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gerrit-public.fairphone.software / platform / hardware / qcom / audio / f59bb22e6d83b9ff56e88e35e6fd7329676ae2fb / . / hal / audio_hw.c
blob: 266dbd9e230c728a416d662eba9fb68b4639d875 [file] [log] [blame]
/*
* Copyright (c) 2013-2015, 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 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/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_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"
#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
/* ToDo: Check and update a proper value in msec */
#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 50
#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
#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
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,
};
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,
};
#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,
};
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_MULTI_CH] = "multi-channel-playback",
[USECASE_AUDIO_PLAYBACK_OFFLOAD] = "compress-offload-playback",
#ifdef MULTIPLE_OFFLOAD_ENABLED
[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",
#endif
[USECASE_AUDIO_RECORD] = "audio-record",
[USECASE_AUDIO_RECORD_COMPRESS] = "audio-record-compress",
[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",
};
static const audio_usecase_t offload_usecases[] = {
USECASE_AUDIO_PLAYBACK_OFFLOAD,
#ifdef MULTIPLE_OFFLOAD_ENABLED
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,
#endif
};
#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_QUAD),/* QUAD_BACK is same as QUAD */
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD_SIDE),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_PENTA),
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1), /* 5POINT1_BACK is same as 5POINT1 */
STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1_SIDE),
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),
};
static struct audio_device *adev = NULL;
static pthread_mutex_t adev_init_lock;
static unsigned int audio_device_ref_count;
static int set_voice_volume_l(struct audio_device *adev, float volume);
static int check_and_set_gapless_mode(struct audio_device *adev) {
char value[PROPERTY_VALUE_MAX] = {0};
bool gapless_enabled = false;
const char *mixer_ctl_name = "Compress Gapless Playback";
struct mixer_ctl *ctl;
ALOGV("%s:", __func__);
property_get("audio.offload.gapless.enabled", value, NULL);
gapless_enabled = atoi(value) || !strncmp("true", value, 4);
ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
if (!ctl) {
ALOGE("%s: Could not get ctl for mixer cmd - %s",
__func__, mixer_ctl_name);
return -EINVAL;
}
if (mixer_ctl_set_value(ctl, 0, gapless_enabled) < 0) {
ALOGE("%s: Could not set gapless mode %d",
__func__, gapless_enabled);
return -EINVAL;
}
return 0;
}
static bool is_supported_format(audio_format_t format)
{
if (format == AUDIO_FORMAT_MP3 ||
format == AUDIO_FORMAT_AAC_LC ||
format == AUDIO_FORMAT_AAC_HE_V1 ||
format == AUDIO_FORMAT_AAC_HE_V2 ||
format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD ||
format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD ||
format == AUDIO_FORMAT_FLAC ||
format == AUDIO_FORMAT_ALAC ||
format == AUDIO_FORMAT_APE ||
format == AUDIO_FORMAT_VORBIS ||
format == AUDIO_FORMAT_WMA ||
format == AUDIO_FORMAT_WMA_PRO)
return true;
return false;
}
static int get_snd_codec_id(audio_format_t format)
{
int id = 0;
switch (format & AUDIO_FORMAT_MAIN_MASK) {
case AUDIO_FORMAT_MP3:
id = SND_AUDIOCODEC_MP3;
break;
case AUDIO_FORMAT_AAC:
id = SND_AUDIOCODEC_AAC;
break;
case AUDIO_FORMAT_PCM_OFFLOAD:
id = SND_AUDIOCODEC_PCM;
break;
case AUDIO_FORMAT_FLAC:
id = SND_AUDIOCODEC_FLAC;
break;
case AUDIO_FORMAT_ALAC:
id = SND_AUDIOCODEC_ALAC;
break;
case AUDIO_FORMAT_APE:
id = SND_AUDIOCODEC_APE;
break;
case AUDIO_FORMAT_VORBIS:
id = SND_AUDIOCODEC_VORBIS;
break;
case AUDIO_FORMAT_WMA:
id = SND_AUDIOCODEC_WMA;
break;
case AUDIO_FORMAT_WMA_PRO:
id = SND_AUDIOCODEC_WMA_PRO;
break;
default:
ALOGE("%s: Unsupported audio format :%x", __func__, format);
}
return id;
}
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);
adev->snd_card_status.state = 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;
}
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];
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_audio_calibration(adev, usecase);
audio_extn_utils_send_app_type_cfg(usecase);
strcpy(mixer_path, use_case_table[usecase->id]);
platform_add_backend_name(mixer_path, snd_device);
ALOGV("%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;
strcpy(mixer_path, use_case_table[usecase->id]);
platform_add_backend_name(mixer_path, snd_device);
ALOGV("%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)
{
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);
/* start usb playback thread */
if(SND_DEVICE_OUT_USB_HEADSET == snd_device ||
SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET == snd_device)
audio_extn_usb_start_playback(adev);
/* start usb capture thread */
if(SND_DEVICE_IN_USB_HEADSET_MIC == snd_device)
audio_extn_usb_start_capture(adev);
if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
audio_extn_spkr_prot_is_enabled()) {
if (audio_extn_spkr_prot_get_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 {
ALOGV("%s: snd_device(%d: %s)", __func__,
snd_device, device_name);
/* 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);
}
return 0;
}
int disable_snd_device(struct audio_device *adev,
snd_device_t snd_device)
{
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) {
ALOGV("%s: snd_device(%d: %s)", __func__,
snd_device, device_name);
/* exit usb play back thread */
if(SND_DEVICE_OUT_USB_HEADSET == snd_device ||
SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET == snd_device)
audio_extn_usb_stop_playback();
/* exit usb capture thread */
if(SND_DEVICE_IN_USB_HEADSET_MIC == snd_device)
audio_extn_usb_stop_capture();
if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
audio_extn_spkr_prot_is_enabled()) {
audio_extn_spkr_prot_stop_processing(snd_device);
} else {
audio_route_reset_and_update_path(adev->audio_route, device_name);
}
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;
}
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];
int i, num_uc_to_switch = 0;
/*
* 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 playback.
