG_NONE:
case AUDIO_INPUT_FLAG_FAST: // just fast, not fast|raw || fast|mmap
break;
default:
return !allowed;
}
switch (source) {
case AUDIO_SOURCE_DEFAULT:
case AUDIO_SOURCE_MIC:
case AUDIO_SOURCE_UNPROCESSED:
break;
default:
return !allowed;
}
switch (adev->mode) {
case 0:
break;
default:
return !allowed;
}
return allowed;
}
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,
audio_input_flags_t flags,
const char *address __unused,
audio_source_t source )
{
struct audio_device *adev = (struct audio_device *)dev;
struct stream_in *in;
int ret = 0, buffer_size, frame_size;
int channel_count;
bool is_low_latency = false;
bool is_usb_dev = audio_is_usb_in_device(devices);
bool may_use_hifi_record = adev_input_allow_hifi_record(adev,
devices,
flags,
source);
ALOGV("%s: enter: flags %#x, is_usb_dev %d, may_use_hifi_record %d,"
" sample_rate %u, channel_mask %#x, format %#x",
__func__, flags, is_usb_dev, may_use_hifi_record,
config->sample_rate, config->channel_mask, config->format);
*stream_in = NULL;
if (is_usb_dev && !is_usb_ready(adev, false /* is_playback */)) {
return -ENOSYS;
}
if (!(is_usb_dev && may_use_hifi_record)) {
if (config->sample_rate == 0)
config->sample_rate = DEFAULT_INPUT_SAMPLING_RATE;
if (config->channel_mask == AUDIO_CHANNEL_NONE)
config->channel_mask = AUDIO_CHANNEL_IN_MONO;
if (config->format == AUDIO_FORMAT_DEFAULT)
config->format = AUDIO_FORMAT_PCM_16_BIT;
channel_count = audio_channel_count_from_in_mask(config->channel_mask);
if (check_input_parameters(config->sample_rate, config->format, channel_count, false) != 0)
return -EINVAL;
}
if (audio_extn_tfa_98xx_is_supported() &&
(audio_extn_hfp_is_active(adev) || voice_is_in_call(adev)))
return -EINVAL;
in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
pthread_mutex_init(&in->pre_lock, (const pthread_mutexattr_t *) NULL);
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->stream.get_capture_position = in_get_capture_position;
in->stream.get_active_microphones = in_get_active_microphones;
in->stream.set_microphone_direction = in_set_microphone_direction;
in->stream.set_microphone_field_dimension = in_set_microphone_field_dimension;
in->stream.update_sink_metadata = in_update_sink_metadata;
in->device = devices;
in->source = source;
in->dev = adev;
in->standby = 1;
in->capture_handle = handle;
in->flags = flags;
in->direction = MIC_DIRECTION_UNSPECIFIED;
in->zoom = 0;
in->mmap_shared_memory_fd = -1; // not open
list_init(&in->aec_list);
list_init(&in->ns_list);
ALOGV("%s: source %d, config->channel_mask %#x", __func__, source, config->channel_mask);
if (source == AUDIO_SOURCE_VOICE_UPLINK ||
source == AUDIO_SOURCE_VOICE_DOWNLINK) {
/* Force channel config requested to mono if incall
record is being requested for only uplink/downlink */
if (config->channel_mask != AUDIO_CHANNEL_IN_MONO) {
config->channel_mask = AUDIO_CHANNEL_IN_MONO;
ret = -EINVAL;
goto err_open;
}
}
if (is_usb_dev && may_use_hifi_record) {
/* HiFi record selects an appropriate format, channel, rate combo
depending on sink capabilities*/
ret = read_usb_sup_params_and_compare(false /*is_playback*/,
&config->format,
&in->supported_formats[0],
MAX_SUPPORTED_FORMATS,
&config->channel_mask,
&in->supported_channel_masks[0],
MAX_SUPPORTED_CHANNEL_MASKS,
&config->sample_rate,
&in->supported_sample_rates[0],
MAX_SUPPORTED_SAMPLE_RATES);
if (ret != 0) {
ret = -EINVAL;
goto err_open;
}
channel_count = audio_channel_count_from_in_mask(config->channel_mask);
} else if (config->format == AUDIO_FORMAT_DEFAULT) {
config->format = AUDIO_FORMAT_PCM_16_BIT;
} else if (config->format == AUDIO_FORMAT_PCM_FLOAT ||
config->format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
config->format == AUDIO_FORMAT_PCM_8_24_BIT) {
bool ret_error = false;
/* 24 bit is restricted to UNPROCESSED source only,also format supported
from HAL is 8_24
*> In case of UNPROCESSED source, for 24 bit, if format requested is other than
8_24 return error indicating supported format is 8_24
*> In case of any other source requesting 24 bit or float return error
indicating format supported is 16 bit only.
