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gerrit-public.fairphone.software / platform / hardware / qcom / audio / 1eac8b0e2df707fa77d12ed4d4bdc3723dfffd6c / . / hal / audio_hw.c
blob: 0b7812cd4175886d0432662cc508a0d940e16ceb [file] [log] [blame]
/*
* Copyright (c) 2013-2017, The Linux Foundation. All rights reserved.
* Not a Contribution.
*
* Copyright (C) 2013 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*
* This file was modified by DTS, Inc. The portions of the
* code modified by DTS, Inc are copyrighted and
* licensed separately, as follows:
*
* (C) 2014 DTS, Inc.
*
* 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|ATRACE_TAG_HAL)
/*#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 <cutils/log.h>
#include <cutils/trace.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <cutils/atomic.h>
#include <cutils/sched_policy.h>
#include <hardware/audio_effect.h>
#include <system/thread_defs.h>
#include <audio_effects/effect_aec.h>
#include <audio_effects/effect_ns.h>
#include <audio_utils/format.h>
#include "audio_hw.h"
#include "platform_api.h"
#include <platform.h>
#include "audio_extn.h"
#include "voice_extn.h"
#include "sound/compress_params.h"
#include "sound/asound.h"
#ifdef DYNAMIC_LOG_ENABLED
#include <log_xml_parser.h>
#define LOG_MASK HAL_MOD_FILE_AUDIO_HW
#include <log_utils.h>
#endif
#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
/*DIRECT PCM has same buffer sizes as DEEP Buffer*/
#define DIRECT_PCM_NUM_FRAGMENTS 2
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
#define DSD_VOLUME_MIN_DB (-110)
#define PROXY_OPEN_RETRY_COUNT 100
#define PROXY_OPEN_WAIT_TIME 20
#ifdef USE_LL_AS_PRIMARY_OUTPUT
#define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_LOW_LATENCY
#define PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY pcm_config_low_latency
#else
#define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_DEEP_BUFFER
#define PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY pcm_config_deep_buffer
#endif
#define ULL_PERIOD_SIZE (DEFAULT_OUTPUT_SAMPLING_RATE/1000)
static unsigned int configured_low_latency_capture_period_size =
LOW_LATENCY_CAPTURE_PERIOD_SIZE;
struct pcm_config pcm_config_deep_buffer = {
.channels = 2,
.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 = 2,
.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,
};
static int af_period_multiplier = 4;
struct pcm_config pcm_config_rt = {
.channels = 2,
.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_audio_capture = {
.channels = 2,
.period_count = AUDIO_CAPTURE_PERIOD_COUNT,
.format = PCM_FORMAT_S16_LE,
};
struct pcm_config pcm_config_audio_capture_rt = {
.channels = 2,
.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
};
#define AFE_PROXY_CHANNEL_COUNT 2
#define AFE_PROXY_SAMPLING_RATE 48000
#define AFE_PROXY_PLAYBACK_PERIOD_SIZE 768
#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 768
#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 = INT_MAX,
.avail_min = AFE_PROXY_RECORD_PERIOD_SIZE,
};
#define AUDIO_MAX_PCM_FORMATS 7
const uint32_t format_to_bitwidth_table[AUDIO_MAX_PCM_FORMATS] = {
[AUDIO_FORMAT_DEFAULT] = 0,
[AUDIO_FORMAT_PCM_16_BIT] = sizeof(uint16_t),
[AUDIO_FORMAT_PCM_8_BIT] = sizeof(uint8_t),
[AUDIO_FORMAT_PCM_32_BIT] = sizeof(uint32_t),
[AUDIO_FORMAT_PCM_8_24_BIT] = sizeof(uint32_t),
[AUDIO_FORMAT_PCM_FLOAT] = sizeof(float),
[AUDIO_FORMAT_PCM_24_BIT_PACKED] = sizeof(uint8_t) * 3,
};
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_ULL] = "audio-ull-playback",
[USECASE_AUDIO_PLAYBACK_MULTI_CH] = "multi-channel-playback",
[USECASE_AUDIO_PLAYBACK_OFFLOAD] = "compress-offload-playback",
//Enabled for Direct_PCM
[USECASE_AUDIO_PLAYBACK_OFFLOAD2] = "compress-offload-playback2",
[USECASE_AUDIO_PLAYBACK_OFFLOAD3] = "compress-offload-playback3",
[USECASE_AUDIO_PLAYBACK_OFFLOAD4] = "compress-offload-playback4",
[USECASE_AUDIO_PLAYBACK_OFFLOAD5] = "compress-offload-playback5",
[USECASE_AUDIO_PLAYBACK_OFFLOAD6] = "compress-offload-playback6",
[USECASE_AUDIO_PLAYBACK_OFFLOAD7] = "compress-offload-playback7",
[USECASE_AUDIO_PLAYBACK_OFFLOAD8] = "compress-offload-playback8",
[USECASE_AUDIO_PLAYBACK_OFFLOAD9] = "compress-offload-playback9",
[USECASE_AUDIO_RECORD] = "audio-record",
[USECASE_AUDIO_RECORD_COMPRESS] = "audio-record-compress",
[USECASE_AUDIO_RECORD_COMPRESS2] = "audio-record-compress2",
[USECASE_AUDIO_RECORD_COMPRESS3] = "audio-record-compress3",
[USECASE_AUDIO_RECORD_COMPRESS4] = "audio-record-compress4",
[USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record",
[USECASE_AUDIO_RECORD_FM_VIRTUAL] = "fm-virtual-record",
[USECASE_AUDIO_PLAYBACK_FM] = "play-fm",
[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_COMPRESS_VOIP_CALL] = "compress-voip-call",
[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_INCALL_REC_UPLINK_COMPRESS] = "incall-rec-uplink-compress",
[USECASE_INCALL_REC_DOWNLINK_COMPRESS] = "incall-rec-downlink-compress",
[USECASE_INCALL_REC_UPLINK_AND_DOWNLINK_COMPRESS] = "incall-rec-uplink-and-downlink-compress",
[USECASE_INCALL_MUSIC_UPLINK] = "incall_music_uplink",
[USECASE_INCALL_MUSIC_UPLINK2] = "incall_music_uplink2",
[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_AUDIO_PLAYBACK_EXT_DISP_SILENCE] = "silence-playback",
};
static const audio_usecase_t offload_usecases[] = {
USECASE_AUDIO_PLAYBACK_OFFLOAD,
USECASE_AUDIO_PLAYBACK_OFFLOAD2,
USECASE_AUDIO_PLAYBACK_OFFLOAD3,
USECASE_AUDIO_PLAYBACK_OFFLOAD4,
USECASE_AUDIO_PLAYBACK_OFFLOAD5,
USECASE_AUDIO_PLAYBACK_OFFLOAD6,
USECASE_AUDIO_PLAYBACK_OFFLOAD7,
USECASE_AUDIO_PLAYBACK_OFFLOAD8,
USECASE_AUDIO_PLAYBACK_OFFLOAD9,
};
#define STRING_TO_ENUM(string) { #string, string }
struct string_to_enum {
const char *name;
uint32_t value;
};
static const struct string_to_enum out_channels_name_to_enum_table[] = {
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_2POINT1),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_SURROUND),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_PENTA),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_6POINT1),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
};
static const struct string_to_enum out_formats_name_to_enum_table[] = {
STRING_TO_ENUM(AUDIO_FORMAT_AC3),
STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
STRING_TO_ENUM(AUDIO_FORMAT_E_AC3_JOC),
STRING_TO_ENUM(AUDIO_FORMAT_DOLBY_TRUEHD),
STRING_TO_ENUM(AUDIO_FORMAT_DTS),
STRING_TO_ENUM(AUDIO_FORMAT_DTS_HD),
};
//list of all supported sample rates by HDMI specification.
static const int out_hdmi_sample_rates[] = {
32000, 44100, 48000, 88200, 96000, 176400, 192000,
};
static const struct string_to_enum out_hdmi_sample_rates_name_to_enum_table[] = {
STRING_TO_ENUM(32000),
STRING_TO_ENUM(44100),
STRING_TO_ENUM(48000),
STRING_TO_ENUM(88200),
STRING_TO_ENUM(96000),
STRING_TO_ENUM(176400),
STRING_TO_ENUM(192000),
};
static struct audio_device *adev = NULL;
static pthread_mutex_t adev_init_lock;
static unsigned int audio_device_ref_count;
//cache last MBDRC cal step level
static int last_known_cal_step = -1 ;
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 (is_offload_usecase(out->usecase) ||
!adev->adm_register_output_stream)
return;
// register stream first for backward compatibility
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)
{
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)
{
struct audio_device *adev = in->dev;
if (adev->adm_abandon_focus)
adev->adm_abandon_focus(adev->adm_data, in->capture_handle);
}
__attribute__ ((visibility ("default")))
bool audio_hw_send_gain_dep_calibration(int level) {
bool ret_val = false;
ALOGV("%s: called ...", __func__);
pthread_mutex_lock(&adev_init_lock);
if (adev != NULL && adev->platform != NULL) {
pthread_mutex_lock(&adev->lock);
ret_val = platform_send_gain_dep_cal(adev->platform, level);
// if cal set fails, cache level info
// if cal set succeds, reset known last cal set
if (!ret_val)
last_known_cal_step = level;
else if (last_known_cal_step != -1)
last_known_cal_step = -1;
pthread_mutex_unlock(&adev->lock);
} else {
ALOGE("%s: %s is NULL", __func__, adev == NULL ? "adev" : "adev->platform");
}
pthread_mutex_unlock(&adev_init_lock);
return ret_val;
}
static int check_and_set_gapless_mode(struct audio_device *adev, bool enable_gapless)
{
bool gapless_enabled = false;
const char *mixer_ctl_name = "Compress Gapless Playback";
struct mixer_ctl *ctl;
ALOGV("%s:", __func__);
gapless_enabled = property_get_bool("audio.offload.gapless.enabled", false);
/*Disable gapless if its AV playback*/
gapless_enabled = gapless_enabled && enable_gapless;
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;
}
__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)
{
if (format == AUDIO_FORMAT_MP3 ||
format == AUDIO_FORMAT_MP2 ||
format == AUDIO_FORMAT_AAC_LC ||
format == AUDIO_FORMAT_AAC_HE_V1 ||
format == AUDIO_FORMAT_AAC_HE_V2 ||
format == AUDIO_FORMAT_AAC_ADTS_LC ||
format == AUDIO_FORMAT_AAC_ADTS_HE_V1 ||
format == AUDIO_FORMAT_AAC_ADTS_HE_V2 ||
format == AUDIO_FORMAT_AAC_LATM_LC ||
format == AUDIO_FORMAT_AAC_LATM_HE_V1 ||
format == AUDIO_FORMAT_AAC_LATM_HE_V2 ||
format == AUDIO_FORMAT_PCM_24_BIT_PACKED ||
format == AUDIO_FORMAT_PCM_8_24_BIT ||
format == AUDIO_FORMAT_PCM_FLOAT ||
format == AUDIO_FORMAT_PCM_32_BIT ||
format == AUDIO_FORMAT_PCM_16_BIT ||
format == AUDIO_FORMAT_AC3 ||
format == AUDIO_FORMAT_E_AC3 ||
format == AUDIO_FORMAT_DOLBY_TRUEHD ||
format == AUDIO_FORMAT_DTS ||
format == AUDIO_FORMAT_DTS_HD ||
format == AUDIO_FORMAT_FLAC ||
format == AUDIO_FORMAT_ALAC ||
format == AUDIO_FORMAT_APE ||
format == AUDIO_FORMAT_DSD ||
format == AUDIO_FORMAT_VORBIS ||
format == AUDIO_FORMAT_WMA ||
format == AUDIO_FORMAT_WMA_PRO ||
format == AUDIO_FORMAT_APTX)
return true;
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);
}
int get_snd_card_state(struct audio_device *adev)
{
int snd_scard_state;
if (!adev)
return SND_CARD_STATE_OFFLINE;
pthread_mutex_lock(&adev->snd_card_status.lock);
snd_scard_state = adev->snd_card_status.state;
pthread_mutex_unlock(&adev->snd_card_status.lock);
return snd_scard_state;
}
static int set_snd_card_state(struct audio_device *adev, int snd_scard_state)
{
if (!adev)
return -ENOSYS;
pthread_mutex_lock(&adev->snd_card_status.lock);
if (adev->snd_card_status.state != snd_scard_state) {
adev->snd_card_status.state = snd_scard_state;
platform_snd_card_update(adev->platform, snd_scard_state);
}
pthread_mutex_unlock(&adev->snd_card_status.lock);
return 0;
}
static int enable_audio_route_for_voice_usecases(struct audio_device *adev,
struct audio_usecase *uc_info)
{
struct listnode *node;
struct audio_usecase *usecase;
if (uc_info == NULL)
return -EINVAL;
/* Re-route all voice 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 ((usecase->type == VOICE_CALL) && (usecase != uc_info))
enable_audio_route(adev, usecase);
}
return 0;
}
static void enable_asrc_mode(struct audio_device *adev)
{
ALOGV("%s", __func__);
audio_route_apply_and_update_path(adev->audio_route,
"asrc-mode");
adev->asrc_mode_enabled = true;
}
static void disable_asrc_mode(struct audio_device *adev)
{
ALOGV("%s", __func__);
audio_route_reset_and_update_path(adev->audio_route,
"asrc-mode");
adev->asrc_mode_enabled = false;
}
/*
* - Enable ASRC mode for incoming mix path use case(Headphone backend)if Headphone
* 44.1 or Native DSD backends are enabled for any of current use case.
* e.g. 48-> + (Naitve DSD or Headphone 44.1)
* - Disable current mix path use case(Headphone backend) and re-enable it with
* ASRC mode for incoming Headphone 44.1 or Native DSD use case.
