/* * Copyright (c) 2013-2017, The Linux Foundation. All rights reserved. * Not a Contribution. * * Copyright (C) 2013 The Android Open Source Project * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. * * This file was modified by DTS, Inc. The portions of the * code modified by DTS, Inc are copyrighted and * licensed separately, as follows: * * (C) 2014 DTS, Inc. * * Licensed under the Apache License, Version 2.0 (the "License"); * you may not use this file except in compliance with the License. * You may obtain a copy of the License at * * http://www.apache.org/licenses/LICENSE-2.0 * * Unless required by applicable law or agreed to in writing, software * distributed under the License is distributed on an "AS IS" BASIS, * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. * See the License for the specific language governing permissions and * limitations under the License. */ #define LOG_TAG "audio_hw_primary" #define ATRACE_TAG (ATRACE_TAG_AUDIO|ATRACE_TAG_HAL) /*#define LOG_NDEBUG 0*/ /*#define VERY_VERY_VERBOSE_LOGGING*/ #ifdef VERY_VERY_VERBOSE_LOGGING #define ALOGVV ALOGV #else #define ALOGVV(a...) do { } while(0) #endif #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include #include "audio_hw.h" #include "platform_api.h" #include #include "audio_extn.h" #include "voice_extn.h" #include "sound/compress_params.h" #include "sound/asound.h" #define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4 /*DIRECT PCM has same buffer sizes as DEEP Buffer*/ #define DIRECT_PCM_NUM_FRAGMENTS 2 #define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000 #define DSD_VOLUME_MIN_DB (-110) #define PROXY_OPEN_RETRY_COUNT 100 #define PROXY_OPEN_WAIT_TIME 20 #ifdef USE_LL_AS_PRIMARY_OUTPUT #define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_LOW_LATENCY #define PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY pcm_config_low_latency #else #define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_DEEP_BUFFER #define PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY pcm_config_deep_buffer #endif #define ULL_PERIOD_SIZE (DEFAULT_OUTPUT_SAMPLING_RATE/1000) static unsigned int configured_low_latency_capture_period_size = LOW_LATENCY_CAPTURE_PERIOD_SIZE; struct pcm_config pcm_config_deep_buffer = { .channels = 2, .rate = DEFAULT_OUTPUT_SAMPLING_RATE, .period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE, .period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT, .format = PCM_FORMAT_S16_LE, .start_threshold = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4, .stop_threshold = INT_MAX, .avail_min = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4, }; struct pcm_config pcm_config_low_latency = { .channels = 2, .rate = DEFAULT_OUTPUT_SAMPLING_RATE, .period_size = LOW_LATENCY_OUTPUT_PERIOD_SIZE, .period_count = LOW_LATENCY_OUTPUT_PERIOD_COUNT, .format = PCM_FORMAT_S16_LE, .start_threshold = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4, .stop_threshold = INT_MAX, .avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4, }; static int af_period_multiplier = 4; struct pcm_config pcm_config_rt = { .channels = 2, .rate = DEFAULT_OUTPUT_SAMPLING_RATE, .period_size = ULL_PERIOD_SIZE, //1 ms .period_count = 512, //=> buffer size is 512ms .format = PCM_FORMAT_S16_LE, .start_threshold = ULL_PERIOD_SIZE*8, //8ms .stop_threshold = INT_MAX, .silence_threshold = 0, .silence_size = 0, .avail_min = ULL_PERIOD_SIZE, //1 ms }; struct pcm_config pcm_config_hdmi_multi = { .channels = HDMI_MULTI_DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */ .rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */ .period_size = HDMI_MULTI_PERIOD_SIZE, .period_count = HDMI_MULTI_PERIOD_COUNT, .format = PCM_FORMAT_S16_LE, .start_threshold = 0, .stop_threshold = INT_MAX, .avail_min = 0, }; struct pcm_config pcm_config_audio_capture = { .channels = 2, .period_count = AUDIO_CAPTURE_PERIOD_COUNT, .format = PCM_FORMAT_S16_LE, }; struct pcm_config pcm_config_audio_capture_rt = { .channels = 2, .rate = DEFAULT_OUTPUT_SAMPLING_RATE, .period_size = ULL_PERIOD_SIZE, .period_count = 512, .format = PCM_FORMAT_S16_LE, .start_threshold = 0, .stop_threshold = INT_MAX, .silence_threshold = 0, .silence_size = 0, .avail_min = ULL_PERIOD_SIZE, //1 ms }; #define AFE_PROXY_CHANNEL_COUNT 2 #define AFE_PROXY_SAMPLING_RATE 48000 #define AFE_PROXY_PLAYBACK_PERIOD_SIZE 768 #define AFE_PROXY_PLAYBACK_PERIOD_COUNT 4 struct pcm_config pcm_config_afe_proxy_playback = { .channels = AFE_PROXY_CHANNEL_COUNT, .rate = AFE_PROXY_SAMPLING_RATE, .period_size = AFE_PROXY_PLAYBACK_PERIOD_SIZE, .period_count = AFE_PROXY_PLAYBACK_PERIOD_COUNT, .format = PCM_FORMAT_S16_LE, .start_threshold = AFE_PROXY_PLAYBACK_PERIOD_SIZE, .stop_threshold = INT_MAX, .avail_min = AFE_PROXY_PLAYBACK_PERIOD_SIZE, }; #define AFE_PROXY_RECORD_PERIOD_SIZE 768 #define AFE_PROXY_RECORD_PERIOD_COUNT 4 struct pcm_config pcm_config_afe_proxy_record = { .channels = AFE_PROXY_CHANNEL_COUNT, .rate = AFE_PROXY_SAMPLING_RATE, .period_size = AFE_PROXY_RECORD_PERIOD_SIZE, .period_count = AFE_PROXY_RECORD_PERIOD_COUNT, .format = PCM_FORMAT_S16_LE, .start_threshold = AFE_PROXY_RECORD_PERIOD_SIZE, .stop_threshold = INT_MAX, .avail_min = AFE_PROXY_RECORD_PERIOD_SIZE, }; #define AUDIO_MAX_PCM_FORMATS 7 const uint32_t format_to_bitwidth_table[AUDIO_MAX_PCM_FORMATS] = { [AUDIO_FORMAT_DEFAULT] = 0, [AUDIO_FORMAT_PCM_16_BIT] = sizeof(uint16_t), [AUDIO_FORMAT_PCM_8_BIT] = sizeof(uint8_t), [AUDIO_FORMAT_PCM_32_BIT] = sizeof(uint32_t), [AUDIO_FORMAT_PCM_8_24_BIT] = sizeof(uint32_t), [AUDIO_FORMAT_PCM_FLOAT] = sizeof(float), [AUDIO_FORMAT_PCM_24_BIT_PACKED] = sizeof(uint8_t) * 3, }; const char * const use_case_table[AUDIO_USECASE_MAX] = { [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = "deep-buffer-playback", [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = "low-latency-playback", [USECASE_AUDIO_PLAYBACK_ULL] = "audio-ull-playback", [USECASE_AUDIO_PLAYBACK_MULTI_CH] = "multi-channel-playback", [USECASE_AUDIO_PLAYBACK_OFFLOAD] = "compress-offload-playback", //Enabled for Direct_PCM [USECASE_AUDIO_PLAYBACK_OFFLOAD2] = "compress-offload-playback2", [USECASE_AUDIO_PLAYBACK_OFFLOAD3] = "compress-offload-playback3", [USECASE_AUDIO_PLAYBACK_OFFLOAD4] = "compress-offload-playback4", [USECASE_AUDIO_PLAYBACK_OFFLOAD5] = "compress-offload-playback5", [USECASE_AUDIO_PLAYBACK_OFFLOAD6] = "compress-offload-playback6", [USECASE_AUDIO_PLAYBACK_OFFLOAD7] = "compress-offload-playback7", [USECASE_AUDIO_PLAYBACK_OFFLOAD8] = "compress-offload-playback8", [USECASE_AUDIO_PLAYBACK_OFFLOAD9] = "compress-offload-playback9", [USECASE_AUDIO_RECORD] = "audio-record", [USECASE_AUDIO_RECORD_COMPRESS] = "audio-record-compress", [USECASE_AUDIO_RECORD_COMPRESS2] = "audio-record-compress2", [USECASE_AUDIO_RECORD_COMPRESS3] = "audio-record-compress3", [USECASE_AUDIO_RECORD_COMPRESS4] = "audio-record-compress4", [USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record", [USECASE_AUDIO_RECORD_FM_VIRTUAL] = "fm-virtual-record", [USECASE_AUDIO_PLAYBACK_FM] = "play-fm", [USECASE_AUDIO_HFP_SCO] = "hfp-sco", [USECASE_AUDIO_HFP_SCO_WB] = "hfp-sco-wb", [USECASE_VOICE_CALL] = "voice-call", [USECASE_VOICE2_CALL] = "voice2-call", [USECASE_VOLTE_CALL] = "volte-call", [USECASE_QCHAT_CALL] = "qchat-call", [USECASE_VOWLAN_CALL] = "vowlan-call", [USECASE_VOICEMMODE1_CALL] = "voicemmode1-call", [USECASE_VOICEMMODE2_CALL] = "voicemmode2-call", [USECASE_COMPRESS_VOIP_CALL] = "compress-voip-call", [USECASE_INCALL_REC_UPLINK] = "incall-rec-uplink", [USECASE_INCALL_REC_DOWNLINK] = "incall-rec-downlink", [USECASE_INCALL_REC_UPLINK_AND_DOWNLINK] = "incall-rec-uplink-and-downlink", [USECASE_INCALL_REC_UPLINK_COMPRESS] = "incall-rec-uplink-compress", [USECASE_INCALL_REC_DOWNLINK_COMPRESS] = "incall-rec-downlink-compress", [USECASE_INCALL_REC_UPLINK_AND_DOWNLINK_COMPRESS] = "incall-rec-uplink-and-downlink-compress", [USECASE_INCALL_MUSIC_UPLINK] = "incall_music_uplink", [USECASE_INCALL_MUSIC_UPLINK2] = "incall_music_uplink2", [USECASE_AUDIO_SPKR_CALIB_RX] = "spkr-rx-calib", [USECASE_AUDIO_SPKR_CALIB_TX] = "spkr-vi-record", [USECASE_AUDIO_PLAYBACK_AFE_PROXY] = "afe-proxy-playback", [USECASE_AUDIO_RECORD_AFE_PROXY] = "afe-proxy-record", [USECASE_AUDIO_PLAYBACK_EXT_DISP_SILENCE] = "silence-playback", /* Transcode loopback cases */ [USECASE_AUDIO_TRANSCODE_LOOPBACK] = "audio-transcode-loopback", /* For Interactive Audio Streams */ [USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM1] = "audio-interactive-stream1", [USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM2] = "audio-interactive-stream2", [USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM3] = "audio-interactive-stream3", [USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM4] = "audio-interactive-stream4", [USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM5] = "audio-interactive-stream5", [USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM6] = "audio-interactive-stream6", [USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM7] = "audio-interactive-stream7", [USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM8] = "audio-interactive-stream8", }; static const audio_usecase_t offload_usecases[] = { USECASE_AUDIO_PLAYBACK_OFFLOAD, USECASE_AUDIO_PLAYBACK_OFFLOAD2, USECASE_AUDIO_PLAYBACK_OFFLOAD3, USECASE_AUDIO_PLAYBACK_OFFLOAD4, USECASE_AUDIO_PLAYBACK_OFFLOAD5, USECASE_AUDIO_PLAYBACK_OFFLOAD6, USECASE_AUDIO_PLAYBACK_OFFLOAD7, USECASE_AUDIO_PLAYBACK_OFFLOAD8, USECASE_AUDIO_PLAYBACK_OFFLOAD9, }; static const audio_usecase_t interactive_usecases[] = { USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM1, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM2, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM3, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM4, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM5, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM6, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM7, USECASE_AUDIO_PLAYBACK_INTERACTIVE_STREAM8, }; #define