*/
bool force_routing = platform_check_and_set_codec_backend_cfg(adev, uc_info);
/* Disable all the usecases on the shared backend other than the
* specified usecase.
* For native(44.1k) usecases, we don't need this as it uses a different
* backend, but we need to make sure that we reconfigure the backend
* if there is bit_width change, this should not affect shared backend
* usecases.
*/
for (i = 0; i < AUDIO_USECASE_MAX; i++)
switch_device[i] = false;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type != PCM_CAPTURE &&
usecase != uc_info &&
(usecase->out_snd_device != snd_device || force_routing) &&
usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND
&& usecase->stream.out->sample_rate != OUTPUT_SAMPLING_RATE_44100) {
ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
__func__, use_case_table[usecase->id],
platform_get_snd_device_name(usecase->out_snd_device));
disable_audio_route(adev, usecase);
switch_device[usecase->id] = true;
num_uc_to_switch++;
} else if (usecase->type == PCM_PLAYBACK &&
usecase->stream.out->sample_rate ==
OUTPUT_SAMPLING_RATE_44100 && force_routing){
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->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, 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 out_snd_device only before enabling the audio route */
if (switch_device[usecase->id] ) {
usecase->out_snd_device = snd_device;
if (usecase->type != VOICE_CALL)
enable_audio_route(adev, usecase);
}
}
}
}
static void check_and_route_capture_usecases(struct audio_device *adev,
struct audio_usecase *uc_info,
snd_device_t snd_device)
{
struct listnode *node;
struct audio_usecase *usecase;
bool switch_device[AUDIO_USECASE_MAX];
int i, num_uc_to_switch = 0;
/*
* This function is to make sure that all the active capture usecases
* are always routed to the same input sound device.
* For example, if audio-record and voice-call usecases are currently
* active on speaker(rx) and speaker-mic (tx) and out_set_parameters(earpiece)
* is received for voice call then we have to make sure that audio-record
* usecase is also switched to earpiece i.e. voice-dmic-ef,
* because of the limitation that two devices cannot be enabled
* at the same time if they share the same backend.
*/
for (i = 0; i < AUDIO_USECASE_MAX; i++)
switch_device[i] = false;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type != PCM_PLAYBACK &&
usecase != uc_info &&
usecase->in_snd_device != snd_device &&
(usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
(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)
enable_audio_route(adev, usecase);
}
}
}
}
/* must be called with hw device mutex locked */
static int read_hdmi_channel_masks(struct stream_out *out)
{
int ret = 0, i = 0;
int channels = platform_edid_get_max_channels(out->dev->platform);
switch (channels) {
/*
* Do not handle stereo output in Multi-channel cases
* Stereo case is handled in normal playback path
*/
case 6:
ALOGV("%s: HDMI supports Quad and 5.1", __func__);
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_QUAD;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_QUAD_SIDE;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_PENTA;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_5POINT1;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_5POINT1_SIDE;
break;
case 8:
ALOGV("%s: HDMI supports Quad, 5.1 and 7.1 channels", __func__);
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_QUAD;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_QUAD_SIDE;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_PENTA;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_5POINT1;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_5POINT1_SIDE;
out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_7POINT1;
break;
default:
ALOGE("HDMI does not support multi channel playback");
ret = -ENOSYS;
break;
}
return ret;
}
audio_usecase_t get_usecase_id_from_usecase_type(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(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;
}
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;
struct listnode *node;
int status = 0;
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)) {
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.