* e.g. Naitve DSD or Headphone 44.1 -> + 48
*/
static void check_and_set_asrc_mode(struct audio_device *adev,
struct audio_usecase *uc_info,
snd_device_t snd_device)
{
ALOGV("%s snd device %d", __func__, snd_device);
int i, num_new_devices = 0;
snd_device_t split_new_snd_devices[SND_DEVICE_OUT_END];
/*
*Split snd device for new combo use case
*e.g. Headphopne 44.1-> + Ringtone (Headphone + Speaker)
*/
if (platform_split_snd_device(adev->platform,
snd_device,
&num_new_devices,
split_new_snd_devices) == 0) {
for (i = 0; i < num_new_devices; i++)
check_and_set_asrc_mode(adev, uc_info, split_new_snd_devices[i]);
} else {
int new_backend_idx = platform_get_backend_index(snd_device);
if (((new_backend_idx == HEADPHONE_BACKEND) ||
(new_backend_idx == HEADPHONE_44_1_BACKEND) ||
(new_backend_idx == DSD_NATIVE_BACKEND)) &&
!adev->asrc_mode_enabled) {
struct listnode *node = NULL;
struct audio_usecase *uc = NULL;
struct stream_out *curr_out = NULL;
int usecase_backend_idx = DEFAULT_CODEC_BACKEND;
int i, num_devices, ret = 0;
snd_device_t split_snd_devices[SND_DEVICE_OUT_END];
list_for_each(node, &adev->usecase_list) {
uc = node_to_item(node, struct audio_usecase, list);
curr_out = (struct stream_out*) uc->stream.out;
if (curr_out && PCM_PLAYBACK == uc->type && uc != uc_info) {
/*
*Split snd device for existing combo use case
*e.g. Ringtone (Headphone + Speaker) + Headphopne 44.1
*/
ret = platform_split_snd_device(adev->platform,
uc->out_snd_device,
&num_devices,
split_snd_devices);
if (ret < 0 || num_devices == 0) {
ALOGV("%s: Unable to split uc->out_snd_device: %d",__func__, uc->out_snd_device);
split_snd_devices[0] = uc->out_snd_device;
num_devices = 1;
}
for (i = 0; i < num_devices; i++) {
usecase_backend_idx = platform_get_backend_index(split_snd_devices[i]);
ALOGD("%s:snd_dev %d usecase_backend_idx %d",__func__, split_snd_devices[i],usecase_backend_idx);
if((new_backend_idx == HEADPHONE_BACKEND) &&
((usecase_backend_idx == HEADPHONE_44_1_BACKEND) ||
(usecase_backend_idx == DSD_NATIVE_BACKEND))) {
ALOGD("%s:DSD or native stream detected enabling asrcmode in hardware",
__func__);
enable_asrc_mode(adev);
break;
} else if(((new_backend_idx == HEADPHONE_44_1_BACKEND) ||
(new_backend_idx == DSD_NATIVE_BACKEND)) &&
(usecase_backend_idx == HEADPHONE_BACKEND)) {
ALOGD("%s:48K stream detected, disabling and enabling it with asrcmode in hardware",
__func__);
disable_audio_route(adev, uc);
disable_snd_device(adev, uc->out_snd_device);
// Apply true-high-quality-mode if DSD or > 44.1KHz or >=24-bit
if (new_backend_idx == DSD_NATIVE_BACKEND)
audio_route_apply_and_update_path(adev->audio_route,
"hph-true-highquality-mode");
else if ((new_backend_idx == HEADPHONE_44_1_BACKEND) &&
(curr_out->bit_width >= 24))
audio_route_apply_and_update_path(adev->audio_route,
"hph-highquality-mode");
enable_asrc_mode(adev);
enable_snd_device(adev, uc->out_snd_device);
enable_audio_route(adev, uc);
break;
}
}
// reset split devices count
num_devices = 0;
}
if (adev->asrc_mode_enabled)
break;
}
}
}
}
int pcm_ioctl(struct pcm *pcm, int request, ...)
{
va_list ap;
void * arg;
int pcm_fd = *(int*)pcm;
va_start(ap, request);
arg = va_arg(ap, void *);
va_end(ap);
return ioctl(pcm_fd, request, arg);
}
int enable_audio_route(struct audio_device *adev,
struct audio_usecase *usecase)
{
snd_device_t snd_device;
char mixer_path[MIXER_PATH_MAX_LENGTH];
struct stream_out *out = NULL;
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;
#ifdef DS1_DOLBY_DAP_ENABLED
audio_extn_dolby_set_dmid(adev);
audio_extn_dolby_set_endpoint(adev);
#endif
audio_extn_dolby_ds2_set_endpoint(adev);
audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_BUSY);
audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_BUSY);
audio_extn_utils_send_app_type_cfg(adev, usecase);
audio_extn_utils_send_audio_calibration(adev, usecase);
if ((usecase->type == PCM_PLAYBACK) && is_offload_usecase(usecase->id)) {
out = usecase->stream.out;
if (out && out->compr)
audio_extn_utils_compress_set_clk_rec_mode(usecase);
}
strlcpy(mixer_path, use_case_table[usecase->id], MIXER_PATH_MAX_LENGTH);
platform_add_backend_name(mixer_path, snd_device, usecase);
ALOGD("%s: apply mixer and update path: %s", __func__, 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 || usecase->id == USECASE_INVALID)
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;
strlcpy(mixer_path, use_case_table[usecase->id], MIXER_PATH_MAX_LENGTH);
platform_add_backend_name(mixer_path, snd_device, usecase);
ALOGD("%s: reset and update mixer path: %s", __func__, mixer_path);
audio_route_reset_and_update_path(adev->audio_route, mixer_path);
audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_FREE);
audio_extn_listen_update_stream_status(usecase, LISTEN_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[SND_DEVICE_OUT_END];
char device_name[DEVICE_NAME_MAX_SIZE] = {0};
if (snd_device < SND_DEVICE_MIN ||
snd_device >= SND_DEVICE_MAX) {
ALOGE("%s: Invalid sound device %d", __func__, snd_device);
return -EINVAL;
}
adev->snd_dev_ref_cnt[snd_device]++;
if(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0 ) {
ALOGE("%s: Invalid sound device returned", __func__);
return -EINVAL;
}
if (adev->snd_dev_ref_cnt[snd_device] > 1) {
ALOGV("%s: snd_device(%d: %s) is already active",
__func__, snd_device, device_name);
return 0;
}
if (audio_extn_spkr_prot_is_enabled())
audio_extn_spkr_prot_calib_cancel(adev);
if (platform_can_enable_spkr_prot_on_device(snd_device) &&
audio_extn_spkr_prot_is_enabled()) {
if (platform_get_spkr_prot_acdb_id(snd_device) < 0) {
adev->snd_dev_ref_cnt[snd_device]--;
return -EINVAL;
}
audio_extn_dev_arbi_acquire(snd_device);
if (audio_extn_spkr_prot_start_processing(snd_device)) {
ALOGE("%s: spkr_start_processing failed", __func__);
audio_extn_dev_arbi_release(snd_device);
return -EINVAL;
}
} else if (platform_split_snd_device(adev->platform,
snd_device,
&num_devices,
new_snd_devices) == 0) {
for (i = 0; i < num_devices; i++) {
enable_snd_device(adev, new_snd_devices[i]);
}
} else {
ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
if ((SND_DEVICE_OUT_BT_A2DP == snd_device) &&
(audio_extn_a2dp_start_playback() < 0)) {
ALOGE(" fail to configure A2dp control path ");
return -EINVAL;
}
/* due to the possibility of calibration overwrite between listen
and audio, notify listen hal before audio calibration is sent */
audio_extn_sound_trigger_update_device_status(snd_device,
ST_EVENT_SND_DEVICE_BUSY);
audio_extn_listen_update_device_status(snd_device,
LISTEN_EVENT_SND_DEVICE_BUSY);
if (platform_get_snd_device_acdb_id(snd_device) < 0) {
adev->snd_dev_ref_cnt[snd_device]--;
audio_extn_sound_trigger_update_device_status(snd_device,
ST_EVENT_SND_DEVICE_FREE);
audio_extn_listen_update_device_status(snd_device,
LISTEN_EVENT_SND_DEVICE_FREE);
return -EINVAL;
}
audio_extn_dev_arbi_acquire(snd_device);
audio_route_apply_and_update_path(adev->audio_route, device_name);
if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
!adev->native_playback_enabled &&
audio_is_true_native_stream_active(adev)) {
ALOGD("%s: %d: napb: enabling native mode in hardware",
__func__, __LINE__);
audio_route_apply_and_update_path(adev->audio_route,
"true-native-mode");
adev->native_playback_enabled = true;
}
}
return 0;
}
int disable_snd_device(struct audio_device *adev,
snd_device_t snd_device)
{
int i, num_devices = 0;
snd_device_t new_snd_devices[SND_DEVICE_OUT_END];
char device_name[DEVICE_NAME_MAX_SIZE] = {0};
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;
}
adev->snd_dev_ref_cnt[snd_device]--;
if(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0) {
ALOGE("%s: Invalid sound device returned", __func__);
return -EINVAL;
}
if (adev->snd_dev_ref_cnt[snd_device] == 0) {
ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
if (platform_can_enable_spkr_prot_on_device(snd_device) &&
audio_extn_spkr_prot_is_enabled()) {
audio_extn_spkr_prot_stop_processing(snd_device);
} else if (platform_split_snd_device(adev->platform,
snd_device,
&num_devices,
new_snd_devices) == 0) {
for (i = 0; i < num_devices; i++) {
disable_snd_device(adev, new_snd_devices[i]);
}
} else {
audio_route_reset_and_update_path(adev->audio_route, device_name);
}
if (SND_DEVICE_OUT_BT_A2DP == snd_device)
audio_extn_a2dp_stop_playback();
if (snd_device == SND_DEVICE_OUT_HDMI || snd_device == SND_DEVICE_OUT_DISPLAY_PORT)
adev->is_channel_status_set = false;
else if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
adev->native_playback_enabled) {
ALOGD("%s: %d: napb: disabling native mode in hardware",
__func__, __LINE__);
audio_route_reset_and_update_path(adev->audio_route,
"true-native-mode");
adev->native_playback_enabled = false;
} else if (SND_DEVICE_OUT_HEADPHONES == snd_device &&
adev->asrc_mode_enabled) {
ALOGD("%s: %d: disabling asrc mode in hardware", __func__, __LINE__);
disable_asrc_mode(adev);
audio_route_apply_and_update_path(adev->audio_route, "hph-lowpower-mode");
}
audio_extn_dev_arbi_release(snd_device);
audio_extn_sound_trigger_update_device_status(snd_device,
ST_EVENT_SND_DEVICE_FREE);
audio_extn_listen_update_device_status(snd_device,
LISTEN_EVENT_SND_DEVICE_FREE);
}
return 0;
}
/*
legend:
uc - existing usecase
new_uc - new usecase
d1, d11, d2 - SND_DEVICE enums
a1, a2 - corresponding ANDROID device enums
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) B1
new_uc->dev d2 (a2) B1
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) B1
new_uc->dev d1 (a1), d2 (a2) B1
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 (a1) B1
resolution: no need to switch
*/
static snd_device_t derive_playback_snd_device(void * platform,
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_split_snd_device(platform,
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_usecases_codec_backend(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];
snd_device_t uc_derive_snd_device;
snd_device_t derive_snd_device[AUDIO_USECASE_MAX];
int i, num_uc_to_switch = 0;
int status = 0;
bool force_restart_session = false;
/*
* 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.
*/
/*
* This call is to check if we need to force routing for a particular stream
* If there is a backend configuration change for the device when a
* new stream starts, then ADM needs to be closed and re-opened with the new
* configuraion. This call check if we need to re-route all the streams
* associated with the backend. Touch tone + 24 bit + native playback.
*/
bool force_routing = platform_check_and_set_codec_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) &&
audio_extn_a2dp_is_force_device_switch()) {
force_routing = true;
force_restart_session = true;
}
ALOGD("%s:becf: force routing %d", __func__, force_routing);
/* 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);
ALOGD("%s:becf: (%d) check_usecases curr device: %s, usecase device:%s "
"backends match %d",__func__, i,
platform_get_snd_device_name(snd_device),
platform_get_snd_device_name(usecase->out_snd_device),
platform_check_backends_match(snd_device, usecase->out_snd_device));
if ((usecase->type != PCM_CAPTURE) && (usecase != uc_info)) {
uc_derive_snd_device = derive_playback_snd_device(adev->platform,
usecase, uc_info, snd_device);
if (((uc_derive_snd_device != usecase->out_snd_device) || force_routing) &&
((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
(usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) ||
(usecase->devices & AUDIO_DEVICE_OUT_USB_DEVICE) ||
(usecase->devices & AUDIO_DEVICE_OUT_ALL_A2DP) ||
(usecase->devices & AUDIO_DEVICE_OUT_ALL_SCO)) &&
((force_restart_session) ||
(platform_check_backends_match(snd_device, usecase->out_snd_device)))) {
ALOGD("%s:becf: check_usecases (%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;
/* Enable existing usecase on derived playback device */
derive_snd_device[usecase->id] = uc_derive_snd_device;
num_uc_to_switch++;
}
}
}
ALOGD("%s:becf: check_usecases num.of Usecases to switch %d", __func__,
num_uc_to_switch);
if (num_uc_to_switch) {
/* All streams have been de-routed. Disable the device */
/* Make sure the previous devices to be disabled first and then enable the
selected devices */
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);
}
}
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, derive_snd_device[usecase->id]);
}
}
/* 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 out_snd_device only before enabling the audio route */
if (switch_device[usecase->id]) {
usecase->out_snd_device = derive_snd_device[usecase->id];
if (usecase->type != VOICE_CALL) {
ALOGD("%s:becf: enabling usecase (%s) on (%s)", __func__,
use_case_table[usecase->id],
platform_get_snd_device_name(usecase->out_snd_device));
/* Update voc calibration before enabling VoIP route */
if (usecase->type == VOIP_CALL)
status = platform_switch_voice_call_device_post(adev->platform,
usecase->out_snd_device,
platform_get_input_snd_device(adev->platform, uc_info->devices));
enable_audio_route(adev, usecase);
}
}
}
}
}
static void check_usecases_capture_codec_backend(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;
int backend_check_cond = AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND;
int status = 0;
bool force_routing = platform_check_and_set_capture_codec_backend_cfg(adev, uc_info,
snd_device);
ALOGD("%s:becf: force routing %d", __func__, force_routing);
/*
* Make sure out devices is checked against out codec backend device and
* also in devices against in codec backend. Checking out device against in
* codec backend or vice versa causes issues.
*/
if (uc_info->type == PCM_CAPTURE)
backend_check_cond = AUDIO_DEVICE_IN_ALL_CODEC_BACKEND;
/*
* 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);
/*
* TODO: Enhance below condition to handle BT sco/USB multi recording
*/
if (usecase->type != PCM_PLAYBACK &&
usecase != uc_info &&
(usecase->in_snd_device != snd_device || force_routing) &&
((uc_info->devices & backend_check_cond) &&
(((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND) ||
(usecase->type == VOIP_CALL))) &&
(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) {
/* All streams have been de-routed. Disable the device */
/* Make sure the previous devices to be disabled first and then enable the
selected devices */
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;
if (usecase->type != VOICE_CALL) {
/* Update voc calibration before enabling VoIP route */
if (usecase->type == VOIP_CALL)
status = platform_switch_voice_call_device_post(adev->platform,
usecase->out_snd_device,
usecase->in_snd_device);
enable_audio_route(adev, usecase);
}
}
}
}
}
static void reset_hdmi_sink_caps(struct stream_out *out) {
int i = 0;
for (i = 0; i<= MAX_SUPPORTED_CHANNEL_MASKS; i++) {
out->supported_channel_masks[i] = 0;
}
for (i = 0; i<= MAX_SUPPORTED_FORMATS; i++) {
out->supported_formats[i] = 0;
}
for (i = 0; i<= MAX_SUPPORTED_SAMPLE_RATES; i++) {
out->supported_sample_rates[i] = 0;
}
}
/* must be called with hw device mutex locked */
static int read_hdmi_sink_caps(struct stream_out *out)
{
int ret = 0, i = 0, j = 0;
int channels = platform_edid_get_max_channels(out->dev->platform);
reset_hdmi_sink_caps(out);
/* Cache ext disp type */
if (platform_get_ext_disp_type(adev->platform) <= 0) {
ALOGE("%s: Failed to query disp type, ret:%d", __func__, ret);
return -EINVAL;
}
switch (channels) {
case 8:
ALOGV("%s: HDMI supports 7.1 channels", __func__);
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_7POINT1;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_6POINT1;
case 6:
ALOGV("%s: HDMI supports 5.1 channels", __func__);
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_5POINT1;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_PENTA;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_QUAD;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_SURROUND;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_2POINT1;
break;
default:
ALOGE("invalid/nonstandard channal count[%d]",channels);
ret = -ENOSYS;
break;
}
// check channel format caps
i = 0;
if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_AC3)) {
ALOGV(":%s HDMI supports AC3/EAC3 formats", __func__);
out->supported_formats[i++] = AUDIO_FORMAT_AC3;
//Adding EAC3/EAC3_JOC formats if AC3 is supported by the sink.