on error flinger will retry with supported format passed
*/
if (!is_supported_24bits_audiosource(source)) {
config->format = AUDIO_FORMAT_PCM_16_BIT;
ret_error = true;
} else if (config->format != AUDIO_FORMAT_PCM_8_24_BIT) {
config->format = AUDIO_FORMAT_PCM_8_24_BIT;
ret_error = true;
}
if (ret_error) {
ret = -EINVAL;
goto err_open;
}
}
in->format = config->format;
in->channel_mask = config->channel_mask;
/* Update config params with the requested sample rate and channels */
if (in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
if (config->sample_rate == 0)
config->sample_rate = AFE_PROXY_SAMPLING_RATE;
if (config->sample_rate != 48000 && config->sample_rate != 16000 &&
config->sample_rate != 8000) {
config->sample_rate = AFE_PROXY_SAMPLING_RATE;
ret = -EINVAL;
goto err_open;
}
if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
config->format = AUDIO_FORMAT_PCM_16_BIT;
ret = -EINVAL;
goto err_open;
}
in->usecase = USECASE_AUDIO_RECORD_AFE_PROXY;
in->config = pcm_config_afe_proxy_record;
in->af_period_multiplier = 1;
} else if (is_usb_dev && may_use_hifi_record) {
in->usecase = USECASE_AUDIO_RECORD_HIFI;
in->config = pcm_config_audio_capture;
frame_size = audio_stream_in_frame_size(&in->stream);
buffer_size = get_stream_buffer_size(AUDIO_CAPTURE_PERIOD_DURATION_MSEC,
config->sample_rate,
config->format,
channel_count,
false /*is_low_latency*/);
in->config.period_size = buffer_size / frame_size;
in->config.rate = config->sample_rate;
in->af_period_multiplier = 1;
in->config.format = pcm_format_from_audio_format(config->format);
} else {
in->usecase = USECASE_AUDIO_RECORD;
if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
(in->flags & AUDIO_INPUT_FLAG_FAST) != 0) {
is_low_latency = true;
#if LOW_LATENCY_CAPTURE_USE_CASE
in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY;
#endif
in->realtime = may_use_noirq_mode(adev, in->usecase, in->flags);
if (!in->realtime) {
in->config = pcm_config_audio_capture;
frame_size = audio_stream_in_frame_size(&in->stream);
buffer_size = get_stream_buffer_size(AUDIO_CAPTURE_PERIOD_DURATION_MSEC,
config->sample_rate,
config->format,
channel_count,
is_low_latency);
in->config.period_size = buffer_size / frame_size;
in->config.rate = config->sample_rate;
in->af_period_multiplier = 1;
} else {
// period size is left untouched for rt mode playback
in->config = pcm_config_audio_capture_rt;
in->af_period_multiplier = af_period_multiplier;
}
} else if ((config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE) &&
((in->flags & AUDIO_INPUT_FLAG_MMAP_NOIRQ) != 0)) {
// FIXME: Add support for multichannel capture over USB using MMAP
in->usecase = USECASE_AUDIO_RECORD_MMAP;
in->config = pcm_config_mmap_capture;
in->stream.start = in_start;
in->stream.stop = in_stop;
in->stream.create_mmap_buffer = in_create_mmap_buffer;
in->stream.get_mmap_position = in_get_mmap_position;
in->af_period_multiplier = 1;
ALOGV("%s: USECASE_AUDIO_RECORD_MMAP", __func__);
} else if (in->source == AUDIO_SOURCE_VOICE_COMMUNICATION &&
in->flags & AUDIO_INPUT_FLAG_VOIP_TX &&
(config->sample_rate == 8000 ||
config->sample_rate == 16000 ||
config->sample_rate == 32000 ||
config->sample_rate == 48000) &&
channel_count == 1) {
in->usecase = USECASE_AUDIO_RECORD_VOIP;
in->config = pcm_config_audio_capture;
frame_size = audio_stream_in_frame_size(&in->stream);
buffer_size = get_stream_buffer_size(VOIP_CAPTURE_PERIOD_DURATION_MSEC,
config->sample_rate,
config->format,