*/
if (voice_is_in_call(adev) && adev->mode == AUDIO_MODE_IN_CALL) {
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_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)) {
voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
if ((voip_usecase) && ((voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
(usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
(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) {
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 &&
adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION &&
out_snd_device != usecase->out_snd_device) {
select_devices(adev, adev->active_input->usecase);
}
}
} else if (usecase->type == PCM_CAPTURE) {
usecase->devices = usecase->stream.in->device;
out_snd_device = SND_DEVICE_NONE;
if (in_snd_device == SND_DEVICE_NONE) {
audio_devices_t out_device = AUDIO_DEVICE_NONE;
if ((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->platform, false);
} 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) {
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);
}
/* Disable current sound devices */
if (usecase->out_snd_device != SND_DEVICE_NONE) {
disable_audio_route(adev, usecase);
disable_snd_device(adev, usecase->out_snd_device);
}
if (usecase->in_snd_device != SND_DEVICE_NONE) {
disable_audio_route(adev, usecase);
disable_snd_device(adev, usecase->in_snd_device);
}
/* Applicable only on the targets that has external modem.
* New device information should be sent to modem before enabling
* the devices to reduce in-call device switch time.
*/
if ((usecase->type == VOICE_CALL) &&
(usecase->in_snd_device != SND_DEVICE_NONE) &&
(usecase->out_snd_device != SND_DEVICE_NONE)) {
status = platform_switch_voice_call_enable_device_config(adev->platform,
out_snd_device,
in_snd_device);
}
/* Enable new sound devices */
if (out_snd_device != SND_DEVICE_NONE) {
if (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)
check_usecases_codec_backend(adev, usecase, out_snd_device);
enable_snd_device(adev, out_snd_device);
}
if (in_snd_device != SND_DEVICE_NONE) {
check_and_route_capture_usecases(adev, usecase, in_snd_device);
enable_snd_device(adev, in_snd_device);
}
if (usecase->type == VOICE_CALL || 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;
if (usecase->type == PCM_PLAYBACK) {
audio_extn_utils_update_stream_app_type_cfg(adev->platform,
&adev->streams_output_cfg_list,
usecase->stream.out->devices,
usecase->stream.out->flags,
usecase->stream.out->format,
usecase->stream.out->sample_rate,
usecase->stream.out->bit_width,
&usecase->stream.out->app_type_cfg);
ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
}
enable_audio_route(adev, usecase);
/* 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 i, ret = 0;
struct audio_usecase *uc_info;
struct audio_device *adev = in->dev;
adev->active_input = NULL;
ALOGV("%s: enter: usecase(%d: %s)", __func__,
in->usecase, use_case_table[in->usecase]);
uc_info = get_usecase_from_list(adev, in->usecase);
if (uc_info == NULL) {
ALOGE("%s: Could not find the usecase (%d) in the list",
__func__, in->usecase);
return -EINVAL;
}
/* 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);
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);
in->usecase = platform_update_usecase_from_source(in->source,in->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
ALOGD("%s: Updated usecase(%d: %s)",
__func__, in->usecase, use_case_table[in->usecase]);
in->pcm_device_id = platform_get_pcm_device_id(in->usecase, PCM_CAPTURE);
if (in->pcm_device_id < 0) {
ALOGE("%s: Could not find PCM device id for the usecase(%d)",
__func__, in->usecase);
ret = -EINVAL;
goto error_config;
}
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();
select_devices(adev, in->usecase);
ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d",
__func__, adev->snd_card, in->pcm_device_id, in->config.channels);
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;
}
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;
}
audio_extn_perf_lock_release();
ALOGV("%s: exit", __func__);
return ret;
error_open:
stop_input_stream(in);
audio_extn_perf_lock_release();
error_config:
adev->active_input = NULL;
ALOGD("%s: exit: status(%d)", __func__, ret);
return ret;
}
/* 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)
{
audio_usecase_t ret = USECASE_AUDIO_PLAYBACK_OFFLOAD;
unsigned int i, num_usecase = sizeof(offload_usecases)/sizeof(offload_usecases[0]);
char value[PROPERTY_VALUE_MAX] = {0};
property_get("audio.offload.multiple.enabled", value, NULL);
if (!(atoi(value) || !strncmp("true", value, 4)))
num_usecase = 1; /* If prop is not set, limit the num of offload usecases to 1 */
ALOGV("%s: num_usecase: %d", __func__, num_usecase);
for (i = 0; i < num_usecase; i++) {
if (!(adev->offload_usecases_state & (0x1<<i))) {
adev->offload_usecases_state |= 0x1 << i;
ret = offload_usecases[i];
break;
}
}
ALOGV("%s: offload usecase is %d", __func__, ret);
return ret;
}
static void free_offload_usecase(struct audio_device *adev,
audio_usecase_t uc_id)
{
unsigned int i;
for (i = 0; i < sizeof(offload_usecases)/sizeof(offload_usecases[0]); i++) {
if (offload_usecases[i] == uc_id) {
adev->offload_usecases_state &= ~(0x1<<i);
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__);
pthread_mutex_lock(&out->lock);
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__);
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;
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;
}
pthread_mutex_lock(&out->lock);
out->offload_thread_blocked = false;
pthread_cond_signal(&out->cond);
if (send_callback) {
ALOGVV("%s: sending offload_callback event %d", __func__, event);
out->offload_callback(event, NULL, out->offload_cookie);
}
free(cmd);
}
pthread_cond_signal(&out->cond);
while (!list_empty(&out->offload_cmd_list)) {
item = list_head(&out->offload_cmd_list);
list_remove(item);
free(node_to_item(item, struct offload_cmd, node));
}
pthread_mutex_unlock(&out->lock);
return NULL;
}
static int create_offload_callback_thread(struct stream_out *out)
{
pthread_cond_init(&out->offload_cond, (const pthread_condattr_t *) NULL);
list_init(&out->offload_cmd_list);
pthread_create(&out->offload_thread, (const pthread_attr_t *) NULL,
offload_thread_loop, out);
return 0;
}
static int destroy_offload_callback_thread(struct stream_out *out)
{
pthread_mutex_lock(&out->lock);
stop_compressed_output_l(out);
send_offload_cmd_l(out, OFFLOAD_CMD_EXIT);
pthread_mutex_unlock(&out->lock);
pthread_join(out->offload_thread, (void **) NULL);
pthread_cond_destroy(&out->offload_cond);
return 0;
}
static bool allow_hdmi_channel_config(struct audio_device *adev)
{
struct listnode *node;
struct audio_usecase *usecase;
bool ret = true;
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
/*
* If voice call is already existing, do not proceed further to avoid
* disabling/enabling both RX and TX devices, CSD calls, etc.
