/* * 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; } audio_usecase_t get_usecase_id_from_usecase_type(struct audio_device *adev, usecase_type_t type) { struct audio_usecase *usecase; struct listnode *node; list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if (usecase->type == type) { ALOGV("%s: usecase id %d", __func__, usecase->id); return usecase->id; } } return USECASE_INVALID; } struct audio_usecase *get_usecase_from_list(struct audio_device *adev, audio_usecase_t uc_id) { struct audio_usecase *usecase; struct listnode *node; list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if (usecase->id == uc_id) return usecase; } return NULL; } int select_devices(struct audio_device *adev, audio_usecase_t uc_id) { snd_device_t out_snd_device = SND_DEVICE_NONE; snd_device_t in_snd_device = SND_DEVICE_NONE; struct audio_usecase *usecase = NULL; struct audio_usecase *vc_usecase = NULL; struct audio_usecase *voip_usecase = NULL; struct audio_usecase *hfp_usecase = NULL; audio_usecase_t hfp_ucid; struct listnode *node; int status = 0; usecase = get_usecase_from_list(adev, uc_id); if (usecase == NULL) { ALOGE("%s: Could not find the usecase(%d)", __func__, uc_id); return -EINVAL; } if ((usecase->type == VOICE_CALL) || (usecase->type == VOIP_CALL) || (usecase->type == PCM_HFP_CALL)) { out_snd_device = platform_get_output_snd_device(adev->platform, usecase->stream.out->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_usecase_id_from_usecase_type(adev, VOICE_CALL)); if ((vc_usecase) && ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) || (usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL))) { in_snd_device = vc_usecase->in_snd_device; out_snd_device = vc_usecase->out_snd_device; } } else if (voice_extn_compress_voip_is_active(adev)) { voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL); if ((voip_usecase) && ((voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) && (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) && (voip_usecase->stream.out != adev->primary_output))) { in_snd_device = voip_usecase->in_snd_device; out_snd_device = voip_usecase->out_snd_device; } } else if (audio_extn_hfp_is_active(adev)) { hfp_ucid = audio_extn_hfp_get_usecase(); hfp_usecase = get_usecase_from_list(adev, hfp_ucid); if ((hfp_usecase) && (hfp_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) { in_snd_device = hfp_usecase->in_snd_device; out_snd_device = hfp_usecase->out_snd_device; } } if (usecase->type == PCM_PLAYBACK) { usecase->devices = usecase->stream.out->devices; in_snd_device = SND_DEVICE_NONE; if (out_snd_device == SND_DEVICE_NONE) { out_snd_device = platform_get_output_snd_device(adev->platform, usecase->stream.out->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 = 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, }, };