channel_count, false /*is_low_latency*/);
in->config.period_size = buffer_size / frame_size;
in->config.period_count = VOIP_CAPTURE_PERIOD_COUNT;
in->config.rate = config->sample_rate;
in->af_period_multiplier = 1;
} else {
in->config = pcm_config_audio_capture;
frame_size = audio_stream_in_frame_size(&in->stream);
buffer_size = get_stream_buffer_size(AUDIO_CAPTURE_PERIOD_DURATION_MSEC,
config->sample_rate,
config->format,
channel_count,
is_low_latency);
in->config.period_size = buffer_size / frame_size;
in->config.rate = config->sample_rate;
in->af_period_multiplier = 1;
}
if (config->format == AUDIO_FORMAT_PCM_8_24_BIT)
in->config.format = PCM_FORMAT_S24_LE;
}
in->config.channels = channel_count;
in->sample_rate = in->config.rate;
register_format(in->format, in->supported_formats);
register_channel_mask(in->channel_mask, in->supported_channel_masks);
register_sample_rate(in->sample_rate, in->supported_sample_rates);
in->error_log = error_log_create(
ERROR_LOG_ENTRIES,
NANOS_PER_SECOND /* aggregate consecutive identical errors within one second */);
/* This stream could be for sound trigger lab,
get sound trigger pcm if present */
audio_extn_sound_trigger_check_and_get_session(in);
if (in->is_st_session)
in->flags |= AUDIO_INPUT_FLAG_HW_HOTWORD;
lock_input_stream(in);
audio_extn_snd_mon_register_listener(in, in_snd_mon_cb);
pthread_mutex_lock(&adev->lock);
in->card_status = adev->card_status;
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&in->lock);
stream_app_type_cfg_init(&in->app_type_cfg);
*stream_in = &in->stream;
ALOGV("%s: exit", __func__);
return 0;
err_open:
free(in);
*stream_in = NULL;
return ret;
}
static void adev_close_input_stream(struct audio_hw_device *dev __unused,
struct audio_stream_in *stream)
{
struct stream_in *in = (struct stream_in *)stream;
ALOGV("%s", __func__);
// must deregister from sndmonitor first to prevent races
// between the callback and close_stream
audio_extn_snd_mon_unregister_listener(stream);
in_standby(&stream->common);
error_log_destroy(in->error_log);
in->error_log = NULL;
pthread_mutex_destroy(&in->pre_lock);
pthread_mutex_destroy(&in->lock);
free(stream);
return;
}
static int adev_dump(const audio_hw_device_t *device __unused, int fd __unused)
{
return 0;
}
/* verifies input and output devices and their capabilities.
*
* This verification is required when enabling extended bit-depth or
* sampling rates, as not all qcom products support it.
*
* Suitable for calling only on initialization such as adev_open().
* It fills the audio_device use_case_table[] array.
*
* Has a side-effect that it needs to configure audio routing / devices
* in order to power up the devices and read the device parameters.
* It does not acquire any hw device lock. Should restore the devices
* back to "normal state" upon completion.
*/
static int adev_verify_devices(struct audio_device *adev)
{
/* enumeration is a bit difficult because one really wants to pull
* the use_case, device id, etc from the hidden pcm_device_table[].
* In this case there are the following use cases and device ids.
*
* [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = {0, 0},
* [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = {15, 15},
* [USECASE_AUDIO_PLAYBACK_HIFI] = {1, 1},
* [USECASE_AUDIO_PLAYBACK_OFFLOAD] = {9, 9},
* [USECASE_AUDIO_RECORD] = {0, 0},
* [USECASE_AUDIO_RECORD_LOW_LATENCY] = {15, 15},
* [USECASE_VOICE_CALL] = {2, 2},
*
* USECASE_AUDIO_PLAYBACK_OFFLOAD, USECASE_AUDIO_PLAYBACK_HIFI omitted.