* Once the voice call done, the HDMI channels can be configured to
* max channels of remaining use cases.
*/
if (usecase->id == USECASE_VOICE_CALL) {
ALOGD("%s: voice call is active, no change in HDMI channels",
__func__);
ret = false;
break;
} else if (usecase->id == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
ALOGD("%s: multi channel playback is active, "
"no change in HDMI channels", __func__);
ret = false;
break;
} else if (is_offload_usecase(usecase->id) &&
audio_channel_count_from_out_mask(usecase->stream.out->channel_mask) > 2) {
ALOGD("%s:multi-channel(%x) compress offload playback is active"
", no change in HDMI channels", __func__,
usecase->stream.out->channel_mask);
ret = false;
break;
}
}
}
return ret;
}
static int check_and_set_hdmi_channels(struct audio_device *adev,
unsigned int channels)
{
struct listnode *node;
struct audio_usecase *usecase;
unsigned int supported_channels = platform_edid_get_max_channels(
adev->platform);
ALOGV("supported_channels %d, channels %d", supported_channels, channels);
/* Check if change in HDMI channel config is allowed */
if (!allow_hdmi_channel_config(adev))
return 0;
if (channels > supported_channels)
channels = supported_channels;
if (channels == adev->cur_hdmi_channels) {
ALOGD("%s: Requested channels are same as current channels(%d)",
__func__, channels);
return 0;
}
/*TODO: CHECK for passthrough don't set channel map for passthrough*/
platform_set_hdmi_channels(adev->platform, channels);
platform_set_edid_channels_configuration(adev->platform, channels);
adev->cur_hdmi_channels = channels;
/*
* Deroute all the playback streams routed to HDMI so that
* the back end is deactivated. Note that backend will not
* be deactivated if any one stream is connected to it.
*/
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_PLAYBACK &&
usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
disable_audio_route(adev, usecase);
}
}
/*
* Enable all the streams disabled above. Now the HDMI backend
* will be activated with new channel configuration
*/
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase->type == PCM_PLAYBACK &&
usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
enable_audio_route(adev, usecase);
}
}
return 0;
}
static int stop_output_stream(struct stream_out *out)
{
int i, ret = 0;
struct audio_usecase *uc_info;
struct audio_device *adev = out->dev;
ALOGV("%s: enter: usecase(%d: %s)", __func__,
out->usecase, use_case_table[out->usecase]);
uc_info = get_usecase_from_list(adev, out->usecase);
if (uc_info == NULL) {
ALOGE("%s: Could not find the usecase (%d) in the list",
__func__, out->usecase);
return -EINVAL;
}
if (is_offload_usecase(out->usecase) &&
!(audio_extn_dolby_is_passthrough_stream(out->flags))) {
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) &&
(out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) &&
(audio_extn_dolby_is_passthrough_stream(out->flags))) {
ALOGV("Disable passthrough , reset mixer to pcm");
/* NO_PASSTHROUGH */
out->compr_config.codec->compr_passthr = 0;
audio_extn_dolby_set_hdmi_config(adev, 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)
check_and_set_hdmi_channels(adev, DEFAULT_HDMI_OUT_CHANNELS);
ALOGV("%s: exit: status(%d)", __func__, ret);
return ret;
}
int start_output_stream(struct stream_out *out)
{
int ret = 0;
int sink_channels = 0;
char prop_value[PROPERTY_VALUE_MAX] = {0};
struct audio_usecase *uc_info;
struct audio_device *adev = out->dev;
int snd_card_status = get_snd_card_state(adev);
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;
}
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;
/* This must be called before adding this usecase to the list */
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
if (is_offload_usecase(out->usecase)) {
if (audio_extn_dolby_is_passthrough_stream(out->flags)) {
audio_extn_dolby_update_passt_stream_configuration(adev, out);
}
}
property_get("audio.use.hdmi.sink.cap", prop_value, NULL);
if (!strncmp("true", prop_value, 4)) {
sink_channels = platform_edid_get_max_channels(out->dev->platform);
ALOGD("%s: set HDMI channel count[%d] based on sink capability",
__func__, sink_channels);
check_and_set_hdmi_channels(adev, sink_channels);
} else {
if (is_offload_usecase(out->usecase)) {
unsigned int ch_count = out->compr_config.codec->ch_in;
if (audio_extn_dolby_is_passthrough_stream(out->flags))
/* backend channel config for passthrough stream is stereo */
ch_count = 2;
check_and_set_hdmi_channels(adev, ch_count);
} else
check_and_set_hdmi_channels(adev, out->config.channels);
}
audio_extn_dolby_set_hdmi_config(adev, out);
}
list_add_tail(&adev->usecase_list, &uc_info->list);
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
flags |= PCM_MONOTONIC;
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);
} 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;
}
if (out->offload_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->use_small_bufs) {
compress_set_max_poll_wait(out->compr, 1000);
}
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_dolby_is_passthrough_stream(out->flags))) {
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);
audio_extn_check_and_set_dts_hpx_state(adev);