STRING_TO_ENUM(string) { #string, string } struct string_to_enum { const char *name; uint32_t value; }; static const struct string_to_enum out_channels_name_to_enum_table[] = { STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO), STRING_TO_ENUM(AUDIO_CHANNEL_OUT_2POINT1), STRING_TO_ENUM(AUDIO_CHANNEL_OUT_QUAD), STRING_TO_ENUM(AUDIO_CHANNEL_OUT_SURROUND), STRING_TO_ENUM(AUDIO_CHANNEL_OUT_PENTA), STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1), STRING_TO_ENUM(AUDIO_CHANNEL_OUT_6POINT1), STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1), }; static const struct string_to_enum out_formats_name_to_enum_table[] = { STRING_TO_ENUM(AUDIO_FORMAT_AC3), STRING_TO_ENUM(AUDIO_FORMAT_E_AC3), STRING_TO_ENUM(AUDIO_FORMAT_E_AC3_JOC), STRING_TO_ENUM(AUDIO_FORMAT_DOLBY_TRUEHD), STRING_TO_ENUM(AUDIO_FORMAT_DTS), STRING_TO_ENUM(AUDIO_FORMAT_DTS_HD), STRING_TO_ENUM(AUDIO_FORMAT_IEC61937) }; //list of all supported sample rates by HDMI specification. static const int out_hdmi_sample_rates[] = { 32000, 44100, 48000, 88200, 96000, 176400, 192000, }; static const struct string_to_enum out_hdmi_sample_rates_name_to_enum_table[] = { STRING_TO_ENUM(32000), STRING_TO_ENUM(44100), STRING_TO_ENUM(48000), STRING_TO_ENUM(88200), STRING_TO_ENUM(96000), STRING_TO_ENUM(176400), STRING_TO_ENUM(192000), }; static struct audio_device *adev = NULL; static pthread_mutex_t adev_init_lock; static unsigned int audio_device_ref_count; //cache last MBDRC cal step level static int last_known_cal_step = -1 ; static bool may_use_noirq_mode(struct audio_device *adev, audio_usecase_t uc_id, int flags __unused) { int dir = 0; switch (uc_id) { case USECASE_AUDIO_RECORD_LOW_LATENCY: dir = 1; case USECASE_AUDIO_PLAYBACK_ULL: break; default: return false; } int dev_id = platform_get_pcm_device_id(uc_id, dir == 0 ? PCM_PLAYBACK : PCM_CAPTURE); if (adev->adm_is_noirq_avail) return adev->adm_is_noirq_avail(adev->adm_data, adev->snd_card, dev_id, dir); return false; } static void register_out_stream(struct stream_out *out) { struct audio_device *adev = out->dev; if (is_offload_usecase(out->usecase) || !adev->adm_register_output_stream) return; // register stream first for backward compatibility adev->adm_register_output_stream(adev->adm_data, out->handle, out->flags); if (!adev->adm_set_config) return; if (out->realtime) adev->adm_set_config(adev->adm_data, out->handle, out->pcm, &out->config); } static void register_in_stream(struct stream_in *in) { struct audio_device *adev = in->dev; if (!adev->adm_register_input_stream) return; adev->adm_register_input_stream(adev->adm_data, in->capture_handle, in->flags); if (!adev->adm_set_config) return; if (in->realtime) adev->adm_set_config(adev->adm_data, in->capture_handle, in->pcm, &in->config); } static void request_out_focus(struct stream_out *out, long ns) { struct audio_device *adev = out->dev; if (adev->adm_request_focus_v2) adev->adm_request_focus_v2(adev->adm_data, out->handle, ns); else if (adev->adm_request_focus) adev->adm_request_focus(adev->adm_data, out->handle); } static