//EAC3/EAC3_JOC will be converted to AC3 for decoding if needed
out->supported_formats[i++] = AUDIO_FORMAT_E_AC3;
out->supported_formats[i++] = AUDIO_FORMAT_E_AC3_JOC;
}
if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_DOLBY_TRUEHD)) {
ALOGV(":%s HDMI supports TRUE HD format", __func__);
out->supported_formats[i++] = AUDIO_FORMAT_DOLBY_TRUEHD;
}
if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_DTS)) {
ALOGV(":%s HDMI supports DTS format", __func__);
out->supported_formats[i++] = AUDIO_FORMAT_DTS;
}
if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_DTS_HD)) {
ALOGV(":%s HDMI supports DTS HD format", __func__);
out->supported_formats[i++] = AUDIO_FORMAT_DTS_HD;
}
// check sample rate caps
i = 0;
for (j = 0; j < MAX_SUPPORTED_SAMPLE_RATES; j++) {
if (platform_is_edid_supported_sample_rate(out->dev->platform, out_hdmi_sample_rates[j])) {
ALOGV(":%s HDMI supports sample rate:%d", __func__, out_hdmi_sample_rates[j]);
out->supported_sample_rates[i++] = out_hdmi_sample_rates[j];
}
}
return ret;
}
audio_usecase_t get_usecase_id_from_usecase_type(const struct audio_device *adev,
usecase_type_t type)
{
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 == type) {
ALOGV("%s: usecase id %d", __func__, usecase->id);
return usecase->id;
}
}
return USECASE_INVALID;
}
struct audio_usecase *get_usecase_from_list(const 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;
}
struct stream_in *get_next_active_input(const struct audio_device *adev)
{
struct audio_usecase *usecase;
struct listnode *node;
list_for_each_reverse(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_CAPTURE)
return usecase->stream.in;
}
return NULL;
}
/*
* is a true native playback active
*/
bool audio_is_true_native_stream_active(struct audio_device *adev)
{
bool active = false;
int i = 0;
struct listnode *node;
if (NATIVE_AUDIO_MODE_TRUE_44_1 != platform_get_native_support()) {
ALOGV("%s:napb: not in true mode or non hdphones device",
__func__);
active = false;
goto exit;
}
list_for_each(node, &adev->usecase_list) {
struct audio_usecase *uc;
uc = node_to_item(node, struct audio_usecase, list);
struct stream_out *curr_out =
(struct stream_out*) uc->stream.out;
if (curr_out && PCM_PLAYBACK == uc->type) {
ALOGD("%s:napb: (%d) (%s)id (%d) sr %d bw "
"(%d) device %s", __func__, i++, use_case_table[uc->id],
uc->id, curr_out->sample_rate,
curr_out->bit_width,
platform_get_snd_device_name(uc->out_snd_device));
if (is_offload_usecase(uc->id) &&
(curr_out->sample_rate == OUTPUT_SAMPLING_RATE_44100)) {
active = true;
ALOGD("%s:napb:native stream detected", __func__);
}
}
}
exit:
return active;
}
/*
* if native DSD playback active
*/
bool audio_is_dsd_native_stream_active(struct audio_device *adev)
{
bool active = false;
struct listnode *node = NULL;
struct audio_usecase *uc = NULL;
struct stream_out *curr_out = NULL;
list_for_each(node, &adev->usecase_list) {
uc = node_to_item(node, struct audio_usecase, list);
curr_out = (struct stream_out*) uc->stream.out;
if (curr_out && PCM_PLAYBACK == uc->type &&
(DSD_NATIVE_BACKEND == platform_get_backend_index(uc->out_snd_device))) {
active = true;
ALOGV("%s:DSD playback is active", __func__);
break;
}
}
return active;
}
static bool force_device_switch(struct audio_usecase *usecase)
{
bool ret = false;
bool is_it_true_mode = false;
if (is_offload_usecase(usecase->id) &&
(usecase->stream.out) &&
(usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) &&
(usecase->stream.out->devices == AUDIO_DEVICE_OUT_WIRED_HEADSET ||
usecase->stream.out->devices == AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) {
is_it_true_mode = (NATIVE_AUDIO_MODE_TRUE_44_1 == platform_get_native_support()? true : false);
if ((is_it_true_mode && !adev->native_playback_enabled) ||
(!is_it_true_mode && adev->native_playback_enabled)){
ret = true;
ALOGD("napb: time to toggle native mode");
}
}
// 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) &&
(usecase->stream.out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) &&
audio_extn_a2dp_is_force_device_switch()) {
ALOGD("Force a2dp device switch to update new encoder config");
ret = true;
}
return ret;
}
int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
{
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 *voip_usecase = NULL;
struct audio_usecase *hfp_usecase = NULL;
audio_usecase_t hfp_ucid;
int status = 0;
ALOGD("%s for use case (%s)", __func__, use_case_table[uc_id]);
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 == VOIP_CALL) ||
(usecase->type == PCM_HFP_CALL)) {
if(usecase->stream.out == NULL) {
ALOGE("%s: stream.out is NULL", __func__);
return -EINVAL;
}
out_snd_device = platform_get_output_snd_device(adev->platform,
usecase->stream.out);
in_snd_device = platform_get_input_snd_device(adev->platform, 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_usecases_codec_backend() is called below.
* choose voice call device only if the use case device is
* also using the codec backend
*/
if (voice_is_in_call(adev) && adev->mode != AUDIO_MODE_NORMAL) {
vc_usecase = get_usecase_from_list(adev,
get_usecase_id_from_usecase_type(adev, VOICE_CALL));
if ((vc_usecase) && (((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
(usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) ||
((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
(usecase->devices & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) ||
(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 (voice_extn_compress_voip_is_active(adev)) {
bool out_snd_device_backend_match = true;
voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
if (usecase->stream.out != NULL) {
out_snd_device_backend_match = platform_check_backends_match(
voip_usecase->out_snd_device,
platform_get_output_snd_device(
adev->platform,
usecase->stream.out));
}
if ((voip_usecase) && ((voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) &&
out_snd_device_backend_match &&
(voip_usecase->stream.out != adev->primary_output))) {
in_snd_device = voip_usecase->in_snd_device;
out_snd_device = voip_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) && (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) {
if (usecase->stream.out == NULL) {
ALOGE("%s: stream.out is NULL", __func__);
return -EINVAL;
}
usecase->devices = usecase->stream.out->devices;
in_snd_device = SND_DEVICE_NONE;
if (out_snd_device == SND_DEVICE_NONE) {
out_snd_device = platform_get_output_snd_device(adev->platform,
usecase->stream.out);
if (usecase->stream.out == adev->primary_output &&
adev->active_input &&
out_snd_device != usecase->out_snd_device) {
select_devices(adev, adev->active_input->usecase);
}
}
} else if (usecase->type == PCM_CAPTURE) {
if (usecase->stream.in == NULL) {
ALOGE("%s: stream.in is NULL", __func__);
return -EINVAL;
}
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;
if (adev->active_input &&
(adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
(adev->mode == AUDIO_MODE_IN_COMMUNICATION &&
adev->active_input->source == AUDIO_SOURCE_MIC)) &&
adev->primary_output && !adev->primary_output->standby) {
out_device = adev->primary_output->devices;
platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
} else if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) {
out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX;
}
in_snd_device = platform_get_input_snd_device(adev->platform, out_device);
}
}
}
if (out_snd_device == usecase->out_snd_device &&
in_snd_device == usecase->in_snd_device) {
if (!force_device_switch(usecase))
return 0;
}
ALOGD("%s: out_snd_device(%d: %s) in_snd_device(%d: %s)", __func__,
out_snd_device, platform_get_snd_device_name(out_snd_device),
in_snd_device, platform_get_snd_device_name(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);
}
if (((usecase->type == VOICE_CALL) ||
(usecase->type == VOIP_CALL)) &&
(usecase->out_snd_device != SND_DEVICE_NONE)) {
/* Disable sidetone only if voice/voip call already exists */
if (voice_is_call_state_active(adev) ||
voice_extn_compress_voip_is_started(adev))
voice_set_sidetone(adev, usecase->out_snd_device, false);
/* Disable aanc only if voice call exists */
if (voice_is_call_state_active(adev))
voice_check_and_update_aanc_path(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) {
check_usecases_codec_backend(adev, usecase, out_snd_device);
if (platform_check_codec_asrc_support(adev->platform))
check_and_set_asrc_mode(adev, usecase, out_snd_device);
enable_snd_device(adev, out_snd_device);
}
if (in_snd_device != SND_DEVICE_NONE) {
check_usecases_capture_codec_backend(adev, usecase, in_snd_device);
enable_snd_device(adev, in_snd_device);
}
if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) {
status = platform_switch_voice_call_device_post(adev->platform,
out_snd_device,
in_snd_device);
enable_audio_route_for_voice_usecases(adev, usecase);
}
usecase->in_snd_device = in_snd_device;
usecase->out_snd_device = out_snd_device;
audio_extn_utils_update_stream_app_type_cfg_for_usecase(adev,
usecase);
if (usecase->type == PCM_PLAYBACK) {
if ((24 == usecase->stream.out->bit_width) &&
(usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
} else if ((out_snd_device == SND_DEVICE_OUT_HDMI ||
out_snd_device == SND_DEVICE_OUT_USB_HEADSET ||
out_snd_device == SND_DEVICE_OUT_DISPLAY_PORT) &&
(usecase->stream.out->sample_rate >= OUTPUT_SAMPLING_RATE_44100)) {
/*
* To best utlize DSP, check if the stream sample rate is supported/multiple of
* configured device sample rate, if not update the COPP rate to be equal to the
* device sample rate, else open COPP at stream sample rate
*/
platform_check_and_update_copp_sample_rate(adev->platform, out_snd_device,
usecase->stream.out->sample_rate,
&usecase->stream.out->app_type_cfg.sample_rate);
} else if (((out_snd_device != SND_DEVICE_OUT_HEADPHONES_44_1 &&
!audio_is_true_native_stream_active(adev)) &&
usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) ||
(usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) {
usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
}
/* Notify device change info to effect clients registered */
pthread_mutex_unlock(&adev->lock);
audio_extn_gef_notify_device_config(
usecase->stream.out->devices,
usecase->stream.out->channel_mask,
usecase->stream.out->app_type_cfg.sample_rate,
platform_get_snd_device_acdb_id(usecase->out_snd_device));
pthread_mutex_lock(&adev->lock);
}
enable_audio_route(adev, usecase);
if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) {
/* Enable aanc only if voice call exists */
if (voice_is_call_state_active(adev))
voice_check_and_update_aanc_path(adev, out_snd_device, true);
/* Enable sidetone only if other voice/voip call already exists */
if (voice_is_call_state_active(adev) ||
voice_extn_compress_voip_is_started(adev))
voice_set_sidetone(adev, out_snd_device, true);
}
/* 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);
ALOGD("%s: done",__func__);
return status;
}
static int stop_input_stream(struct stream_in *in)
{
int ret = 0;
struct audio_usecase *uc_info;
struct audio_device *adev = in->dev;
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;
}
/* 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);
adev->active_input = get_next_active_input(adev);
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;
int snd_card_status = get_snd_card_state(adev);
int usecase = platform_update_usecase_from_source(in->source,in->usecase);
if (get_usecase_from_list(adev, usecase) == NULL)
in->usecase = usecase;
ALOGD("%s: enter: stream(%p)usecase(%d: %s)",
__func__, &in->stream, in->usecase, use_case_table[in->usecase]);
if (SND_CARD_STATE_OFFLINE == snd_card_status) {
ALOGE("%s: sound card is not active/SSR returning error", __func__);
ret = -EIO;
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);
if (get_usecase_from_list(adev, in->usecase) != NULL) {
ALOGE("%s: use case assigned already in use, stream(%p)usecase(%d: %s)",
__func__, &in->stream, in->usecase, use_case_table[in->usecase]);
return -EINVAL;
}
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;
}
adev->active_input = in;
uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
if (!uc_info) {
ret = -ENOMEM;
goto error_config;
}
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_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
adev->perf_lock_opts,
adev->perf_lock_opts_size);
select_devices(adev, in->usecase);
ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d format %d",
__func__, adev->snd_card, in->pcm_device_id, in->config.channels, in->config.format);
if (audio_extn_cin_attached_usecase(in->usecase)) {
ret = audio_extn_cin_start_input_stream(in);
if (ret)
goto error_open;
else
goto done_open;
}
unsigned int flags = PCM_IN;
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;
}
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;
}
register_in_stream(in);
if (in->realtime) {
ret = pcm_start(in->pcm);
if (ret < 0)
goto error_open;
}
done_open:
audio_extn_perf_lock_release(&adev->perf_lock_handle);
ALOGD("%s: exit", __func__);
return ret;
error_open:
audio_extn_perf_lock_release(&adev->perf_lock_handle);
stop_input_stream(in);
error_config:
adev->active_input = get_next_active_input(adev);
/*
* sleep 50ms to allow sufficient time for kernel
* drivers to recover incases like SSR.