}
}
ALOGV("%s: exit", __func__);
return 0;
error_open:
stop_output_stream(out);
error_config:
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) &&
!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 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:
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;
/* ToDo: should use frame_size computed based on the format and
channel_count here. */
size *= sizeof(short) * 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 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 (is_offload_usecase(out->usecase))
return out->compr_config.fragment_size;
else if(out->usecase == USECASE_COMPRESS_VOIP_CALL)
return voice_extn_compress_voip_out_get_buffer_size(out);
return out->config.period_size *
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]);
if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
/* Ignore standby in case of voip call because the voip output
* stream is closed in adev_close_output_stream()
*/
ALOGD("%s: Ignore Standby in VOIP call", __func__);
return 0;
}
pthread_mutex_lock(&out->lock);
if (!out->standby) {
pthread_mutex_lock(&adev->lock);
out->standby = true;
if (!is_offload_usecase(out->usecase)) {
if (out->pcm) {
pcm_close(out->pcm);
out->pcm = NULL;
}
} else {
ALOGD("copl(%p):standby", out);
stop_compressed_output_l(out);
out->gapless_mdata.encoder_delay = 0;
out->gapless_mdata.encoder_padding = 0;
if (out->compr != NULL) {
compress_close(out->compr);
out->compr = NULL;
}
}
stop_output_stream(out);
pthread_mutex_unlock(&adev->lock);
}
pthread_mutex_unlock(&out->lock);
ALOGV("%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];
bool is_meta_data_params = false;
if (!out || !parms) {
ALOGE("%s: return invalid ",__func__);
return -EINVAL;
}
ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_FORMAT, value, sizeof(value));
if (ret >= 0) {
if (atoi(value) == SND_AUDIOSTREAMFORMAT_MP4ADTS) {
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
ALOGV("ADTS format is set in offload mode");
}
out->send_new_metadata = 1;
}
ret = audio_extn_parse_compress_metadata(out, parms);
ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_SAMPLE_RATE, value, sizeof(value));
if(ret >= 0)
is_meta_data_params = true;
ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_NUM_CHANNEL, value, sizeof(value));
if(ret >= 0)
is_meta_data_params = true;
ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE, value, sizeof(value));
if(ret >= 0)
is_meta_data_params = true;
ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value));
if (ret >= 0) {
is_meta_data_params = true;
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) {
is_meta_data_params = true;
out->gapless_mdata.encoder_padding = atoi(value);
}
if(!is_meta_data_params) {
ALOGV("%s: Not gapless meta data params", __func__);
return 0;
}
out->send_new_metadata = 1;
ALOGV("%s new encoder delay %u and padding %u", __func__,
out->gapless_mdata.encoder_delay, out->gapless_mdata.encoder_padding);
return 0;
}
static bool output_drives_call(struct audio_device *adev, struct stream_out *out)
{
return out == adev->primary_output || out == adev->voice_tx_output;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
struct audio_usecase *usecase;
struct listnode *node;
struct str_parms *parms;
char value[32];
int ret = 0, val = 0, err;
bool select_new_device = false;
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);
pthread_mutex_lock(&out->lock);
pthread_mutex_lock(&adev->lock);
/*
* When HDMI cable is unplugged/usb hs 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 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 ||
out->devices == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET) &&
val == AUDIO_DEVICE_NONE) {
if (!audio_extn_dolby_is_passthrough_stream(out->flags))
val = AUDIO_DEVICE_OUT_SPEAKER;
}
/*
* 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) {
out->devices = val;
if (!out->standby)
select_devices(adev, out->usecase);
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);
}
}
}
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)) {
pthread_mutex_lock(&out->lock);
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);
}
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 value[256];
struct str_parms *reply = str_parms_create();
size_t i, j;
int ret;
bool first = true;
if (!query || !reply) {
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) {
strcat(value, "|");
}
strcat(value, out_channels_name_to_enum_table[j].name);
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, 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) {
strcat(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);
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)
{
struct stream_out *out = (struct stream_out *)stream;
uint32_t latency = 0;
if (is_offload_usecase(out->usecase)) {
latency = COMPRESS_OFFLOAD_PLAYBACK_LATENCY;
} else {
latency = (out->config.period_count * out->config.period_size * 1000) /
(out->config.rate);
}
ALOGV("%s: Latency %d", __func__, latency);
return latency;
}
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_dolby_is_passthrough_stream(out->flags)) {
/*
* Set mute or umute on HDMI passthrough stream.
