diff options
Diffstat (limited to 'msm8909/hal/audio_hw.c')
| -rw-r--r-- | msm8909/hal/audio_hw.c | 3539 |
1 files changed, 3539 insertions, 0 deletions
diff --git a/msm8909/hal/audio_hw.c b/msm8909/hal/audio_hw.c new file mode 100644 index 00000000..1c177fc0 --- /dev/null +++ b/msm8909/hal/audio_hw.c @@ -0,0 +1,3539 @@ +/* + * 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. + */ + +#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 96 +#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000 + +#define PROXY_OPEN_RETRY_COUNT 100 +#define PROXY_OPEN_WAIT_TIME 20 + +#define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_DEEP_BUFFER + +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_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_5POINT1), + 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->type == VOIP_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); + ALOGD("%s: apply mixer and update path: %s", __func__, mixer_path); + audio_route_apply_and_update_path(adev->audio_route, mixer_path); + ALOGV("%s: exit", __func__); + return 0; +} + +int disable_audio_route(struct audio_device *adev, + struct audio_usecase *usecase) +{ + snd_device_t snd_device; + char mixer_path[MIXER_PATH_MAX_LENGTH]; + + if (usecase == NULL || usecase->id == USECASE_INVALID) + return -EINVAL; + + ALOGV("%s: enter: usecase(%d)", __func__, usecase->id); + if (usecase->type == PCM_CAPTURE) + snd_device = usecase->in_snd_device; + else + snd_device = usecase->out_snd_device; + strcpy(mixer_path, use_case_table[usecase->id]); + platform_add_backend_name(mixer_path, snd_device); + ALOGD("%s: reset and update mixer path: %s", __func__, mixer_path); + audio_route_reset_and_update_path(adev->audio_route, mixer_path); + audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_FREE); + audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_FREE); + ALOGV("%s: exit", __func__); + return 0; +} + +int enable_snd_device(struct audio_device *adev, + snd_device_t snd_device) +{ + 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_OUT_BT_A2DP == snd_device || + (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP) == snd_device) + audio_extn_a2dp_start_playback(); + + 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; + } + if (audio_extn_spkr_prot_start_processing(snd_device)) { + ALOGE("%s: spkr_start_processing failed", __func__); + return -EINVAL; + } + } else { + ALOGD("%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) { + ALOGD("%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_OUT_BT_A2DP == snd_device || + (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP) == snd_device) + audio_extn_a2dp_stop_playback(); + + 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 (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) { + ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..", + __func__, use_case_table[usecase->id], + platform_get_snd_device_name(usecase->out_snd_device)); + disable_audio_route(adev, usecase); + switch_device[usecase->id] = true; + num_uc_to_switch++; + } + } + + if (num_uc_to_switch) { + /* 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 && usecase->type != VOIP_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) { + 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 && usecase->type != VOIP_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; + int channels = platform_edid_get_max_channels(out->dev->platform); + + switch (channels) { + /* + * Do not handle stereo output in Multi-channel cases + * Stereo case is handled in normal playback path + */ + case 6: + ALOGV("%s: HDMI supports 5.1", __func__); + out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1; + break; + case 8: + ALOGV("%s: HDMI supports 5.1 and 7.1 channels", __func__); + out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1; + out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_7POINT1; + break; + default: + ALOGE("HDMI does not support multi channel playback"); + ret = -ENOSYS; + break; + } + return ret; +} + +static audio_usecase_t get_voice_usecase_id_from_list(struct audio_device *adev) +{ + struct audio_usecase *usecase; + struct listnode *node; + + list_for_each(node, &adev->usecase_list) { + usecase = node_to_item(node, struct audio_usecase, list); + if (usecase->type == VOICE_CALL) { + ALOGV("%s: usecase id %d", __func__, usecase->id); + return usecase->id; + } + } + return USECASE_INVALID; +} + +struct audio_usecase *get_usecase_from_list(struct audio_device *adev, + audio_usecase_t uc_id) +{ + struct audio_usecase *usecase; + struct listnode *node; + + list_for_each(node, &adev->usecase_list) { + usecase = node_to_item(node, struct audio_usecase, list); + if (usecase->id == uc_id) + return usecase; + } + return NULL; +} + +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->devices); + 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_voice_usecase_id_from_list(adev)); + 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->devices); + if (usecase->stream.out == adev->primary_output && + adev->active_input && + out_snd_device != usecase->out_snd_device) { + select_devices(adev, adev->active_input->usecase); + } + } + } else if (usecase->type == PCM_CAPTURE) { + 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); + + int usecase = platform_update_usecase_from_source(in->source,in->usecase); + if (get_usecase_from_list(adev, usecase) == NULL) + in->usecase = usecase; + + ALOGD("%s: enter: stream(%p)usecase(%d: %s)", + __func__, &in->stream, in->usecase, use_case_table[in->usecase]); + + + if (SND_CARD_STATE_OFFLINE == snd_card_status) { + ALOGE("%s: sound card is not active/SSR returning error", __func__); + ret = -EIO; + goto error_config; + } + + /* Check if source matches incall recording usecase criteria */ + ret = voice_check_and_set_incall_rec_usecase(adev, in); + if (ret) + goto error_config; + else + ALOGV("%s: usecase(%d)", __func__, in->usecase); + + if (get_usecase_from_list(adev, in->usecase) != NULL) { + ALOGE("%s: use case assigned already in use, stream(%p)usecase(%d: %s)", + __func__, &in->stream, in->usecase, use_case_table[in->usecase]); + goto error_config; + } + + 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; + } + + ALOGV("%s: pcm_prepare", __func__); + ret = pcm_prepare(in->pcm); + if (ret < 0) { + ALOGE("%s: pcm_prepare returned %d", __func__, ret); + pcm_close(in->pcm); + in->pcm = NULL; + goto error_open; + } + + audio_extn_perf_lock_release(); + + ALOGD("%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) { + 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; + + /* Check if change in HDMI channel config is allowed */ + if (!allow_hdmi_channel_config(adev)) + return 0; + + if (channels == adev->cur_hdmi_channels) { + ALOGD("%s: Requested channels are same as current channels(%d)", __func__, channels); + return 0; + } + + platform_set_hdmi_channels(adev->platform, channels); + adev->cur_hdmi_channels = channels; + + /* + * Deroute all the playback streams routed to HDMI so that + * the back end is deactivated. Note that backend will not + * be deactivated if any one stream is connected to it. + */ + list_for_each(node, &adev->usecase_list) { + usecase = node_to_item(node, struct audio_usecase, list); + if (usecase->type == PCM_PLAYBACK && + usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { + disable_audio_route(adev, usecase); + } + } + + /* + * Enable all the streams disabled above. Now the HDMI backend + * will be activated with new channel configuration + */ + list_for_each(node, &adev->usecase_list) { + usecase = node_to_item(node, struct audio_usecase, list); + if (usecase->type == PCM_PLAYBACK && + usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { + enable_audio_route(adev, usecase); + } + } + + return 0; +} + +static int 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)) { + if (adev->visualizer_stop_output != NULL) + adev->visualizer_stop_output(out->handle, out->pcm_device_id); + if (adev->offload_effects_stop_output != NULL) + adev->offload_effects_stop_output(out->handle, out->pcm_device_id); + } + + /* 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); + + /* 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_config; + } + + 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) { + 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)) + check_and_set_hdmi_channels(adev, out->compr_config.codec->ch_in); + else + check_and_set_hdmi_channels(adev, out->config.channels); + } + } + + 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; + } + + ALOGV("%s: pcm_prepare", __func__); + if (pcm_is_ready(out->pcm)) { + ret = pcm_prepare(out->pcm); + if (ret < 0) { + ALOGE("%s: pcm_prepare returned %d", __func__, ret); + pcm_close(out->pcm); + out->pcm = NULL; + goto error_open; + } + } + } else { + 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); + +#ifdef DS1_DOLBY_DDP_ENABLED + if (audio_extn_is_dolby_format(out->format)) + audio_extn_dolby_send_ddp_endp_params(adev); +#endif + + 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); + } + + ALOGD("%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) { + 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 ((adev->mode == AUDIO_MODE_IN_CALL) && + output_drives_call(adev, out)) { + adev->current_call_output = out; + if (!voice_is_in_call(adev)) + ret = voice_start_call(adev); + else + 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); + 