*/
usleep(50000);
ALOGD("%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));
if (!cmd) {
ALOGE("failed to allocate mem for command 0x%x", command);
return -ENOMEM;
}
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);
}
}
}
bool is_offload_usecase(audio_usecase_t uc_id)
{
unsigned int i;
for (i = 0; i < sizeof(offload_usecases)/sizeof(offload_usecases[0]); i++) {
if (uc_id == offload_usecases[i])
return true;
}
return false;
}
static audio_usecase_t get_offload_usecase(struct audio_device *adev, bool is_compress)
{
audio_usecase_t ret_uc = USECASE_INVALID;
unsigned int offload_uc_index;
unsigned int num_usecase = sizeof(offload_usecases)/sizeof(offload_usecases[0]);
if (!adev->multi_offload_enable) {
if (!is_compress)
ret_uc = USECASE_AUDIO_PLAYBACK_OFFLOAD2;
else
ret_uc = USECASE_AUDIO_PLAYBACK_OFFLOAD;
pthread_mutex_lock(&adev->lock);
if (get_usecase_from_list(adev, ret_uc) != NULL)
ret_uc = USECASE_INVALID;
pthread_mutex_unlock(&adev->lock);
return ret_uc;
}
ALOGV("%s: num_usecase: %d", __func__, num_usecase);
for (offload_uc_index = 0; offload_uc_index < num_usecase; offload_uc_index++) {
if (!(adev->offload_usecases_state & (0x1 << offload_uc_index))) {
adev->offload_usecases_state |= 0x1 << offload_uc_index;
ret_uc = offload_usecases[offload_uc_index];
break;
}
}
ALOGV("%s: offload usecase is %d", __func__, ret_uc);
return ret_uc;
}
static void free_offload_usecase(struct audio_device *adev,
audio_usecase_t uc_id)
{
unsigned int offload_uc_index;
unsigned int num_usecase = sizeof(offload_usecases)/sizeof(offload_usecases[0]);
if (!adev->multi_offload_enable)
return;
for (offload_uc_index = 0; offload_uc_index < num_usecase; offload_uc_index++) {
if (offload_usecases[offload_uc_index] == uc_id) {
adev->offload_usecases_state &= ~(0x1 << offload_uc_index);
break;
}
}
ALOGV("%s: free offload usecase %d", __func__, uc_id);
}
static void *offload_thread_loop(void *context)
{
struct stream_out *out = (struct stream_out *) context;
struct listnode *item;
int ret = 0;
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);
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:
ALOGD("copl(%p):calling compress_wait", out);
compress_wait(out->compr, -1);
ALOGD("copl(%p):out of compress_wait", out);
send_callback = true;
event = STREAM_CBK_EVENT_WRITE_READY;
break;
case OFFLOAD_CMD_PARTIAL_DRAIN:
ret = compress_next_track(out->compr);
if(ret == 0) {
ALOGD("copl(%p):calling compress_partial_drain", out);
ret = compress_partial_drain(out->compr);
ALOGD("copl(%p):out of compress_partial_drain", out);
if (ret < 0)
ret = -errno;
}
else if (ret == -ETIMEDOUT)
compress_drain(out->compr);
else
ALOGE("%s: Next track returned error %d",__func__, ret);
if (ret != -ENETRESET) {
send_callback = true;
pthread_mutex_lock(&out->lock);
out->send_new_metadata = 1;
out->send_next_track_params = true;
pthread_mutex_unlock(&out->lock);
event = STREAM_CBK_EVENT_DRAIN_READY;
ALOGV("copl(%p):send drain callback, ret %d", out, ret);
} else
ALOGE("%s: Block drain ready event during SSR", __func__);
break;
case OFFLOAD_CMD_DRAIN:
ALOGD("copl(%p):calling compress_drain", out);
compress_drain(out->compr);
ALOGD("copl(%p):calling compress_drain", out);
send_callback = true;
event = STREAM_CBK_EVENT_DRAIN_READY;
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 && out->client_callback) {
ALOGVV("%s: sending client_callback event %d", __func__, event);
out->client_callback(event, NULL, out->client_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 int stop_output_stream(struct stream_out *out)
{
int 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 (is_offload_usecase(out->usecase) &&
!(audio_extn_passthru_is_passthrough_stream(out))) {
if (adev->visualizer_stop_output != NULL)
adev->visualizer_stop_output(out->handle, out->pcm_device_id);
audio_extn_dts_remove_state_notifier_node(out->usecase);
if (adev->offload_effects_stop_output != NULL)
adev->offload_effects_stop_output(out->handle, out->pcm_device_id);
}
/* 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);
free(uc_info);
if (is_offload_usecase(out->usecase) &&
(audio_extn_passthru_is_passthrough_stream(out))) {
ALOGV("Disable passthrough , reset mixer to pcm");
/* NO_PASSTHROUGH */
out->compr_config.codec->compr_passthr = 0;
audio_extn_passthru_on_stop(out);
audio_extn_dolby_set_dap_bypass(adev, DAP_STATE_ON);
}
/* Must be called after removing the usecase from list */
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
audio_extn_keep_alive_start();
/*reset delay_param to 0*/
out->delay_param.start_delay = 0;
ALOGV("%s: exit: status(%d)", __func__, ret);
return ret;
}
int start_output_stream(struct stream_out *out)
{
int ret = 0;
struct audio_usecase *uc_info;
struct audio_device *adev = out->dev;
int snd_card_status = get_snd_card_state(adev);
char mixer_ctl_name[128];
struct mixer_ctl *ctl = NULL;
char* perf_mode[] = {"ULL", "ULL_PP", "LL"};
if ((out->usecase < 0) || (out->usecase >= AUDIO_USECASE_MAX)) {
ret = -EINVAL;
goto error_config;
}
ALOGD("%s: enter: stream(%p)usecase(%d: %s) devices(%#x)",
__func__, &out->stream, out->usecase, use_case_table[out->usecase],
out->devices);
if (SND_CARD_STATE_OFFLINE == snd_card_status) {
ALOGE("%s: sound card is not active/SSR returning error", __func__);
ret = -EIO;
goto error_config;
}
if (out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) {
if (!audio_extn_a2dp_is_ready()) {
if (out->devices & AUDIO_DEVICE_OUT_SPEAKER) {
//combo usecase just by pass a2dp
ALOGW("%s: A2DP profile is not ready, route it to speaker", __func__);
out->devices = AUDIO_DEVICE_OUT_SPEAKER;
} else {
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_open;
}
uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
if (!uc_info) {
ret = -ENOMEM;
goto error_config;
}
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;
list_add_tail(&adev->usecase_list, &uc_info->list);
audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
adev->perf_lock_opts,
adev->perf_lock_opts_size);
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
audio_extn_keep_alive_stop();
if (audio_extn_passthru_is_enabled() &&
audio_extn_passthru_is_passthrough_stream(out)) {
audio_extn_passthru_on_start(out);
audio_extn_passthru_update_stream_configuration(adev, out);
}
}
select_devices(adev, out->usecase);
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 (!is_offload_usecase(out->usecase)) {
unsigned int flags = PCM_OUT;
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;
} else
flags |= PCM_MONOTONIC;
if ((adev->vr_audio_mode_enabled) &&
(out->flags & AUDIO_OUTPUT_FLAG_RAW)) {
snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
"PCM_Dev %d Topology", out->pcm_device_id);
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGI("%s: Could not get ctl for mixer cmd might be ULL - %s",
__func__, mixer_ctl_name);
} else {
//if success use ULLPP
ALOGI("%s: mixer ctrl %s succeeded setting up ULL for %d",
__func__, mixer_ctl_name, out->pcm_device_id);
//There is a still a possibility that some sessions
// that request for FAST|RAW when 3D audio is active
//can go through ULLPP. Ideally we expects apps to
//listen to audio focus and stop concurrent playback
//Also, we will look for mode flag (voice_in_communication)
//before enabling the realtime flag.
mixer_ctl_set_enum_by_string(ctl, perf_mode[1]);
}
}
while (1) {
out->pcm = pcm_open(adev->snd_card, out->pcm_device_id,
flags, &out->config);
if (out->pcm == NULL || !pcm_is_ready(out->pcm)) {
ALOGE("%s: %s", __func__, pcm_get_error(out->pcm));
if (out->pcm != NULL) {
pcm_close(out->pcm);
out->pcm = NULL;
}
if (pcm_open_retry_count-- == 0) {
ret = -EIO;
goto error_open;
}
usleep(PROXY_OPEN_WAIT_TIME * 1000);
continue;
}
break;
}
platform_set_stream_channel_map(adev->platform, out->channel_mask,
out->pcm_device_id);
ALOGV("%s: pcm_prepare", __func__);
if (pcm_is_ready(out->pcm)) {
ret = pcm_prepare(out->pcm);
if (ret < 0) {
ALOGE("%s: pcm_prepare returned %d", __func__, ret);
pcm_close(out->pcm);
out->pcm = NULL;
goto error_open;
}
}
platform_set_stream_channel_map(adev->platform, out->channel_mask,
out->pcm_device_id);
} else {
platform_set_stream_channel_map(adev->platform, out->channel_mask,
out->pcm_device_id);
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;
}
/* compress_open sends params of the track, so reset the flag here */
out->is_compr_metadata_avail = false;
if (out->client_callback)
compress_nonblock(out->compr, out->non_blocking);
/* Since small bufs uses blocking writes, a write will be blocked
for the default max poll time (20s) in the event of an SSR.
Reduce the poll time to observe and deal with SSR faster.
*/
if (!out->non_blocking) {
compress_set_max_poll_wait(out->compr, 1000);
}
audio_extn_utils_compress_set_render_mode(out);
audio_extn_utils_compress_set_clk_rec_mode(uc_info);
/* set render window if it was set before compress_open() */
if (out->render_window.render_ws != 0 && out->render_window.render_we != 0)
audio_extn_utils_compress_set_render_window(out,
&out->render_window);
audio_extn_utils_compress_set_start_delay(out, &out->delay_param);
audio_extn_dts_create_state_notifier_node(out->usecase);
audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
popcount(out->channel_mask),
out->playback_started);
#ifdef DS1_DOLBY_DDP_ENABLED
if (audio_extn_is_dolby_format(out->format))
audio_extn_dolby_send_ddp_endp_params(adev);
#endif
if (!(audio_extn_passthru_is_passthrough_stream(out)) &&
(out->sample_rate != 176400 && out->sample_rate <= 192000)) {
if (adev->visualizer_start_output != NULL)
adev->visualizer_start_output(out->handle, out->pcm_device_id);
if (adev->offload_effects_start_output != NULL)
adev->offload_effects_start_output(out->handle, out->pcm_device_id, adev->mixer);
audio_extn_check_and_set_dts_hpx_state(adev);
}
}
if (ret == 0) {
register_out_stream(out);
if (out->realtime) {
ret = pcm_start(out->pcm);
if (ret < 0)
goto error_open;
}
}
audio_extn_perf_lock_release(&adev->perf_lock_handle);
ALOGD("%s: exit", __func__);
return ret;
error_open:
audio_extn_perf_lock_release(&adev->perf_lock_handle);
stop_output_stream(out);
error_config:
/*
* sleep 50ms to allow sufficient time for kernel
* drivers to recover incases like SSR.
*/
usleep(50000);
return ret;
}
static int check_input_parameters(uint32_t sample_rate,
audio_format_t format,
int channel_count)
{
int ret = 0;
if (((format != AUDIO_FORMAT_PCM_16_BIT) && (format != AUDIO_FORMAT_PCM_8_24_BIT) &&
(format != AUDIO_FORMAT_PCM_24_BIT_PACKED) && (format != AUDIO_FORMAT_PCM_32_BIT) &&
(format != AUDIO_FORMAT_PCM_FLOAT)) &&
!voice_extn_compress_voip_is_format_supported(format) &&
!audio_extn_compr_cap_format_supported(format)) ret = -EINVAL;
switch (channel_count) {
case 1:
case 2:
case 3:
case 4:
case 6:
break;
default:
ret = -EINVAL;
}
switch (sample_rate) {
case 8000:
case 11025:
case 12000:
case 16000:
case 22050:
case 24000:
case 32000:
case 44100:
case 48000:
case 96000:
case 192000:
break;
default:
ret = -EINVAL;
}
return ret;
}
static size_t get_input_buffer_size(uint32_t sample_rate,
audio_format_t format,
int channel_count,
bool is_low_latency)
{
size_t size = 0;
if (check_input_parameters(sample_rate, format, channel_count) != 0)
return 0;
size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000;
if (is_low_latency)
size = configured_low_latency_capture_period_size;
size *= audio_bytes_per_sample(format) * channel_count;
/* make sure the size is multiple of 32 bytes
* At 48 kHz mono 16-bit PCM:
* 5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15)
* 3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10)
*/
size += 0x1f;
size &= ~0x1f;
return size;
}
static size_t get_output_period_size(uint32_t sample_rate,
audio_format_t format,
int channel_count,
int duration /*in millisecs*/)
{
size_t size = 0;
uint32_t bytes_per_sample = audio_bytes_per_sample(format);
if ((duration == 0) || (sample_rate == 0) ||
(bytes_per_sample == 0) || (channel_count == 0)) {
ALOGW("Invalid config duration %d sr %d bps %d ch %d", duration, sample_rate,
bytes_per_sample, channel_count);
return -EINVAL;
}
size = (sample_rate *
duration *
bytes_per_sample *
channel_count) / 1000;
/*
* To have same PCM samples for all channels, the buffer size requires to
* be multiple of (number of channels * bytes per sample)
* For writes to succeed, the buffer must be written at address which is multiple of 32
*/
size = ALIGN(size, (bytes_per_sample * channel_count * 32));
return (size/(channel_count * bytes_per_sample));
}
static uint64_t get_actual_pcm_frames_rendered(struct stream_out *out)
{
uint64_t actual_frames_rendered = 0;
size_t kernel_buffer_size = out->compr_config.fragment_size * out->compr_config.fragments;
/* This adjustment accounts for buffering after app processor.
* It is based on estimated DSP latency per use case, rather than exact.
*/
int64_t platform_latency = platform_render_latency(out->usecase) *
out->sample_rate / 1000000LL;
/* not querying actual state of buffering in kernel as it would involve an ioctl call
* which then needs protection, this causes delay in TS query for pcm_offload usecase
* hence only estimate.
*/
int64_t signed_frames = out->written - kernel_buffer_size;
signed_frames = signed_frames / (audio_bytes_per_sample(out->format) * popcount(out->channel_mask)) - platform_latency;
if (signed_frames > 0)
actual_frames_rendered = signed_frames;
ALOGVV("%s signed frames %lld out_written %lld kernel_buffer_size %d"
"bytes/sample %zu channel count %d", __func__,(long long int)signed_frames,
(long long int)out->written, (int)kernel_buffer_size,
audio_bytes_per_sample(out->compr_config.codec->format),
popcount(out->channel_mask));
return actual_frames_rendered;
}
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->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
return out->compr_config.fragment_size;
else if(out->usecase == USECASE_COMPRESS_VOIP_CALL)
return voice_extn_compress_voip_out_get_buffer_size(out);
else if (is_offload_usecase(out->usecase) &&
out->flags == AUDIO_OUTPUT_FLAG_DIRECT)
return out->hal_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;
}
static int out_standby(struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
stream, out->usecase, use_case_table[out->usecase]);
lock_output_stream(out);
if (!out->standby) {
if (adev->adm_deregister_stream)
adev->adm_deregister_stream(adev->adm_data, out->handle);
if (is_offload_usecase(out->usecase))
stop_compressed_output_l(out);
pthread_mutex_lock(&adev->lock);
out->standby = true;
if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
voice_extn_compress_voip_close_output_stream(stream);
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&out->lock);
ALOGD("VOIP output entered standby");
return 0;
} else if (!is_offload_usecase(out->usecase)) {
if (out->pcm) {
pcm_close(out->pcm);
out->pcm = NULL;
}
} else {
ALOGD("copl(%p):standby", out);
out->send_next_track_params = false;
out->is_compr_metadata_avail = false;
out->gapless_mdata.encoder_delay = 0;
out->gapless_mdata.encoder_padding = 0;
if (out->compr != NULL) {
compress_close(out->compr);
out->compr = NULL;
}
}
stop_output_stream(out);
pthread_mutex_unlock(&adev->lock);
}
pthread_mutex_unlock(&out->lock);
ALOGD("%s: exit", __func__);
return 0;
}
static int out_dump(const struct audio_stream *stream __unused,
int fd __unused)
{
return 0;
}
static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms)
{
int ret = 0;
char value[32];
if (!out || !parms) {
ALOGE("%s: return invalid ",__func__);
return -EINVAL;
}
ret = audio_extn_parse_compress_metadata(out, parms);
ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value));
if (ret >= 0) {
out->gapless_mdata.encoder_delay = atoi(value); //whats a good limit check?