* Only take left channel into account.
* Mute is 0 and unmute 1
*/
audio_extn_dolby_set_passt_volume(out, (left == 0.0f));
} 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;
pthread_mutex_lock(&out->lock);
if (SND_CARD_STATE_OFFLINE == snd_scard_state) {
// increase written size during SSR to avoid mismatch
// with the written frames count in AF
if (!is_offload_usecase(out->usecase))
out->written += bytes / (out->config.channels * sizeof(short));
if (out->pcm) {
ALOGD(" %s: sound card is not active/SSR state", __func__);
ret= -EIO;
goto exit;
} else if (is_offload_usecase(out->usecase)) {
//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;
}
}
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 (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;
}
ret = compress_write(out->compr, buffer, bytes);
if (ret < 0)
ret = -errno;
ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret);
if (ret >= 0 && ret < (ssize_t)bytes) {
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 (!out->playback_started && ret >= 0) {
compress_start(out->compr);
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 (%d bytes) to pcm device", __func__, bytes);
if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY)
ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
else
ret = pcm_write(out->pcm, (void *)buffer, bytes);
if (ret < 0)
ret = -errno;
else if (ret == 0)
out->written += bytes / (out->config.channels * sizeof(short));
}
}
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 %ld - %s", __func__, 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);
usleep(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;
pthread_mutex_lock(&out->lock);
if (out->compr != NULL) {
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;
pthread_mutex_lock(&out->lock);
if (is_offload_usecase(out->usecase)) {
if (out->compr != NULL) {
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;
}
}
}
}
pthread_mutex_unlock(&out->lock);
return ret;
}
static int out_set_callback(struct audio_stream_out *stream,
stream_callback_t callback, void *cookie)
{
struct stream_out *out = (struct stream_out *)stream;
ALOGV("%s", __func__);
pthread_mutex_lock(&out->lock);
out->offload_callback = callback;
out->offload_cookie = cookie;
pthread_mutex_unlock(&out->lock);
return 0;
}
static int out_pause(struct audio_stream_out* stream)
{
struct stream_out *out = (struct stream_out *)stream;
int status = -ENOSYS;
ALOGV("%s", __func__);
if (is_offload_usecase(out->usecase)) {
ALOGD("copl(%p):pause compress driver", out);
pthread_mutex_lock(&out->lock);
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;
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;
pthread_mutex_lock(&out->lock);
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)
status = compress_resume(out->compr);
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)) {
pthread_mutex_lock(&out->lock);
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);
pthread_mutex_lock(&out->lock);
stop_compressed_output_l(out);
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);
return in->config.period_size *
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]);
if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
/* Ignore standby in case of voip call because the voip input
* stream is closed in adev_close_input_stream()
*/
ALOGV("%s: Ignore Standby in VOIP call", __func__);
return status;
}
pthread_mutex_lock(&in->lock);
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) {
pthread_mutex_lock(&adev->lock);
in->standby = true;
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 *str;
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;
pthread_mutex_lock(&in->lock);
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) &&
(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)
ret = select_devices(adev, in->usecase);
}
}
done:
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;
char value[256];
struct str_parms *reply = str_parms_create();
if (!query || !reply) {
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 i, ret = -1;
int snd_scard_state = get_snd_card_state(adev);
pthread_mutex_lock(&in->lock);
if (in->pcm) {
if(SND_CARD_STATE_OFFLINE == snd_scard_state) {
ALOGD(" %s: sound card is not active/SSR state", __func__);
ret= -EIO;;
goto exit;
} else {
if (in->is_st_session && !in->is_st_session_active) {
ALOGD(" %s: Sound trigger is not active/SSR", __func__);
ret= -EIO;;
goto exit;
}
}
}
if (in->standby) {
if (!in->is_st_session) {
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;
}
if (in->pcm) {
if (audio_extn_ssr_get_enabled() &&
audio_channel_count_from_in_mask(in->channel_mask) == 6)
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 (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY)
ret = pcm_mmap_read(in->pcm, buffer, bytes);
else
ret = pcm_read(in->pcm, buffer, bytes);
if (ret < 0)
ret = -errno;
}
/*
* 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) {
/* CPE SSR results in kernel returning ENETRESET for sound trigger
session reading on LAB data. In this case do not set sound card state
offline, instead mark this sound trigger session inactive to avoid
further reading of LAB data from CPE driver. Marking the session
inactive handles both CPE and ADSP SSR for sound trigger session */
if (!in->is_st_session)
set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
else
in->is_st_session_active = false;
}
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;
}
memset(buffer, 0, bytes);
in_standby(&in->stream.common);
ALOGV("%s: read failed status %d- sleeping for buffer duration", __func__, ret);
usleep(bytes * 1000000 / audio_stream_in_frame_size(stream) /
in_get_sample_rate(&in->stream.common));
}
return bytes;
}
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;