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 (!strncmp(str, "", sizeof(""))) { + free(str); + str = strdup(keys); + } + } + 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; + + if (is_offload_usecase(out->usecase)) + return COMPRESS_OFFLOAD_PLAYBACK_LATENCY; + + return (out->config.period_count * out->config.period_size * 1000) / + (out->config.rate); +} + +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)) { + 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { + //during SSR for compress usecase we should return error to flinger + ALOGD(" copl %s: sound card is not active/SSR state", __func__); + pthread_mutex_unlock(&out->lock); + return -ENETRESET; + } + } + + 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)) { + ALOGD("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); + out->playback_started = 1; + out->offload_state = OFFLOAD_STATE_PLAYING; + } + 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; + } + 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; + } + 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; + } + } + + 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)) + memset(buffer, 0, bytes); + +exit: + /* ToDo: There may be a corner case when SSR happens back to back during + start/stop. Need to post different error to handle that. */ + if (-ENETRESET == ret) { + set_snd_card_state(adev,SND_CARD_STATE_OFFLINE); + } + pthread_mutex_unlock(&in->lock); + + if (ret != 0) { + if (in->usecase == USECASE_COMPRESS_VOIP_CALL) { + pthread_mutex_lock(&adev->lock); + voice_extn_compress_voip_close_input_stream(&in->stream.common); + pthread_mutex_unlock(&adev->lock); + in->standby = true; + } + 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; + } + + 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; + + /* 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 (!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 { + 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 = + compress_get_alsa_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; + + 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; + + out->send_new_metadata = 1; + out->offload_state = OFFLOAD_STATE_IDLE; + out->playback_started = 0; + + 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 { + /* 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_deep_buffer; + 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); + 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() */ + + pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL); + pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL); + + config->format = out->stream.common.get_format(&out->stream.common); + config->channel_mask = out->stream.common.get_channels(&out->stream.common); + config->sample_rate = out->stream.common.get_sample_rate(&out->stream.common); + + *stream_out = &out->stream; + ALOGD("%s: Stream (%p) picks up usecase (%s)", __func__, &out->stream, + use_case_table[out->usecase]); + 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)) { + 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 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); + + pthread_mutex_lock(&adev->lock); + //close compress session on OFFLINE status + usecase = get_usecase_from_list(adev,USECASE_AUDIO_PLAYBACK_OFFLOAD); + if (usecase && usecase->stream.out) { + ALOGD(" %s closing compress session on OFFLINE state", __func__); + + struct stream_out *out = usecase->stream.out; + + pthread_mutex_unlock(&adev->lock); + out_standby(&out->stream.common); + } else + pthread_mutex_unlock(&adev->lock); + } else if (strstr(snd_card_status, "ONLINE")) { + ALOGD("Received sound card ONLINE status"); + set_snd_card_state(adev,SND_CARD_STATE_ONLINE); + if (!platform_is_acdb_initialized(adev->platform)) { + ret = platform_acdb_init(adev->platform); + if(ret) + ALOGE("acdb initialization is failed"); + + } + } + } + + 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; + } + + 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 __unused) +{ + 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)",__func__, config->sample_rate, config->channel_mask, + devices, &in->stream, handle); + + 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 = AUDIO_SOURCE_DEFAULT; + 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 (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; + } + + /* 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->cur_codec_backend_samplerate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE; + adev->cur_codec_backend_bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH; + 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->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, + }, +}; |