}
ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value));
if (ret >= 0) {
out->gapless_mdata.encoder_padding = atoi(value);
}
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 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 str_parms *parms;
char value[32];
int ret = 0, val = 0, err;
ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s",
__func__, out->usecase, use_case_table[out->usecase], kvpairs);
parms = str_parms_create_str(kvpairs);
if (!parms)
goto error;
err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
if (err >= 0) {
val = atoi(value);
lock_output_stream(out);
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) &&
!audio_extn_passthru_is_passthrough_stream(out) &&
(platform_get_edid_info(adev->platform) != 0) /* HDMI disconnected */) {
val = AUDIO_DEVICE_OUT_SPEAKER;
}
/*
* 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_ALL_A2DP) &&
(val == AUDIO_DEVICE_NONE)) {
val = AUDIO_DEVICE_OUT_SPEAKER;
}
/* 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) {
//combo usecase just by pass a2dp
ALOGW("%s: A2DP profile is not ready,routing to speaker only", __func__);
val = AUDIO_DEVICE_OUT_SPEAKER;
} 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);
goto error;
}
}
}
/*
* select_devices() call below switches all the usecases on the same
* backend to the new device. Refer to check_usecases_codec_backend() 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 (val != 0) {
audio_devices_t new_dev = val;
bool same_dev = out->devices == new_dev;
out->devices = new_dev;
if (output_drives_call(adev, out)) {
if(!voice_is_in_call(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) {
if (!same_dev) {
ALOGV("update routing change");
audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0,
adev->perf_lock_opts,
adev->perf_lock_opts_size);
if (adev->adm_on_routing_change)
adev->adm_on_routing_change(adev->adm_data,
out->handle);
}
select_devices(adev, out->usecase);
if (!same_dev)
audio_extn_perf_lock_release(&adev->perf_lock_handle);
}
}
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&out->lock);
}
if (out == adev->primary_output) {
pthread_mutex_lock(&adev->lock);
audio_extn_set_parameters(adev, parms);
pthread_mutex_unlock(&adev->lock);
}
if (is_offload_usecase(out->usecase)) {
lock_output_stream(out);
parse_compress_metadata(out, parms);
audio_extn_dts_create_state_notifier_node(out->usecase);
audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
popcount(out->channel_mask),
out->playback_started);
pthread_mutex_unlock(&out->lock);
}
err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_PROFILE, value, sizeof(value));
if (err >= 0) {
strlcpy(out->profile, value, sizeof(out->profile));
ALOGV("updating stream profile with value '%s'", out->profile);
lock_output_stream(out);
audio_extn_utils_update_stream_output_app_type_cfg(adev->platform,
&adev->streams_output_cfg_list,
out->devices, out->flags, out->format,
out->sample_rate, out->bit_width,
out->channel_mask, out->profile,
&out->app_type_cfg);
pthread_mutex_unlock(&out->lock);
}
str_parms_destroy(parms);
error:
ALOGV("%s: exit: code(%d)", __func__, ret);
return ret;
}
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 = (char*) NULL;
char value[256];
struct str_parms *reply = str_parms_create();
size_t i, j;
int ret;
bool first = true;
if (!query || !reply) {
if (reply) {
str_parms_destroy(reply);
}
if (query) {
str_parms_destroy(query);
}
ALOGE("out_get_parameters: failed to allocate mem for query or reply");
return NULL;
}
ALOGV("%s: enter: keys - %s", __func__, keys);
ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value));
if (ret >= 0) {
value[0] = '\0';
i = 0;
while (out->supported_channel_masks[i] != 0) {
for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) {
if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) {
if (!first) {
strlcat(value, "|", sizeof(value));
}
strlcat(value, out_channels_name_to_enum_table[j].name, sizeof(value));
first = false;
break;
}
}
i++;
}
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value);
str = str_parms_to_str(reply);
} else {
voice_extn_out_get_parameters(out, query, reply);
str = str_parms_to_str(reply);
if (str && !strncmp(str, "", sizeof(""))) {
free(str);
str = strdup(keys);
}
}
ret = str_parms_get_str(query, "is_direct_pcm_track", value, sizeof(value));
if (ret >= 0) {
value[0] = '\0';
if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT &&
!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
ALOGV("in direct_pcm");
strlcat(value, "true", sizeof(value ));
} else {
ALOGV("not in direct_pcm");
strlcat(value, "false", sizeof(value));
}
str_parms_add_str(reply, "is_direct_pcm_track", value);
if (str)
free(str);
str = str_parms_to_str(reply);
}
ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value, sizeof(value));
if (ret >= 0) {
value[0] = '\0';
i = 0;
first = true;
while (out->supported_formats[i] != 0) {
for (j = 0; j < ARRAY_SIZE(out_formats_name_to_enum_table); j++) {
if (out_formats_name_to_enum_table[j].value == out->supported_formats[i]) {
if (!first) {
strlcat(value, "|", sizeof(value));
}
strlcat(value, out_formats_name_to_enum_table[j].name, sizeof(value));
first = false;
break;
}
}
i++;
}
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value);
if (str)
free(str);
str = str_parms_to_str(reply);
}
ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES, value, sizeof(value));
if (ret >= 0) {
value[0] = '\0';
i = 0;
first = true;
while (out->supported_sample_rates[i] != 0) {
for (j = 0; j < ARRAY_SIZE(out_hdmi_sample_rates_name_to_enum_table); j++) {
if (out_hdmi_sample_rates_name_to_enum_table[j].value == out->supported_sample_rates[i]) {
if (!first) {
strlcat(value, "|", sizeof(value));
}
strlcat(value, out_hdmi_sample_rates_name_to_enum_table[j].name, sizeof(value));
first = false;
break;
}
}
i++;
}
str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES, value);
if (str)
free(str);
str = str_parms_to_str(reply);
}
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 period_ms;
struct stream_out *out = (struct stream_out *)stream;
uint32_t latency = 0;
if (is_offload_usecase(out->usecase)) {
lock_output_stream(out);
latency = audio_extn_utils_compress_get_dsp_latency(out);
pthread_mutex_unlock(&out->lock);
} else if (out->realtime) {
// since the buffer won't be filled up faster than realtime,
// return a smaller number
if (out->config.rate)
period_ms = (out->af_period_multiplier * out->config.period_size *
1000) / (out->config.rate);
else
period_ms = 0;
latency = period_ms + platform_render_latency(out->usecase)/1000;
} else {
latency = (out->config.period_count * out->config.period_size * 1000) /
(out->config.rate);
}
if ((AUDIO_DEVICE_OUT_BLUETOOTH_A2DP == out->devices) &&
!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))
latency += audio_extn_a2dp_get_encoder_latency();
ALOGV("%s: Latency %d", __func__, latency);
return latency;
}
static float AmpToDb(float amplification)
{
float db = DSD_VOLUME_MIN_DB;
if (amplification > 0) {
db = 20 * log10(amplification);
if(db < DSD_VOLUME_MIN_DB)
return DSD_VOLUME_MIN_DB;
}
return db;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
struct stream_out *out = (struct stream_out *)stream;
int volume[2];
if (out->usecase == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
/* only take left channel into account: the API is for stereo anyway */
out->muted = (left == 0.0f);
return 0;
} else if (is_offload_usecase(out->usecase)) {
if (audio_extn_passthru_is_passthrough_stream(out)) {
/*
* Set mute or umute on HDMI passthrough stream.
* Only take left channel into account.
* Mute is 0 and unmute 1
*/
audio_extn_passthru_set_volume(out, (left == 0.0f));
} else if (out->format == AUDIO_FORMAT_DSD){
char mixer_ctl_name[128] = "DSD Volume";
struct audio_device *adev = out->dev;
struct mixer_ctl *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;
}
volume[0] = (int)(AmpToDb(left));
volume[1] = (int)(AmpToDb(right));
mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
return 0;
} 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),
"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;
}
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]));
return 0;
}
}
return -ENOSYS;
}
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;
int snd_scard_state = get_snd_card_state(adev);
ssize_t ret = 0;
lock_output_stream(out);
if (SND_CARD_STATE_OFFLINE == snd_scard_state) {
if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
/*during SSR for compress usecase we should return error to flinger*/
ALOGD(" copl %s: sound card is not active/SSR state", __func__);
pthread_mutex_unlock(&out->lock);
return -ENETRESET;
} else {
/* increase written size during SSR to avoid mismatch
* with the written frames count in AF
*/
// bytes per frame
size_t bpf = audio_bytes_per_sample(out->format) *
audio_channel_count_from_out_mask(out->channel_mask);
if (bpf != 0)
out->written += bytes / bpf;
ALOGD(" %s: sound card is not active/SSR state", __func__);
ret= -EIO;
goto exit;
}
}
if (audio_extn_passthru_should_drop_data(out)) {
ALOGV(" %s : Drop data as compress passthrough session is going on", __func__);
if (audio_bytes_per_sample(out->format) != 0)
out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format));
ret = -EIO;
goto exit;
}
if (out->standby) {
out->standby = false;
pthread_mutex_lock(&adev->lock);
if (out->usecase == USECASE_COMPRESS_VOIP_CALL)
ret = voice_extn_compress_voip_start_output_stream(out);
else
ret = start_output_stream(out);
pthread_mutex_unlock(&adev->lock);
/* ToDo: If use case is compress offload should return 0 */
if (ret != 0) {
out->standby = true;
goto exit;
}
if (last_known_cal_step != -1) {
ALOGD("%s: retry previous failed cal level set", __func__);
audio_hw_send_gain_dep_calibration(last_known_cal_step);
}
}
if (adev->is_channel_status_set == false && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)){
audio_utils_set_hdmi_channel_status(out, (void *)buffer, bytes);
adev->is_channel_status_set = true;
}
if (is_offload_usecase(out->usecase)) {
ALOGVV("copl(%p): writing buffer (%zu bytes) to compress device", out, bytes);
if (out->send_new_metadata) {
ALOGD("copl(%p):send new gapless metadata", out);
compress_set_gapless_metadata(out->compr, &out->gapless_mdata);
out->send_new_metadata = 0;
if (out->send_next_track_params && out->is_compr_metadata_avail) {
ALOGD("copl(%p):send next track params in gapless", out);
compress_set_next_track_param(out->compr, &(out->compr_config.codec->options));
out->send_next_track_params = false;
out->is_compr_metadata_avail = false;
}
}
if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
(out->convert_buffer) != NULL) {
if ((bytes > out->hal_fragment_size)) {
ALOGW("Error written bytes %zu > %d (fragment_size)",
bytes, out->hal_fragment_size);
pthread_mutex_unlock(&out->lock);
return -EINVAL;
} else {
audio_format_t dst_format = out->hal_op_format;
audio_format_t src_format = out->hal_ip_format;
uint32_t frames = bytes / format_to_bitwidth_table[src_format];
uint32_t bytes_to_write = frames * format_to_bitwidth_table[dst_format];
memcpy_by_audio_format(out->convert_buffer,
dst_format,
buffer,
src_format,
frames);
ret = compress_write(out->compr, out->convert_buffer,
bytes_to_write);
/*Convert written bytes in audio flinger format*/
if (ret > 0)
ret = ((ret * format_to_bitwidth_table[out->format]) /
format_to_bitwidth_table[dst_format]);
}
} else
ret = compress_write(out->compr, buffer, bytes);
if (ret < 0)
ret = -errno;
ALOGVV("%s: writing buffer (%zu bytes) to compress device returned %zd", __func__, bytes, ret);
/*msg to cb thread only if non blocking write is enabled*/
if (ret >= 0 && ret < (ssize_t)bytes && out->non_blocking) {
ALOGD("No space available in compress driver, post msg to cb thread");
send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
} else if (-ENETRESET == ret) {
ALOGE("copl %s: received sound card offline state on compress write", __func__);
set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
pthread_mutex_unlock(&out->lock);
out_standby(&out->stream.common);
return ret;
}
if ( ret == (ssize_t)bytes && !out->non_blocking)
out->written += bytes;
/* Call compr start only when non-zero bytes of data is there to be rendered */
if (!out->playback_started && ret > 0) {
int status = compress_start(out->compr);
if (status < 0) {
ret = status;
ALOGE("%s: compr start failed with err %d", __func__, errno);
goto exit;
}
audio_extn_dts_eagle_fade(adev, true, out);
out->playback_started = 1;
out->offload_state = OFFLOAD_STATE_PLAYING;
audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
popcount(out->channel_mask),
out->playback_started);
}
pthread_mutex_unlock(&out->lock);
return ret;
} else {
if (out->pcm) {
if (out->muted)
memset((void *)buffer, 0, bytes);
ALOGVV("%s: writing buffer (%zu bytes) to pcm device", __func__, bytes);
long ns = 0;
if (out->config.rate)
ns = pcm_bytes_to_frames(out->pcm, bytes)*1000000000LL/
out->config.rate;
bool use_mmap = is_mmap_usecase(out->usecase) || out->realtime;
request_out_focus(out, ns);
if (use_mmap)
ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
else if (out->hal_op_format != out->hal_ip_format &&
out->convert_buffer != NULL) {
memcpy_by_audio_format(out->convert_buffer,
out->hal_op_format,
buffer,
out->hal_ip_format,
out->config.period_size * out->config.channels);
ret = pcm_write(out->pcm, out->convert_buffer,
(out->config.period_size *
out->config.channels *
format_to_bitwidth_table[out->hal_op_format]));
} else {
ret = pcm_write(out->pcm, (void *)buffer, bytes);
}
release_out_focus(out);
if (ret < 0)
ret = -errno;
else if (ret == 0 && (audio_bytes_per_sample(out->format) != 0))
out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format));
else
ret = -EINVAL;
}
}
exit:
/* ToDo: There may be a corner case when SSR happens back to back during
start/stop. Need to post different error to handle that. */
if (-ENETRESET == ret) {
set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
}
pthread_mutex_unlock(&out->lock);
if (ret != 0) {
if (out->pcm)
ALOGE("%s: error %d, %s", __func__, (int)ret, pcm_get_error(out->pcm));
if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
pthread_mutex_lock(&adev->lock);
voice_extn_compress_voip_close_output_stream(&out->stream.common);
pthread_mutex_unlock(&adev->lock);
out->standby = true;
}
out_standby(&out->stream.common);
if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD))
usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
out_get_sample_rate(&out->stream.common));
}
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;
struct audio_device *adev = out->dev;
if (dsp_frames == NULL)
return -EINVAL;
*dsp_frames = 0;
if (is_offload_usecase(out->usecase)) {
ssize_t ret = 0;
/* Below piece of code is not guarded against any lock beacuse audioFliner serializes
* this operation and adev_close_output_stream(where out gets reset).