pthread_mutex_lock(&in->lock);
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);
}
static int adev_open_output_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
struct audio_config *config,
struct audio_stream_out **stream_out,
const char *address __unused)
{
struct audio_device *adev = (struct audio_device *)dev;
struct stream_out *out;
int i, ret = 0;
audio_format_t format;
*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: sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)\
stream_handle(%p)",__func__, 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_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->use_small_bufs = false;
/* Init use case and pcm_config */
if ((out->flags == AUDIO_OUTPUT_FLAG_DIRECT) &&
(out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ||
out->devices & AUDIO_DEVICE_OUT_PROXY)) {
pthread_mutex_lock(&adev->lock);
if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
ret = read_hdmi_channel_masks(out);
if (out->devices & AUDIO_DEVICE_OUT_PROXY)
ret = audio_extn_read_afe_proxy_channel_masks(out);
pthread_mutex_unlock(&adev->lock);
if (ret != 0)
goto error_open;
if (config->sample_rate == 0)
config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
if (config->channel_mask == 0)
config->channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
out->channel_mask = config->channel_mask;
out->sample_rate = config->sample_rate;
out->usecase = USECASE_AUDIO_PLAYBACK_MULTI_CH;
out->config = pcm_config_hdmi_multi;
out->config.rate = config->sample_rate;
out->config.channels = audio_channel_count_from_out_mask(out->channel_mask);
out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels * 2);
} else if ((out->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
(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) {
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) &&
((audio_extn_dolby_is_passthrough_stream(out->flags)))) {
ALOGV("read and update_pass through formats");
ret = audio_extn_dolby_update_passt_formats(adev, out);
if(ret != 0) {
goto error_open;
}
if(config->offload_info.format == 0)
config->offload_info.format = out->supported_formats[0];
}
if (!is_supported_format(config->offload_info.format) &&
!audio_extn_is_dolby_format(config->offload_info.format)) {
ALOGE("%s: Unsupported audio format", __func__);
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->usecase = get_offload_usecase(adev);
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;
}
format = out->format = config->offload_info.format;
out->sample_rate = config->offload_info.sample_rate;
out->stream.set_callback = out_set_callback;
out->stream.pause = out_pause;
out->stream.resume = out_resume;
out->stream.drain = out_drain;
out->stream.flush = out_flush;
out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
if (audio_extn_is_dolby_format(config->offload_info.format))
out->compr_config.codec->id =
audio_extn_dolby_get_snd_codec_id(adev, out,
config->offload_info.format);
else
out->compr_config.codec->id =
get_snd_codec_id(config->offload_info.format);
if (audio_is_offload_pcm(config->offload_info.format)) {
out->compr_config.fragment_size =
platform_get_pcm_offload_buffer_size(&config->offload_info);
} else if (audio_extn_dolby_is_passthrough_stream(out->flags)) {
out->compr_config.fragment_size =
audio_extn_dolby_get_passt_buffer_size(&config->offload_info);
} else {
out->compr_config.fragment_size =
platform_get_compress_offload_buffer_size(&config->offload_info);
}
out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
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(config->channel_mask);
out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
out->bit_width = PCM_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_AAC)
out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
if (config->offload_info.format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD)
out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
if(config->offload_info.format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD)
out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
if (out->bit_width == 24) {
out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
}
if (config->offload_info.format == AUDIO_FORMAT_FLAC)
out->compr_config.codec->options.flac_dec.sample_size = PCM_OUTPUT_BIT_WIDTH;
if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
out->non_blocking = 1;
if (config->offload_info.use_small_bufs) {
//this flag is set from framework only if its for PCM formats
//no need to check for PCM format again
out->non_blocking = 0;
out->use_small_bufs = true;
ALOGI("Keep write blocking for small buff: non_blockling %d",
out->non_blocking);
}
out->send_new_metadata = 1;
out->offload_state = OFFLOAD_STATE_IDLE;
out->playback_started = 0;
audio_extn_dts_create_state_notifier_node(out->usecase);
create_offload_callback_thread(out);
ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
__func__, config->offload_info.version,
config->offload_info.bit_rate);
//Decide if we need to use gapless mode by default
check_and_set_gapless_mode(adev);
} else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
ret = voice_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 if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
format = AUDIO_FORMAT_PCM_16_BIT;
out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
out->config = pcm_config_low_latency;
out->sample_rate = out->config.rate;
} else if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) {
format = AUDIO_FORMAT_PCM_16_BIT;
out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER;
out->config = pcm_config_deep_buffer;
out->sample_rate = out->config.rate;
} else {
/* primary path is the default path selected if no other outputs are available/suitable */
format = AUDIO_FORMAT_PCM_16_BIT;