* USECASE_VOICE_CALL omitted, but possible for either input or output.
*/
/* should be the usecases enabled in adev_open_input_stream() */
static const int test_in_usecases[] = {
USECASE_AUDIO_RECORD,
USECASE_AUDIO_RECORD_LOW_LATENCY, /* does not appear to be used */
};
/* should be the usecases enabled in adev_open_output_stream()*/
static const int test_out_usecases[] = {
USECASE_AUDIO_PLAYBACK_DEEP_BUFFER,
USECASE_AUDIO_PLAYBACK_LOW_LATENCY,
};
static const usecase_type_t usecase_type_by_dir[] = {
PCM_PLAYBACK,
PCM_CAPTURE,
};
static const unsigned flags_by_dir[] = {
PCM_OUT,
PCM_IN,
};
size_t i;
unsigned dir;
const unsigned card_id = adev->snd_card;
char info[512]; /* for possible debug info */
for (dir = 0; dir < 2; ++dir) {
const usecase_type_t usecase_type = usecase_type_by_dir[dir];
const unsigned flags_dir = flags_by_dir[dir];
const size_t testsize =
dir ? ARRAY_SIZE(test_in_usecases) : ARRAY_SIZE(test_out_usecases);
const int *testcases =
dir ? test_in_usecases : test_out_usecases;
const audio_devices_t audio_device =
dir ? AUDIO_DEVICE_IN_BUILTIN_MIC : AUDIO_DEVICE_OUT_SPEAKER;
for (i = 0; i < testsize; ++i) {
const audio_usecase_t audio_usecase = testcases[i];
int device_id;
snd_device_t snd_device;
struct pcm_params **pparams;
struct stream_out out;
struct stream_in in;
struct audio_usecase uc_info;
int retval;
pparams = &adev->use_case_table[audio_usecase];
pcm_params_free(*pparams); /* can accept null input */
*pparams = NULL;
/* find the device ID for the use case (signed, for error) */
device_id = platform_get_pcm_device_id(audio_usecase, usecase_type);
if (device_id < 0)
continue;
/* prepare structures for device probing */
memset(&uc_info, 0, sizeof(uc_info));
uc_info.id = audio_usecase;
uc_info.type = usecase_type;
if (dir) {
memset(&in, 0, sizeof(in));
in.device = audio_device;
in.source = AUDIO_SOURCE_VOICE_COMMUNICATION;
uc_info.stream.in = &in;
}
memset(&out, 0, sizeof(out));
out.devices = audio_device; /* only field needed in select_devices */
uc_info.stream.out = &out;
uc_info.devices = audio_device;
uc_info.in_snd_device = SND_DEVICE_NONE;
uc_info.out_snd_device = SND_DEVICE_NONE;
list_add_tail(&adev->usecase_list, &uc_info.list);
/* select device - similar to start_(in/out)put_stream() */
retval = select_devices(adev, audio_usecase);
if (retval >= 0) {
*pparams = pcm_params_get(card_id, device_id, flags_dir);
#if LOG_NDEBUG == 0
if (*pparams) {
ALOGV("%s: (%s) card %d device %d", __func__,
dir ? "input" : "output", card_id, device_id);
pcm_params_to_string(*pparams, info, ARRAY_SIZE(info));
} else {
ALOGV("%s: cannot locate card %d device %d", __func__, card_id, device_id);
}
#endif
}
/* deselect device - similar to stop_(in/out)put_stream() */
/* 1. Get and set stream specific mixer controls */
retval = disable_audio_route(adev, &uc_info);
/* 2. Disable the rx device */
retval = disable_snd_device(adev,
dir ? uc_info.in_snd_device : uc_info.out_snd_device);
list_remove(&uc_info.list);
}
}
return 0;
}
static int adev_close(hw_device_t *device)
{
size_t i;
pthread_mutex_lock(&adev_init_lock);
if (!device || ((struct audio_device *)device != adev))
goto done;
if ((--audio_device_ref_count) == 0) {
audio_extn_snd_mon_unregister_listener(adev);
audio_extn_tfa_98xx_deinit();
audio_extn_ma_deinit();
audio_route_free(adev->audio_route);
free(adev->snd_dev_ref_cnt);
platform_deinit(adev->platform);
audio_extn_extspk_deinit(adev->extspk);
audio_extn_sound_trigger_deinit(adev);
audio_extn_snd_mon_deinit();
for (i = 0; i < ARRAY_SIZE(adev->use_case_table); ++i) {
pcm_params_free(adev->use_case_table[i]);
}
if (adev->adm_deinit)
adev->adm_deinit(adev->adm_data);
pthread_mutex_destroy(&adev->lock);
free(device);
adev = NULL;
}
done:
pthread_mutex_unlock(&adev_init_lock);
return 0;
}
/* This returns 1 if the input parameter looks at all plausible as a low latency period size,
* or 0 otherwise. A return value of 1 doesn't mean the value is guaranteed to work,
* just that it _might_ work.