*/
if (!out->non_blocking && !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
*dsp_frames = get_actual_pcm_frames_rendered(out);
ALOGVV("dsp_frames %d sampleRate %d",(int)*dsp_frames,out->sample_rate);
return 0;
}
lock_output_stream(out);
if (out->compr != NULL && out->non_blocking) {
ret = compress_get_tstamp(out->compr, (unsigned long *)dsp_frames,
&out->sample_rate);
if (ret < 0)
ret = -errno;
ALOGVV("%s rendered frames %d sample_rate %d",
__func__, *dsp_frames, out->sample_rate);
}
pthread_mutex_unlock(&out->lock);
if (-ENETRESET == ret) {
ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
return -EINVAL;
} else if(ret < 0) {
ALOGE(" ERROR: Unable to get time stamp from compress driver");
return -EINVAL;
} else if (get_snd_card_state(adev) == SND_CARD_STATE_OFFLINE){
/*
* Handle corner case where compress session is closed during SSR
* and timestamp is queried
*/
ALOGE(" ERROR: sound card not active, return error");
return -EINVAL;
} else {
return 0;
}
} else if (audio_is_linear_pcm(out->format)) {
*dsp_frames = out->written;
return 0;
} else
return -EINVAL;
}
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 -EINVAL;
}
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 = -1;
unsigned long dsp_frames;
/* below piece of code is not guarded against any lock because audioFliner serializes
* this operation and adev_close_output_stream( where out gets reset).
*/
if (is_offload_usecase(out->usecase) && !out->non_blocking &&
!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
*frames = get_actual_pcm_frames_rendered(out);
/* this is the best we can do */
clock_gettime(CLOCK_MONOTONIC, timestamp);
ALOGVV("frames %lld playedat %lld",(long long int)*frames,
timestamp->tv_sec * 1000000LL + timestamp->tv_nsec / 1000);
return 0;
}
lock_output_stream(out);
if (is_offload_usecase(out->usecase) && out->compr != NULL && out->non_blocking) {
ret = compress_get_tstamp(out->compr, &dsp_frames,
&out->sample_rate);
ALOGVV("%s rendered frames %ld sample_rate %d",
__func__, dsp_frames, out->sample_rate);
*frames = dsp_frames;
if (ret < 0)
ret = -errno;
if (-ENETRESET == ret) {
ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
ret = -EINVAL;
} else
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) {
size_t kernel_buffer_size = out->config.period_size * out->config.period_count;
int64_t signed_frames = out->written - kernel_buffer_size + avail;
// 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->usecase) * out->sample_rate / 1000000LL);
// It would be unusual for this value to be negative, but check just in case ...
if (signed_frames >= 0) {
*frames = signed_frames;
ret = 0;
}
}
} else if (adev->snd_card_status.state == SND_CARD_STATE_OFFLINE) {
*frames = out->written;
clock_gettime(CLOCK_MONOTONIC, timestamp);
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;
int ret;
ALOGV("%s", __func__);
lock_output_stream(out);
out->client_callback = callback;
out->client_cookie = cookie;
if (out->adsp_hdlr_stream_handle) {
ret = audio_extn_adsp_hdlr_stream_set_callback(
out->adsp_hdlr_stream_handle,
callback,
cookie);
if (ret)
ALOGW("%s:adsp hdlr callback registration failed %d",
__func__, ret);
}
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 (is_offload_usecase(out->usecase)) {
ALOGD("copl(%p):pause compress driver", out);
lock_output_stream(out);
if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) {
struct audio_device *adev = out->dev;
int snd_scard_state = get_snd_card_state(adev);
if (SND_CARD_STATE_ONLINE == snd_scard_state)
status = compress_pause(out->compr);
out->offload_state = OFFLOAD_STATE_PAUSED;
if (audio_extn_passthru_is_active()) {
ALOGV("offload use case, pause passthru");
audio_extn_passthru_on_pause(out);
}
audio_extn_dts_eagle_fade(adev, false, out);
audio_extn_dts_notify_playback_state(out->usecase, 0,
out->sample_rate, popcount(out->channel_mask),
0);
}
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 (is_offload_usecase(out->usecase)) {
ALOGD("copl(%p):resume compress driver", out);
status = 0;
lock_output_stream(out);
if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) {
struct audio_device *adev = out->dev;
int snd_scard_state = get_snd_card_state(adev);
if (SND_CARD_STATE_ONLINE == snd_scard_state) {
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
pthread_mutex_lock(&out->dev->lock);
ALOGV("offload resume, check and set hdmi backend again");
pthread_mutex_unlock(&out->dev->lock);
}
status = compress_resume(out->compr);
}
if (!status) {
out->offload_state = OFFLOAD_STATE_PLAYING;
}
audio_extn_dts_eagle_fade(adev, true, out);
audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
popcount(out->channel_mask), 1);
}
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 (is_offload_usecase(out->usecase)) {
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 (is_offload_usecase(out->usecase)) {
ALOGD("copl(%p):calling compress flush", out);
lock_output_stream(out);
if (out->offload_state == OFFLOAD_STATE_PAUSED) {
stop_compressed_output_l(out);
out->written = 0;
} else {
ALOGW("%s called in invalid state %d", __func__, out->offload_state);
}
pthread_mutex_unlock(&out->lock);
ALOGD("copl(%p):out of compress flush", out);
return 0;
}
return -ENOSYS;
}
/** 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;
if(in->usecase == USECASE_COMPRESS_VOIP_CALL)
return voice_extn_compress_voip_in_get_buffer_size(in);
else if(audio_extn_compr_cap_usecase_supported(in->usecase))
return audio_extn_compr_cap_get_buffer_size(in->config.format);
else if(audio_extn_cin_attached_usecase(in->usecase))
return audio_extn_cin_get_buffer_size(in);
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;
ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
stream, in->usecase, use_case_table[in->usecase]);
lock_input_stream(in);
if (!in->standby && in->is_st_session) {
ALOGD("%s: sound trigger pcm stop lab", __func__);
audio_extn_sound_trigger_stop_lab(in);
in->standby = 1;
}
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_COMPRESS_VOIP_CALL) {
voice_extn_compress_voip_close_input_stream(stream);
ALOGD("VOIP input entered standby");
} else {
if (audio_extn_cin_attached_usecase(in->usecase))
audio_extn_cin_stop_input_stream(in);
if (in->pcm) {
pcm_close(in->pcm);
in->pcm = NULL;
}
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 __unused,
int fd __unused)
{
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 value[32];
int ret = 0, val = 0, err;
ALOGD("%s: enter: kvpairs=%s", __func__, kvpairs);
parms = str_parms_create_str(kvpairs);
if (!parms)
goto error;
lock_input_stream(in);
pthread_mutex_lock(&adev->lock);
err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value));
if (err >= 0) {
val = atoi(value);
/* no audio source uses val == 0 */
if ((in->source != val) && (val != 0)) {
in->source = val;
if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
(in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
(voice_extn_compress_voip_is_format_supported(in->format)) &&
(in->config.rate == 8000 || in->config.rate == 16000 ||
in->config.rate == 32000 || in->config.rate == 48000 ) &&
(audio_channel_count_from_in_mask(in->channel_mask) == 1)) {
err = voice_extn_compress_voip_open_input_stream(in);
if (err != 0) {
ALOGE("%s: Compress voip input cannot be opened, error:%d",
__func__, err);
}
}
}
}
err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
if (err >= 0) {
val = atoi(value);
if (((int)in->device != val) && (val != 0)) {
in->device = val;
/* If recording is in progress, change the tx device to new device */
if (!in->standby && !in->is_st_session) {
ALOGV("update input routing change");
if (adev->adm_on_routing_change)
adev->adm_on_routing_change(adev->adm_data,
in->capture_handle);
ret = select_devices(adev, in->usecase);
}
}
}
err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_PROFILE, value, sizeof(value));
if (err >= 0) {
strlcpy(in->profile, value, sizeof(in->profile));
ALOGV("updating stream profile with value '%s'", in->profile);
audio_extn_utils_update_stream_input_app_type_cfg(adev->platform,
&adev->streams_input_cfg_list,
in->device, in->flags, in->format,
in->sample_rate, in->bit_width,
in->profile, &in->app_type_cfg);
}
pthread_mutex_unlock(&adev->lock);
pthread_mutex_unlock(&in->lock);
str_parms_destroy(parms);
error:
ALOGV("%s: exit: status(%d)", __func__, ret);
return ret;
}
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();
if (!query || !reply) {
if (reply) {
str_parms_destroy(reply);
}
if (query) {
str_parms_destroy(query);
}
ALOGE("in_get_parameters: failed to create query or reply");
return NULL;
}
ALOGV("%s: enter: keys - %s", __func__, keys);
voice_extn_in_get_parameters(in, query, reply);
str = str_parms_to_str(reply);
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 __unused,
float gain __unused)
{
return 0;
}
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 ret = -1;
int snd_scard_state = get_snd_card_state(adev);
size_t bytes_read = 0;
lock_input_stream(in);
if (in->is_st_session) {
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 (SND_CARD_STATE_OFFLINE == snd_scard_state) {
ALOGD(" %s: sound card is not active/SSR state", __func__);
ret= -EIO;;
goto exit;
}
if (in->standby) {
pthread_mutex_lock(&adev->lock);
if (in->usecase == USECASE_COMPRESS_VOIP_CALL)
ret = voice_extn_compress_voip_start_input_stream(in);
else
ret = start_input_stream(in);
pthread_mutex_unlock(&adev->lock);
if (ret != 0) {
goto exit;
}
in->standby = 0;
}
// what's the duration requested by the client?
long ns = 0;
if (in->pcm && in->config.rate)
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 (audio_extn_cin_attached_usecase(in->usecase)) {
ret = audio_extn_cin_read(in, buffer, bytes, &bytes_read);
} else if (in->pcm) {
if (audio_extn_ssr_get_stream() == in) {
ret = audio_extn_ssr_read(stream, buffer, bytes);
} else if (audio_extn_compr_cap_usecase_supported(in->usecase)) {
ret = audio_extn_compr_cap_read(in, buffer, bytes);
} else if (use_mmap) {
ret = pcm_mmap_read(in->pcm, buffer, bytes);
} else {
ret = pcm_read(in->pcm, buffer, bytes);
/* data from DSP comes in 24_8 format, convert it to 8_24 */
if (!ret && bytes > 0 && (in->format == AUDIO_FORMAT_PCM_8_24_BIT)) {
if (audio_extn_utils_convert_format_24_8_to_8_24(buffer, bytes)
!= bytes) {
ret = -EINVAL;
goto exit;
}
} else if (ret < 0) {
ret = -errno;
}
}
/* bytes read is always set to bytes for non compress usecases */
bytes_read = bytes;
}
release_in_focus(in);
/*
* Instead of writing zeroes here, we could trust the hardware
* to always provide zeroes when muted.
*/
if (ret == 0 && voice_get_mic_mute(adev) && !voice_is_in_call_rec_stream(in) &&
in->usecase != USECASE_AUDIO_RECORD_AFE_PROXY)
memset(buffer, 0, bytes);
exit:
/* ToDo: There may be a corner case when SSR happens back to back during
start/stop. Need to post different error to handle that. */
if (-ENETRESET == ret)
set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
pthread_mutex_unlock(&in->lock);
if (ret != 0) {
if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
pthread_mutex_lock(&adev->lock);
voice_extn_compress_voip_close_input_stream(&in->stream.common);
pthread_mutex_unlock(&adev->lock);
in->standby = true;
}
if (!audio_extn_cin_attached_usecase(in->usecase)) {
bytes_read = bytes;
memset(buffer, 0, bytes);
}
in_standby(&in->stream.common);
ALOGV("%s: read failed status %d- sleeping for buffer duration", __func__, ret);
usleep((uint64_t)bytes * 1000000 / audio_stream_in_frame_size(stream) /
in_get_sample_rate(&in->stream.common));
}
return bytes_read;
}
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream __unused)
{
return 0;
}
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;
int status = 0;
effect_descriptor_t desc;
status = (*effect)->get_descriptor(effect, &desc);
if (status != 0)
return status;
lock_input_stream(in);
pthread_mutex_lock(&in->dev->lock);
if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
in->enable_aec != enable &&
(memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0)) {
in->enable_aec = enable;
if (!in->standby)
select_devices(in->dev, in->usecase);
}
if (in->enable_ns != enable &&
(memcmp(&desc.type, FX_IID_NS, sizeof(effect_uuid_t)) == 0)) {
in->enable_ns = enable;
if (!in->standby)
select_devices(in->dev, in->usecase);
}
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);
}
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 ret = 0;
audio_format_t format;
struct adsp_hdlr_stream_cfg hdlr_stream_cfg;
*stream_out = NULL;
if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
(SND_CARD_STATE_OFFLINE == get_snd_card_state(adev))) {
ALOGE("sound card is not active rejecting compress output open request");
return -EINVAL;
}
out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
ALOGD("%s: enter: format(%#x) sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)\
stream_handle(%p)", __func__, config->format, config->sample_rate, config->channel_mask,
devices, flags, &out->stream);
if (!out) {
return -ENOMEM;
}
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);
if (devices == AUDIO_DEVICE_NONE)
devices = AUDIO_DEVICE_OUT_SPEAKER;
out->flags = flags;
out->devices = devices;
out->dev = adev;
format = out->format = config->format;
out->sample_rate = config->sample_rate;
out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
out->handle = handle;
out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
out->non_blocking = 0;
out->convert_buffer = NULL;
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL &&
(flags & AUDIO_OUTPUT_FLAG_DIRECT)) {
pthread_mutex_lock(&adev->lock);
ALOGV("AUDIO_DEVICE_OUT_AUX_DIGITAL and DIRECT|OFFLOAD, check hdmi caps");
ret = read_hdmi_sink_caps(out);
pthread_mutex_unlock(&adev->lock);
if (ret != 0) {
if (ret == -ENOSYS) {
/* ignore and go with default */
ret = 0;
} else {
ALOGE("error reading hdmi sink caps");
goto error_open;
}
}
}
/* Init use case and pcm_config */
if ((out->dev->mode == AUDIO_MODE_IN_COMMUNICATION || voice_extn_compress_voip_is_active(out->dev)) &&
(out->flags == (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_VOIP_RX)) &&
(voice_extn_compress_voip_is_config_supported(config))) {
ret = voice_extn_compress_voip_open_output_stream(out);
if (ret != 0) {
ALOGE("%s: Compress voip output cannot be opened, error:%d",
__func__, ret);
goto error_open;
}
} else if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) ||
(out->flags == AUDIO_OUTPUT_FLAG_DIRECT)) {
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 (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
if(config->offload_info.format == 0)
config->offload_info.format = out->supported_formats[0];
if (config->offload_info.sample_rate == 0)
config->offload_info.sample_rate = out->supported_sample_rates[0];
}
if (!is_supported_format(config->offload_info.format) &&
!audio_extn_passthru_is_supported_format(config->offload_info.format)) {
ALOGE("%s: Unsupported audio format %x " , __func__, config->offload_info.format);
ret = -EINVAL;
goto error_open;
}
out->compr_config.codec = (struct snd_codec *)
calloc(1, sizeof(struct snd_codec));
if (!out->compr_config.codec) {
ret = -ENOMEM;
goto error_open;
}
out->stream.pause = out_pause;
out->stream.resume = out_resume;
out->stream.flush = out_flush;
if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
out->stream.set_callback = out_set_callback;
out->stream.drain = out_drain;
out->usecase = get_offload_usecase(adev, true /* is_compress */);
ALOGV("Compress Offload usecase .. usecase selected %d", out->usecase);
} else {
out->usecase = get_offload_usecase(adev, false /* is_compress */);
ALOGV("non-offload DIRECT_usecase ... usecase selected %d ", out->usecase);
}
if (out->usecase == USECASE_INVALID) {
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL &&
config->format == 0 && config->sample_rate == 0 &&
config->channel_mask == 0) {
ALOGI("%s dummy open to query sink capability",__func__);
out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD;
} else {
ALOGE("%s, Max allowed OFFLOAD usecase reached ... ", __func__);
ret = -EEXIST;
goto error_open;
}
}
if (config->offload_info.channel_mask)
out->channel_mask = config->offload_info.channel_mask;
else if (config->channel_mask) {
out->channel_mask = config->channel_mask;
config->offload_info.channel_mask = config->channel_mask;
} else {
ALOGE("out->channel_mask not set for OFFLOAD/DIRECT usecase");
ret = -EINVAL;
goto error_open;
}
format = out->format = config->offload_info.format;
out->sample_rate = config->offload_info.sample_rate;
out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
out->compr_config.codec->id = get_snd_codec_id(config->offload_info.format);
if (audio_extn_is_dolby_format(config->offload_info.format)) {
audio_extn_dolby_send_ddp_endp_params(adev);
audio_extn_dolby_set_dmid(adev);
}
out->compr_config.codec->sample_rate =
config->offload_info.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;
/* Update bit width only for non passthrough usecases.