out->usecase = USECASE_AUDIO_PLAYBACK_PRIMARY;
out->config = PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY;
out->sample_rate = out->config.rate;
}
ALOGV("%s devices %d,flags %x, format %x, out->sample_rate %d, out->bit_width %d",
__func__, devices, flags, format, out->sample_rate, out->bit_width);
/* TODO remove this hardcoding and check why width is zero*/
if (out->bit_width == 0)
out->bit_width = 16;
audio_extn_utils_update_stream_app_type_cfg(adev->platform,
&adev->streams_output_cfg_list,
devices, flags, format, out->sample_rate,
out->bit_width, &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->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);
ALOGV("%s: exit", __func__);
return 0;
error_open:
free(out);
*stream_out = NULL;
ALOGD("%s: exit: ret %d", __func__, ret);
return ret;
}
static void adev_close_output_stream(struct audio_hw_device *dev __unused,
struct audio_stream_out *stream)
{
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out->dev;
int ret = 0;
ALOGD("%s: enter:stream_handle(%p)",__func__, out);
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 (adev->voice_tx_output == out)
adev->voice_tx_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;
struct audio_usecase *usecase;
pthread_mutex_lock(&adev->lock);
list_for_each(node, &adev->usecase_list) {
usecase = node_to_item(node, struct audio_usecase, list);
if (usecase && is_offload_usecase(usecase->id)) {
if (usecase && 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);
}
}
}
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 *str;
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")) {
struct listnode *node;
struct audio_usecase *usecase;
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);
}
}
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 edid");
platform_cache_edid(adev->platform);
}
}
ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value, sizeof(value));
if (ret >= 0) {
val = atoi(value);
if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
ALOGV("invalidate cached edid");
platform_invalidate_edid(adev->platform);
}
}
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) {
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;
}
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 || mode == AUDIO_MODE_IN_COMMUNICATION) &&
voice_is_in_call(adev)) {
voice_stop_call(adev);
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 __unused,
audio_devices_t devices,
struct audio_config *config,
struct audio_stream_in **stream_in,
audio_input_flags_t flags __unused,
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;
*stream_in = NULL;
if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
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)",__func__, config->sample_rate, config->channel_mask,
devices, &in->stream, handle, source);
pthread_mutex_init(&in->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;
/* Update config params with the requested sample rate and channels */
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->config = pcm_config_audio_capture;
in->config.rate = config->sample_rate;
in->format = config->format;
if (in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
if (adev->mode != AUDIO_MODE_IN_CALL) {
ret = -EINVAL;
goto err_open;
}
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;
} else if (channel_count == 6) {
if(audio_extn_ssr_get_enabled()) {
if(audio_extn_ssr_init(in)) {
ALOGE("%s: audio_extn_ssr_init failed", __func__);
ret = -EINVAL;
goto err_open;
}
} else {
ALOGW("%s: surround sound recording is not supported", __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 {
in->config.channels = channel_count;
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_format_supported(in->format)) &&
(in->config.rate == 8000 || in->config.rate == 16000) &&
(audio_channel_count_from_in_mask(in->channel_mask) == 1)) {
voice_extn_compress_voip_open_input_stream(in);
}
}
/* This stream could be for sound trigger lab,
get sound trigger pcm if present */
audio_extn_sound_trigger_check_and_get_session(in);
audio_extn_perf_lock_init();
*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->platform, false);
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_enabled() &&
(audio_channel_count_from_in_mask(in->channel_mask) == 6)) {
audio_extn_ssr_deinit();
}
if(audio_extn_compr_cap_enabled() &&
audio_extn_compr_cap_format_supported(in->config.format))
audio_extn_compr_cap_deinit();
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_output_cfg_list(&adev->streams_output_cfg_list);
audio_route_free(adev->audio_route);
free(adev->snd_dev_ref_cnt);
platform_deinit(adev->platform);
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 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 i, ret;
ALOGD("%s: enter", __func__);
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL;
pthread_mutex_lock(&adev_init_lock);
if (audio_device_ref_count != 0){
*device = &adev->device.common;
audio_device_ref_count++;
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);
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->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;
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);
return -EINVAL;
}
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_listen_init(adev, adev->snd_card);
audio_extn_sound_trigger_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))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->bt_wb_speech_enabled = false;
audio_extn_ds2_enable(adev);
*device = &adev->device.common;
audio_extn_utils_update_streams_output_cfg_list(adev->platform, adev->mixer,
&adev->streams_output_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;
}
}
pthread_mutex_unlock(&adev_init_lock);
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,
},
};