*/
static int period_size_is_plausible_for_low_latency(int period_size)
{
switch (period_size) {
case 48:
case 96:
case 144:
case 160:
case 192:
case 240:
case 320:
case 480:
return 1;
default:
return 0;
}
}
static void adev_snd_mon_cb(void * stream __unused, struct str_parms * parms)
{
int card;
card_status_t status;
if (!parms)
return;
if (parse_snd_card_status(parms, &card, &status) < 0)
return;
pthread_mutex_lock(&adev->lock);
bool valid_cb = (card == adev->snd_card);
if (valid_cb) {
if (adev->card_status != status) {
adev->card_status = status;
platform_snd_card_update(adev->platform, status);
}
}
pthread_mutex_unlock(&adev->lock);
return;
}
/* out and adev lock held */
static int check_a2dp_restore_l(struct audio_device *adev, struct stream_out *out, bool restore)
{
struct audio_usecase *uc_info;
float left_p;
float right_p;
audio_devices_t devices;
uc_info = get_usecase_from_list(adev, out->usecase);
if (uc_info == NULL) {
ALOGE("%s: Could not find the usecase (%d) in the list",
__func__, out->usecase);
return -EINVAL;
}
ALOGD("%s: enter: usecase(%d: %s)", __func__,
out->usecase, use_case_table[out->usecase]);
if (restore) {
// restore A2DP device for active usecases and unmute if required
if ((out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) &&
!is_a2dp_device(uc_info->out_snd_device)) {
ALOGD("%s: restoring A2DP and unmuting stream", __func__);
select_devices(adev, uc_info->id);
pthread_mutex_lock(&out->compr_mute_lock);
if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
(out->a2dp_compress_mute)) {
out->a2dp_compress_mute = false;
out_set_compr_volume(&out->stream, out->volume_l, out->volume_r);
}
pthread_mutex_unlock(&out->compr_mute_lock);
}
} else {
// mute compress stream if suspended
pthread_mutex_lock(&out->compr_mute_lock);
if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
(!out->a2dp_compress_mute)) {
if (!out->standby) {
ALOGD("%s: selecting speaker and muting stream", __func__);
devices = out->devices;
out->devices = AUDIO_DEVICE_OUT_SPEAKER;
left_p = out->volume_l;
right_p = out->volume_r;
if (out->offload_state == OFFLOAD_STATE_PLAYING)
compress_pause(out->compr);
out_set_compr_volume(&out->stream, 0.0f, 0.0f);
out->a2dp_compress_mute = true;
select_devices(adev, out->usecase);
if (out->offload_state == OFFLOAD_STATE_PLAYING)
compress_resume(out->compr);
out->devices = devices;
out->volume_l = left_p;
out->volume_r = right_p;
}
}
pthread_mutex_unlock(&out->compr_mute_lock);
}
ALOGV("%s: exit", __func__);
return 0;
}
int check_a2dp_restore(struct audio_device *adev, struct stream_out *out, bool restore)
{
int ret = 0;
lock_output_stream(out);
pthread_mutex_lock(&adev->lock);
ret = check_a2dp_restore_l(adev, out, restore);
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&out->lock);
return ret;
}
static int adev_open(const hw_module_t *module, const char *name,
hw_device_t **device)
{
int i, ret;
ALOGD("%s: enter", __func__);
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL;
pthread_mutex_lock(&adev_init_lock);
if (audio_device_ref_count != 0) {
*device = &adev->device.common;
audio_device_ref_count++;
ALOGV("%s: returning existing instance of adev", __func__);
ALOGV("%s: exit", __func__);
pthread_mutex_unlock(&adev_init_lock);
return 0;
}
adev = calloc(1, sizeof(struct audio_device));
pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL);
adev->device.common.tag = HARDWARE_DEVICE_TAG;
adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
adev->device.common.module = (struct hw_module_t *)module;
adev->device.common.close = adev_close;
adev->device.init_check = adev_init_check;
adev->device.set_voice_volume = adev_set_voice_volume;
adev->device.set_master_volume = adev_set_master_volume;
adev->device.get_master_volume = adev_get_master_volume;
adev->device.set_master_mute = adev_set_master_mute;
adev->device.get_master_mute = adev_get_master_mute;
adev->device.set_mode = adev_set_mode;
adev->device.set_mic_mute = adev_set_mic_mute;
adev->device.get_mic_mute = adev_get_mic_mute;
adev->device.set_parameters = adev_set_parameters;
adev->device.get_parameters = adev_get_parameters;
adev->device.get_input_buffer_size = adev_get_input_buffer_size;
adev->device.open_output_stream = adev_open_output_stream;
adev->device.close_output_stream = adev_close_output_stream;
adev->device.open_input_stream = adev_open_input_stream;
adev->device.close_input_stream = adev_close_input_stream;
adev->device.dump = adev_dump;
adev->device.get_microphones = adev_get_microphones;
/* Set the default route before the PCM stream is opened */
pthread_mutex_lock(&adev->lock);
adev->mode = AUDIO_MODE_NORMAL;
adev->primary_output = NULL;
adev->bluetooth_nrec = true;
adev->acdb_settings = TTY_MODE_OFF;
adev->a2dp_started = false;
/* adev->cur_hdmi_channels = 0; by calloc() */
adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int));
voice_init(adev);
list_init(&adev->usecase_list);
pthread_mutex_unlock(&adev->lock);
/* Loads platform specific libraries dynamically */
adev->platform = platform_init(adev);
if (!adev->platform) {
free(adev->snd_dev_ref_cnt);
free(adev);
ALOGE("%s: Failed to init platform data, aborting.", __func__);
*device = NULL;
pthread_mutex_unlock(&adev_init_lock);
return -EINVAL;
}
adev->extspk = audio_extn_extspk_init(adev);
adev->visualizer_lib = dlopen(VISUALIZER_LIBRARY_PATH, RTLD_NOW);
if (adev->visualizer_lib == NULL) {
ALOGW("%s: DLOPEN failed for %s", __func__, VISUALIZER_LIBRARY_PATH);
} else {
ALOGV("%s: DLOPEN successful for %s", __func__, VISUALIZER_LIBRARY_PATH);
adev->visualizer_start_output =
(int (*)(audio_io_handle_t, int, int, int))dlsym(adev->visualizer_lib,
"visualizer_hal_start_output");
adev->visualizer_stop_output =
(int (*)(audio_io_handle_t, int))dlsym(adev->visualizer_lib,
"visualizer_hal_stop_output");
}
adev->offload_effects_lib = dlopen(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, RTLD_NOW);
if (adev->offload_effects_lib == NULL) {
ALOGW("%s: DLOPEN failed for %s", __func__,
OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
} else {
ALOGV("%s: DLOPEN successful for %s", __func__,
OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
adev->offload_effects_start_output =
(int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
"offload_effects_bundle_hal_start_output");
adev->offload_effects_stop_output =
(int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
"offload_effects_bundle_hal_stop_output");
}
adev->adm_lib = dlopen(ADM_LIBRARY_PATH, RTLD_NOW);
if (adev->adm_lib == NULL) {
ALOGW("%s: DLOPEN failed for %s", __func__, ADM_LIBRARY_PATH);
} else {
ALOGV("%s: DLOPEN successful for %s", __func__, ADM_LIBRARY_PATH);