* For passthrough usecases, the output will always be opened @16 bit
*/
if (!audio_extn_passthru_is_passthrough_stream(out))
out->bit_width = AUDIO_OUTPUT_BIT_WIDTH;
/*TODO: Do we need to change it for passthrough */
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC)
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
else if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS)
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
else if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_LATM)
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4LATM;
if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) ==
AUDIO_FORMAT_PCM) {
/*Based on platform support, configure appropriate alsa format for corresponding
*hal input format.
*/
out->compr_config.codec->format = hal_format_to_alsa(
config->offload_info.format);
out->hal_op_format = alsa_format_to_hal(
out->compr_config.codec->format);
out->hal_ip_format = out->format;
/*for direct non-compress playback populate bit_width based on selected alsa format as
*hal input format and alsa format might differ based on platform support.
*/
out->bit_width = audio_bytes_per_sample(
out->hal_op_format) << 3;
out->compr_config.fragments = DIRECT_PCM_NUM_FRAGMENTS;
/* Check if alsa session is configured with the same format as HAL input format,
* if not then derive correct fragment size needed to accomodate the
* conversion of HAL input format to alsa format.
*/
audio_extn_utils_update_direct_pcm_fragment_size(out);
/*if hal input and output fragment size is different this indicates HAL input format is
*not same as the alsa format
*/
if (out->hal_fragment_size != out->compr_config.fragment_size) {
/*Allocate a buffer to convert input data to the alsa configured format.
*size of convert buffer is equal to the size required to hold one fragment size
*worth of pcm data, this is because flinger does not write more than fragment_size
*/
out->convert_buffer = calloc(1,out->compr_config.fragment_size);
if (out->convert_buffer == NULL){
ALOGE("Allocation failed for convert buffer for size %d", out->compr_config.fragment_size);
ret = -ENOMEM;
goto error_open;
}
}
} else if (audio_extn_passthru_is_passthrough_stream(out)) {
out->compr_config.fragment_size =
audio_extn_passthru_get_buffer_size(&config->offload_info);
out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
} else {
out->compr_config.fragment_size =
platform_get_compress_offload_buffer_size(&config->offload_info);
out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
}
if (config->offload_info.format == AUDIO_FORMAT_FLAC)
out->compr_config.codec->options.flac_dec.sample_size = AUDIO_OUTPUT_BIT_WIDTH;
if (config->offload_info.format == AUDIO_FORMAT_APTX) {
audio_extn_send_aptx_dec_bt_addr_to_dsp(out);
}
if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
out->non_blocking = 1;
if ((flags & AUDIO_OUTPUT_FLAG_TIMESTAMP) &&
(flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC)) {
out->render_mode = RENDER_MODE_AUDIO_STC_MASTER;
} else if(flags & AUDIO_OUTPUT_FLAG_TIMESTAMP) {
out->render_mode = RENDER_MODE_AUDIO_MASTER;
} else {
out->render_mode = RENDER_MODE_AUDIO_NO_TIMESTAMP;
}
memset(&out->render_window, 0,
sizeof(struct audio_out_render_window_param));
out->send_new_metadata = 1;
out->send_next_track_params = false;
out->is_compr_metadata_avail = false;
out->offload_state = OFFLOAD_STATE_IDLE;
out->playback_started = 0;
audio_extn_dts_create_state_notifier_node(out->usecase);
ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
__func__, config->offload_info.version,
config->offload_info.bit_rate);
/* Check if DSD audio format is supported in codec
* and there is no active native DSD use case
*/
if ((config->format == AUDIO_FORMAT_DSD) &&
(!platform_check_codec_dsd_support(adev->platform) ||
audio_is_dsd_native_stream_active(adev))) {
ret = -EINVAL;
goto error_open;
}
/* Disable gapless if any of the following is true
* passthrough playback
* AV playback
* non compressed Direct playback
*/
if (audio_extn_passthru_is_passthrough_stream(out) ||
(config->format == AUDIO_FORMAT_DSD) ||
config->offload_info.has_video ||
!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) {
check_and_set_gapless_mode(adev, false);
} else
check_and_set_gapless_mode(adev, true);
if (audio_extn_passthru_is_passthrough_stream(out)) {
out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
}
if (config->format == AUDIO_FORMAT_DSD) {
out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH;
out->compr_config.codec->compr_passthr = PASSTHROUGH_DSD;
}
create_offload_callback_thread(out);
} else 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_open;
}
} else if (out->devices == AUDIO_DEVICE_OUT_TELEPHONY_TX) {
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 error_open;
}
out->sample_rate = config->sample_rate;
out->config.rate = config->sample_rate;
if (config->format == AUDIO_FORMAT_DEFAULT)
config->format = AUDIO_FORMAT_PCM_16_BIT;
if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
config->format = AUDIO_FORMAT_PCM_16_BIT;
ret = -EINVAL;
goto error_open;
}
out->format = config->format;
out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY;
out->config = pcm_config_afe_proxy_playback;
adev->voice_tx_output = out;
} else {
unsigned int channels = 0;
/*Update config params to default if not set by the caller*/
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_STEREO;
if (config->format == AUDIO_FORMAT_DEFAULT)
config->format = AUDIO_FORMAT_PCM_16_BIT;
channels = audio_channel_count_from_out_mask(out->channel_mask);
if (out->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 (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
out->config = pcm_config_low_latency;
} else if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
out->config = pcm_config_deep_buffer;
out->config.period_size = get_output_period_size(config->sample_rate, out->format,
channels, DEEP_BUFFER_OUTPUT_PERIOD_DURATION);
if (out->config.period_size <= 0) {
ALOGE("Invalid configuration period size is not valid");
ret = -EINVAL;
goto error_open;
}
} else {
/* primary path is the default path selected if no other outputs are available/suitable */
out->usecase = USECASE_AUDIO_PLAYBACK_PRIMARY;
out->config = PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY;
}
out->hal_ip_format = format = out->format;
out->config.format = hal_format_to_pcm(out->hal_ip_format);
out->hal_op_format = pcm_format_to_hal(out->config.format);
out->bit_width = format_to_bitwidth_table[out->hal_op_format] << 3;
out->config.rate = config->sample_rate;
out->sample_rate = out->config.rate;
out->config.channels = channels;
if (out->hal_ip_format != out->hal_op_format) {
uint32_t buffer_size = out->config.period_size *
format_to_bitwidth_table[out->hal_op_format] *
out->config.channels;
out->convert_buffer = calloc(1, buffer_size);
if (out->convert_buffer == NULL){
ALOGE("Allocation failed for convert buffer for size %d",
out->compr_config.fragment_size);
ret = -ENOMEM;
goto error_open;
}
ALOGD("Convert buffer allocated of size %d", buffer_size);
}
}
ALOGV("%s devices:%d, format:%x, out->sample_rate:%d,out->bit_width:%d out->format:%d out->flags:%x, flags: %x usecase %d",
__func__, devices, format, out->sample_rate, out->bit_width, out->format, out->flags, flags, out->usecase);
/* TODO remove this hardcoding and check why width is zero*/
if (out->bit_width == 0)
out->bit_width = 16;
audio_extn_utils_update_stream_output_app_type_cfg(adev->platform,
&adev->streams_output_cfg_list,
devices, out->flags, format, out->sample_rate,
out->bit_width, out->channel_mask, out->profile,
&out->app_type_cfg);
if ((out->usecase == USECASE_AUDIO_PLAYBACK_PRIMARY) ||
(flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
/* Ensure the default output is not selected twice */
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) &&
(out->usecase != USECASE_COMPRESS_VOIP_CALL)) {
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->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;
out->stream.write = out_write;
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;
out->af_period_multiplier = out->realtime ? af_period_multiplier : 1;
out->standby = 1;
/* out->muted = false; by calloc() */
/* out->written = 0; by calloc() */
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);
*stream_out = &out->stream;
ALOGD("%s: Stream (%p) picks up usecase (%s)", __func__, &out->stream,
use_case_table[out->usecase]);
if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)
audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate,
popcount(out->channel_mask), out->playback_started);
/* setup a channel for client <--> adsp communication for stream events */
if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) ||
(out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) {
hdlr_stream_cfg.pcm_device_id = platform_get_pcm_device_id(
out->usecase, PCM_PLAYBACK);
hdlr_stream_cfg.flags = out->flags;
hdlr_stream_cfg.type = PCM_PLAYBACK;
ret = audio_extn_adsp_hdlr_stream_open(&out->adsp_hdlr_stream_handle,
&hdlr_stream_cfg);
if (ret) {
ALOGE("%s: adsp_hdlr_stream_open failed %d",__func__, ret);
out->adsp_hdlr_stream_handle = NULL;
}
}
ALOGV("%s: exit", __func__);
return 0;
error_open:
if (out->convert_buffer)
free(out->convert_buffer);
free(out);
*stream_out = NULL;
ALOGD("%s: exit: ret %d", __func__, ret);
return ret;
}
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;
int ret = 0;
ALOGD("%s: enter:stream_handle(%p)",__func__, out);
/* close adsp hdrl session before standby */
if (out->adsp_hdlr_stream_handle) {
ret = audio_extn_adsp_hdlr_stream_close(out->adsp_hdlr_stream_handle);
if (ret)
ALOGE("%s: adsp_hdlr_stream_close failed %d",__func__, ret);
out->adsp_hdlr_stream_handle = NULL;
}
if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
pthread_mutex_lock(&adev->lock);
ret = voice_extn_compress_voip_close_output_stream(&stream->common);
pthread_mutex_unlock(&adev->lock);
if(ret != 0)
ALOGE("%s: Compress voip output cannot be closed, error:%d",
__func__, ret);
} else
out_standby(&stream->common);
if (is_offload_usecase(out->usecase)) {
audio_extn_dts_remove_state_notifier_node(out->usecase);
destroy_offload_callback_thread(out);
free_offload_usecase(adev, out->usecase);
if (out->compr_config.codec != NULL)
free(out->compr_config.codec);
}
if (out->convert_buffer != NULL) {
free(out->convert_buffer);
out->convert_buffer = NULL;
}
if (adev->voice_tx_output == out)
adev->voice_tx_output = NULL;
if (adev->primary_output == out)
adev->primary_output = NULL;
pthread_cond_destroy(&out->cond);
pthread_mutex_destroy(&out->lock);
free(stream);
ALOGV("%s: exit", __func__);
}
static void close_compress_sessions(struct audio_device *adev)
{
struct stream_out *out;
struct listnode *node, *tempnode;
struct audio_usecase *usecase;
pthread_mutex_lock(&adev->lock);
list_for_each_safe(node, tempnode, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (is_offload_usecase(usecase->id)) {
if (usecase->stream.out) {
ALOGI(" %s closing compress session %d on OFFLINE state", __func__, usecase->id);
out = usecase->stream.out;
pthread_mutex_unlock(&adev->lock);
out_standby(&out->stream.common);
pthread_mutex_lock(&adev->lock);
tempnode = list_head(&adev->usecase_list);
}
}
}
pthread_mutex_unlock(&adev->lock);
}
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 value[32];
int val;
int ret;
int status = 0;
ALOGD("%s: enter: %s", __func__, kvpairs);
parms = str_parms_create_str(kvpairs);
if (!parms)
goto error;
ret = str_parms_get_str(parms, "SND_CARD_STATUS", value, sizeof(value));
if (ret >= 0) {
char *snd_card_status = value+2;
if (strstr(snd_card_status, "OFFLINE")) {
ALOGD("Received sound card OFFLINE status");
set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
//close compress sessions on OFFLINE status
close_compress_sessions(adev);
} else if (strstr(snd_card_status, "ONLINE")) {
ALOGD("Received sound card ONLINE status");
set_snd_card_state(adev,SND_CARD_STATE_ONLINE);
//send dts hpx license if enabled
audio_extn_dts_eagle_send_lic();
}
}
pthread_mutex_lock(&adev->lock);
status = voice_set_parameters(adev, parms);
if (status != 0)
goto done;
status = platform_set_parameters(adev->platform, 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;
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;
break;
case 0:
case 90:
case 180:
break;
default:
ALOGE("%s: unexpected rotation of %d", __func__, val);
status = -EINVAL;
}
if (status == 0) {
if (adev->speaker_lr_swap != reverse_speakers) {
adev->speaker_lr_swap = reverse_speakers;
// only update the selected device if there is active pcm playback
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);
break;
}
}
}
}
}
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_SCO_WB, value, sizeof(value));
if (ret >= 0) {
if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
adev->bt_wb_speech_enabled = true;
else
adev->bt_wb_speech_enabled = false;
}
ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
ALOGV("cache new ext disp type and edid");
ret = platform_get_ext_disp_type(adev->platform);
if (ret < 0) {
ALOGE("%s: Failed to query disp type, ret:%d", __func__, ret);
status = ret;
goto done;
}
platform_cache_edid(adev->platform);
} else if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) ||
!(val ^ AUDIO_DEVICE_IN_USB_DEVICE)) {
/*
* Do not allow AFE proxy port usage by WFD source when USB headset is connected.
* Per AudioPolicyManager, USB device is higher priority than WFD.
* For Voice call over USB headset, voice call audio is routed to AFE proxy ports.