adev->adm_init = (adm_init_t)
dlsym(adev->adm_lib, "adm_init");
adev->adm_deinit = (adm_deinit_t)
dlsym(adev->adm_lib, "adm_deinit");
adev->adm_register_input_stream = (adm_register_input_stream_t)
dlsym(adev->adm_lib, "adm_register_input_stream");
adev->adm_register_output_stream = (adm_register_output_stream_t)
dlsym(adev->adm_lib, "adm_register_output_stream");
adev->adm_deregister_stream = (adm_deregister_stream_t)
dlsym(adev->adm_lib, "adm_deregister_stream");
adev->adm_request_focus = (adm_request_focus_t)
dlsym(adev->adm_lib, "adm_request_focus");
adev->adm_abandon_focus = (adm_abandon_focus_t)
dlsym(adev->adm_lib, "adm_abandon_focus");
adev->adm_set_config = (adm_set_config_t)
dlsym(adev->adm_lib, "adm_set_config");
adev->adm_request_focus_v2 = (adm_request_focus_v2_t)
dlsym(adev->adm_lib, "adm_request_focus_v2");
adev->adm_is_noirq_avail = (adm_is_noirq_avail_t)
dlsym(adev->adm_lib, "adm_is_noirq_avail");
adev->adm_on_routing_change = (adm_on_routing_change_t)
dlsym(adev->adm_lib, "adm_on_routing_change");
}
adev->bt_wb_speech_enabled = false;
adev->enable_voicerx = false;
*device = &adev->device.common;
if (k_enable_extended_precision)
adev_verify_devices(adev);
char value[PROPERTY_VALUE_MAX];
int trial;
if ((property_get("vendor.audio_hal.period_size", value, NULL) > 0) ||
(property_get("audio_hal.period_size", value, NULL) > 0)) {
trial = atoi(value);
if (period_size_is_plausible_for_low_latency(trial)) {
pcm_config_low_latency.period_size = trial;
pcm_config_low_latency.start_threshold = trial / 4;
pcm_config_low_latency.avail_min = trial / 4;
configured_low_latency_capture_period_size = trial;
}
}
if ((property_get("vendor.audio_hal.in_period_size", value, NULL) > 0) ||
(property_get("audio_hal.in_period_size", value, NULL) > 0)) {
trial = atoi(value);
if (period_size_is_plausible_for_low_latency(trial)) {
configured_low_latency_capture_period_size = trial;
}
}
adev->mic_break_enabled = property_get_bool("vendor.audio.mic_break", false);
adev->camera_orientation = CAMERA_DEFAULT;
// commented as full set of app type cfg is sent from platform
// audio_extn_utils_send_default_app_type_cfg(adev->platform, adev->mixer);
audio_device_ref_count++;
if ((property_get("vendor.audio_hal.period_multiplier", value, NULL) > 0) ||
(property_get("audio_hal.period_multiplier", value, NULL) > 0)) {
af_period_multiplier = atoi(value);
if (af_period_multiplier < 0) {
af_period_multiplier = 2;
} else if (af_period_multiplier > 4) {
af_period_multiplier = 4;
}
ALOGV("new period_multiplier = %d", af_period_multiplier);
}
audio_extn_tfa_98xx_init(adev);
audio_extn_ma_init(adev->platform);
audio_extn_audiozoom_init();
pthread_mutex_unlock(&adev_init_lock);
if (adev->adm_init)
adev->adm_data = adev->adm_init();
audio_extn_perf_lock_init();
audio_extn_snd_mon_init();
pthread_mutex_lock(&adev->lock);
audio_extn_snd_mon_register_listener(NULL, adev_snd_mon_cb);
adev->card_status = CARD_STATUS_ONLINE;
pthread_mutex_unlock(&adev->lock);
audio_extn_sound_trigger_init(adev);/* dependent on snd_mon_init() */
ALOGD("%s: exit", __func__);
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 = "QCOM Audio HAL",
.author = "Code Aurora Forum",
.methods = &hal_module_methods,
},
};
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