* If WFD use case occupies AFE proxy, it may result unintended behavior while
* starting voice call on USB
*/
ret = str_parms_get_str(parms, "card", value, sizeof(value));
if (ret >= 0) {
audio_extn_usb_add_device(AUDIO_DEVICE_OUT_USB_DEVICE, atoi(value));
audio_extn_usb_add_device(AUDIO_DEVICE_IN_USB_DEVICE, atoi(value));
}
ALOGV("detected USB connect .. disable proxy");
adev->allow_afe_proxy_usage = false;
}
}
ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
/*
* The HDMI / Displayport disconnect handling has been moved to
* audio extension to ensure that its parameters are not
* invalidated prior to updating sysfs of the disconnect event
* Invalidate will be handled by audio_extn_ext_disp_set_parameters()
*/
if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) ||
!(val ^ AUDIO_DEVICE_IN_USB_DEVICE)) {
ret = str_parms_get_str(parms, "card", value, sizeof(value));
if (ret >= 0) {
audio_extn_usb_remove_device(AUDIO_DEVICE_OUT_USB_DEVICE, atoi(value));
audio_extn_usb_remove_device(AUDIO_DEVICE_IN_USB_DEVICE, atoi(value));
}
ALOGV("detected USB disconnect .. enable proxy");
adev->allow_afe_proxy_usage = true;
}
}
ret = str_parms_get_str(parms,"reconfigA2dp", value, sizeof(value));
if (ret >= 0) {
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("reconfigure a2dp... forcing device switch");
lock_output_stream(usecase->stream.out);
audio_extn_a2dp_set_handoff_mode(true);
//force device switch to re configure encoder
select_devices(adev, usecase->id);
audio_extn_a2dp_set_handoff_mode(false);
pthread_mutex_unlock(&usecase->stream.out->lock);
break;
}
}
}
//handle vr audio setparam
ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_VR_AUDIO_MODE,
value, sizeof(value));
if (ret >= 0) {
ALOGI("Setting vr mode to be %s", value);
if (!strncmp(value, "true", 4)) {
adev->vr_audio_mode_enabled = true;
ALOGI("Setting vr mode to true");
} else if (!strncmp(value, "false", 5)) {
adev->vr_audio_mode_enabled = false;
ALOGI("Setting vr mode to false");
} else {
ALOGI("wrong vr mode set");
}
}
audio_extn_set_parameters(adev, parms);
done:
str_parms_destroy(parms);
pthread_mutex_unlock(&adev->lock);
error:
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;
char value[256] = {0};
int ret = 0;
if (!query || !reply) {
if (reply) {
str_parms_destroy(reply);
}
if (query) {
str_parms_destroy(query);
}
ALOGE("adev_get_parameters: failed to create query or reply");
return NULL;
}
ret = str_parms_get_str(query, "SND_CARD_STATUS", value,
sizeof(value));
if (ret >=0) {
int val = 1;
pthread_mutex_lock(&adev->snd_card_status.lock);
if (SND_CARD_STATE_OFFLINE == adev->snd_card_status.state)
val = 0;
pthread_mutex_unlock(&adev->snd_card_status.lock);
str_parms_add_int(reply, "SND_CARD_STATUS", val);
goto exit;
}
//handle vr audio getparam
ret = str_parms_get_str(query,
AUDIO_PARAMETER_KEY_VR_AUDIO_MODE,
value, sizeof(value));
if (ret >= 0) {
bool vr_audio_enabled = false;
pthread_mutex_lock(&adev->lock);
vr_audio_enabled = adev->vr_audio_mode_enabled;
pthread_mutex_unlock(&adev->lock);
ALOGI("getting vr mode to %d", vr_audio_enabled);
if (vr_audio_enabled) {
str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VR_AUDIO_MODE,
"true");
goto exit;
} else {
str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VR_AUDIO_MODE,
"false");
goto exit;
}
}
pthread_mutex_lock(&adev->lock);
audio_extn_get_parameters(adev, query, reply);
voice_get_parameters(adev, query, reply);
platform_get_parameters(adev->platform, query, reply);
pthread_mutex_unlock(&adev->lock);
exit:
str = str_parms_to_str(reply);
str_parms_destroy(query);
str_parms_destroy(reply);
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;
pthread_mutex_lock(&adev->lock);
/* cache volume */
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\n", __func__, mode);
adev->mode = mode;
if ((mode == AUDIO_MODE_NORMAL) && voice_is_in_call(adev)) {
voice_stop_call(adev);
platform_set_gsm_mode(adev->platform, false);
adev->current_call_output = NULL;
}
}
pthread_mutex_unlock(&adev->lock);
return 0;
}
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
int ret;
pthread_mutex_lock(&adev->lock);
ALOGD("%s state %d\n", __func__, state);
ret = voice_set_mic_mute((struct audio_device *)dev, 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);
return get_input_buffer_size(config->sample_rate, config->format, channel_count,
false /* is_low_latency: since we don't know, be conservative */);
}
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 = audio_channel_count_from_in_mask(config->channel_mask);
bool is_low_latency = false;
bool channel_mask_updated = false;
*stream_in = NULL;
if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) {
ALOGE("%s: invalid input parameters", __func__);
return -EINVAL;
}
in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
if (!in) {
ALOGE("failed to allocate input stream");
return -ENOMEM;
}
ALOGD("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x)\
stream_handle(%p) io_handle(%d) source(%d) format %x",__func__, config->sample_rate,
config->channel_mask, devices, &in->stream, handle, source, config->format);
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->device = devices;
in->source = source;
in->dev = adev;
in->standby = 1;
in->channel_mask = config->channel_mask;
in->capture_handle = handle;
in->flags = flags;
in->usecase = USECASE_AUDIO_RECORD;
if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
(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);
}
in->format = config->format;
if (in->realtime) {
in->config = pcm_config_audio_capture_rt;
in->sample_rate = in->config.rate;
in->af_period_multiplier = af_period_multiplier;
} else {
in->config = pcm_config_audio_capture;
in->config.rate = config->sample_rate;
in->sample_rate = config->sample_rate;
in->af_period_multiplier = 1;
}
in->bit_width = 16;
/* restrict 24 bit capture for unprocessed source only
* for other sources if 24 bit requested reject 24 and set 16 bit capture only
*/
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_32_BIT) ||
(config->format == AUDIO_FORMAT_PCM_24_BIT_PACKED) ||
(config->format == AUDIO_FORMAT_PCM_8_24_BIT)) {
bool ret_error = false;
in->bit_width = 24;
/* 24 bit is restricted to UNPROCESSED source only,also format supported
from HAL is 24_packed and 8_24
*> In case of UNPROCESSED source, for 24 bit, if format requested is other than
24_packed return error indicating supported format is 24_packed
*> 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 ((source != AUDIO_SOURCE_UNPROCESSED) &&
(source != AUDIO_SOURCE_CAMCORDER)) {
config->format = AUDIO_FORMAT_PCM_16_BIT;
if (config->sample_rate > 48000)
config->sample_rate = 48000;
ret_error = true;
} else if (config->format == AUDIO_FORMAT_PCM_24_BIT_PACKED) {
in->config.format = PCM_FORMAT_S24_3LE;
} else if (config->format == AUDIO_FORMAT_PCM_8_24_BIT) {
in->config.format = PCM_FORMAT_S24_LE;
} else {
config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
ret_error = true;
}
if (ret_error) {
ret = -EINVAL;
goto err_open;
}
}
/* Update config params with the requested sample rate and channels */
if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) &&
(adev->mode != AUDIO_MODE_IN_CALL)) {
ret = -EINVAL;
goto err_open;
}
if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) ||
(in->device == AUDIO_DEVICE_IN_PROXY)) {
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_DEFAULT)
config->format = AUDIO_FORMAT_PCM_16_BIT;
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->config.channels = channel_count;
in->config.rate = config->sample_rate;
in->sample_rate = config->sample_rate;
} else if (!audio_extn_check_and_set_multichannel_usecase(adev,
in, config, &channel_mask_updated)) {
if (channel_mask_updated == true) {
ALOGD("%s: return error to retry with updated channel mask (%#x)",
__func__, config->channel_mask);
ret = -EINVAL;
goto err_open;
}
ALOGD("%s: created surround sound session succesfully",__func__);
} else if (audio_extn_compr_cap_enabled() &&
audio_extn_compr_cap_format_supported(config->format) &&
(in->dev->mode != AUDIO_MODE_IN_COMMUNICATION)) {
audio_extn_compr_cap_init(in);
} else if (audio_extn_cin_applicable_stream(in)) {
ret = audio_extn_cin_configure_input_stream(in);
if (ret)
goto err_open;
} else {
in->config.channels = channel_count;
if (!in->realtime) {
in->format = config->format;
frame_size = audio_stream_in_frame_size(&in->stream);
buffer_size = get_input_buffer_size(config->sample_rate,
config->format,
channel_count,
is_low_latency);
in->config.period_size = buffer_size / frame_size;
}
if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
(in->dev->mode == AUDIO_MODE_IN_COMMUNICATION || voice_extn_compress_voip_is_active(in->dev)) &&
(voice_extn_compress_voip_is_format_supported(in->format)) &&
(in->config.rate == 8000 || in->config.rate == 16000 ||
in->config.rate == 32000 || in->config.rate == 48000) &&
(audio_channel_count_from_in_mask(in->channel_mask) == 1)) {
voice_extn_compress_voip_open_input_stream(in);
}
}
audio_extn_utils_update_stream_input_app_type_cfg(adev->platform,
&adev->streams_input_cfg_list,
devices, flags, in->format, in->sample_rate,
in->bit_width, in->profile, &in->app_type_cfg);
/* This stream could be for sound trigger lab,
get sound trigger pcm if present */
audio_extn_sound_trigger_check_and_get_session(in);
*stream_in = &in->stream;
ALOGV("%s: exit", __func__);
return ret;
err_open:
free(in);
*stream_in = NULL;
return ret;
}
static void adev_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *stream)
{
int ret;
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = (struct audio_device *)dev;
ALOGD("%s: enter:stream_handle(%p)",__func__, in);
/* Disable echo reference while closing input stream */
platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE);
if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
pthread_mutex_lock(&adev->lock);
ret = voice_extn_compress_voip_close_input_stream(&stream->common);
pthread_mutex_unlock(&adev->lock);
if (ret != 0)
ALOGE("%s: Compress voip input cannot be closed, error:%d",
__func__, ret);
} else
in_standby(&stream->common);
if (audio_extn_ssr_get_stream() == in) {
audio_extn_ssr_deinit();
}
if (audio_extn_compr_cap_enabled() &&
audio_extn_compr_cap_format_supported(in->config.format))
audio_extn_compr_cap_deinit();
if (audio_extn_cin_attached_usecase(in->usecase))
audio_extn_cin_close_input_stream(in);
if (in->is_st_session) {
ALOGV("%s: sound trigger pcm stop lab", __func__);
audio_extn_sound_trigger_stop_lab(in);
}
free(stream);
return;
}
static int adev_dump(const audio_hw_device_t *device __unused,
int fd __unused)
{
return 0;
}
static int adev_close(hw_device_t *device)
{
struct audio_device *adev = (struct audio_device *)device;
if (!adev)
return 0;
pthread_mutex_lock(&adev_init_lock);
if ((--audio_device_ref_count) == 0) {
audio_extn_sound_trigger_deinit(adev);
audio_extn_listen_deinit(adev);
audio_extn_utils_release_streams_cfg_lists(
&adev->streams_output_cfg_list,
&adev->streams_input_cfg_list);
if (audio_extn_qaf_is_enabled())
audio_extn_qaf_deinit();
audio_route_free(adev->audio_route);
audio_extn_gef_deinit();
free(adev->snd_dev_ref_cnt);
platform_deinit(adev->platform);
if (adev->adm_deinit)
adev->adm_deinit(adev->adm_data);
qahwi_deinit(device);
audio_extn_adsp_hdlr_deinit();
free(device);
adev = NULL;
}
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 160:
case 192:
case 240:
case 320:
case 480:
return 1;
default:
return 0;
}
}
static int adev_open(const hw_module_t *module, const char *name,
hw_device_t **device)
{
int 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++;
ALOGD("%s: returning existing instance of adev", __func__);
ALOGD("%s: exit", __func__);
pthread_mutex_unlock(&adev_init_lock);
return 0;
}
adev = calloc(1, sizeof(struct audio_device));
if (!adev) {
pthread_mutex_unlock(&adev_init_lock);
return -ENOMEM;
}
pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL);
#ifdef DYNAMIC_LOG_ENABLED
register_for_dynamic_logging("hal");
#endif
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;
/* Set the default route before the PCM stream is opened */
adev->mode = AUDIO_MODE_NORMAL;
adev->active_input = NULL;
adev->primary_output = NULL;
adev->out_device = AUDIO_DEVICE_NONE;
adev->bluetooth_nrec = true;
adev->acdb_settings = TTY_MODE_OFF;
adev->allow_afe_proxy_usage = true;
/* 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);
adev->cur_wfd_channels = 2;
adev->offload_usecases_state = 0;
adev->is_channel_status_set = false;
adev->perf_lock_opts[0] = 0x101;
adev->perf_lock_opts[1] = 0x20E;
adev->perf_lock_opts_size = 2;
pthread_mutex_init(&adev->snd_card_status.lock, (const pthread_mutexattr_t *) NULL);
adev->snd_card_status.state = SND_CARD_STATE_OFFLINE;
/* 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);
pthread_mutex_destroy(&adev->lock);
pthread_mutex_destroy(&adev->snd_card_status.lock);
return -EINVAL;
}
if (audio_extn_qaf_is_enabled()) {
ret = audio_extn_qaf_init(adev);
if (ret < 0) {
free(adev);
ALOGE("%s: Failed to init platform data, aborting.", __func__);
*device = NULL;
pthread_mutex_unlock(&adev_init_lock);
pthread_mutex_destroy(&adev->lock);
return ret;
}
adev->device.open_output_stream = audio_extn_qaf_open_output_stream;
adev->device.close_output_stream = audio_extn_qaf_close_output_stream;
}
adev->snd_card_status.state = SND_CARD_STATE_ONLINE;
if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) {
adev->visualizer_lib = dlopen(VISUALIZER_LIBRARY_PATH, RTLD_NOW);
if (adev->visualizer_lib == NULL) {
ALOGE("%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))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");
}
}
audio_extn_init(adev);
audio_extn_listen_init(adev, adev->snd_card);
audio_extn_sound_trigger_init(adev);
audio_extn_gef_init(adev);
if (access(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, R_OK) == 0) {
adev->offload_effects_lib = dlopen(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, RTLD_NOW);
if (adev->offload_effects_lib == NULL) {
ALOGE("%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, struct mixer *))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->offload_effects_set_hpx_state =
(int (*)(bool))dlsym(adev->offload_effects_lib,
"offload_effects_bundle_set_hpx_state");
adev->offload_effects_get_parameters =
(void (*)(struct str_parms *, struct str_parms *))
dlsym(adev->offload_effects_lib,
"offload_effects_bundle_get_parameters");
adev->offload_effects_set_parameters =
(void (*)(struct str_parms *))dlsym(adev->offload_effects_lib,
"offload_effects_bundle_set_parameters");
}
}
if (access(ADM_LIBRARY_PATH, R_OK) == 0) {
adev->adm_lib = dlopen(ADM_LIBRARY_PATH, RTLD_NOW);
if (adev->adm_lib == NULL) {
ALOGE("%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;
//initialize this to false for now,
//this will be set to true through set param
adev->vr_audio_mode_enabled = false;
audio_extn_ds2_enable(adev);
*device = &adev->device.common;
audio_extn_utils_update_streams_cfg_lists(adev->platform, adev->mixer,
&adev->streams_output_cfg_list,
&adev->streams_input_cfg_list);
audio_device_ref_count++;
char value[PROPERTY_VALUE_MAX];
int trial;
if (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("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;
}
}
if (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);
}
adev->multi_offload_enable = property_get_bool("audio.offload.multiple.enabled", false);
pthread_mutex_unlock(&adev_init_lock);
if (adev->adm_init)
adev->adm_data = adev->adm_init();
qahwi_init(*device);
audio_extn_perf_lock_init();
audio_extn_adsp_hdlr_init(adev->mixer);
ALOGV("%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 = "The Linux Foundation",
.methods = &hal_module_methods,
},
};