void request_in_focus(struct stream_in *in, long ns) { struct audio_device *adev = in->dev; if (adev->adm_request_focus_v2) adev->adm_request_focus_v2(adev->adm_data, in->capture_handle, ns); else if (adev->adm_request_focus) adev->adm_request_focus(adev->adm_data, in->capture_handle); } static void release_out_focus(struct stream_out *out) { struct audio_device *adev = out->dev; if (adev->adm_abandon_focus) adev->adm_abandon_focus(adev->adm_data, out->handle); } static void release_in_focus(struct stream_in *in) { struct audio_device *adev = in->dev; if (adev->adm_abandon_focus) adev->adm_abandon_focus(adev->adm_data, in->capture_handle); } __attribute__ ((visibility ("default"))) bool audio_hw_send_gain_dep_calibration(int level) { bool ret_val = false; ALOGV("%s: called ...", __func__); pthread_mutex_lock(&adev_init_lock); if (adev != NULL && adev->platform != NULL) { pthread_mutex_lock(&adev->lock); ret_val = platform_send_gain_dep_cal(adev->platform, level); // cache level info for any of the use case which // was not started. last_known_cal_step = level;; pthread_mutex_unlock(&adev->lock); } else { ALOGE("%s: %s is NULL", __func__, adev == NULL ? "adev" : "adev->platform"); } pthread_mutex_unlock(&adev_init_lock); return ret_val; } static int check_and_set_gapless_mode(struct audio_device *adev, bool enable_gapless) { bool gapless_enabled = false; const char *mixer_ctl_name = "Compress Gapless Playback"; struct mixer_ctl *ctl; ALOGV("%s:", __func__); gapless_enabled = property_get_bool("audio.offload.gapless.enabled", false); /*Disable gapless if its AV playback*/ gapless_enabled = gapless_enabled && enable_gapless; ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name); if (!ctl) { ALOGE("%s: Could not get ctl for mixer cmd - %s", __func__, mixer_ctl_name); return -EINVAL; } if (mixer_ctl_set_value(ctl, 0, gapless_enabled) < 0) { ALOGE("%s: Could not set gapless mode %d", __func__, gapless_enabled); return -EINVAL; } return 0; } __attribute__ ((visibility ("default"))) int audio_hw_get_gain_level_mapping(struct amp_db_and_gain_table *mapping_tbl, int table_size) { int ret_val = 0; ALOGV("%s: enter ... ", __func__); pthread_mutex_lock(&adev_init_lock); if (adev == NULL) { ALOGW("%s: adev is NULL .... ", __func__); goto done; } pthread_mutex_lock(&adev->lock); ret_val = platform_get_gain_level_mapping(mapping_tbl, table_size); pthread_mutex_unlock(&adev->lock); done: pthread_mutex_unlock(&adev_init_lock); ALOGV("%s: exit ... ", __func__); return ret_val; } static bool is_supported_format(audio_format_t format) { if (format == AUDIO_FORMAT_MP3 || format == AUDIO_FORMAT_MP2 || format == AUDIO_FORMAT_AAC_LC || format == AUDIO_FORMAT_AAC_HE_V1 || format == AUDIO_FORMAT_AAC_HE_V2 || format == AUDIO_FORMAT_AAC_ADTS_LC || format == AUDIO_FORMAT_AAC_ADTS_HE_V1 || format == AUDIO_FORMAT_AAC_ADTS_HE_V2 || format == AUDIO_FORMAT_AAC_LATM_LC || format == AUDIO_FORMAT_AAC_LATM_HE_V1 || format == AUDIO_FORMAT_AAC_LATM_HE_V2 || format == AUDIO_FORMAT_PCM_24_BIT_PACKED || format == AUDIO_FORMAT_PCM_8_24_BIT || format == AUDIO_FORMAT_PCM_FLOAT || format == AUDIO_FORMAT_PCM_32_BIT || format == AUDIO_FORMAT_PCM_16_BIT || format == AUDIO_FORMAT_AC3 || format == AUDIO_FORMAT_E_AC3 || format == AUDIO_FORMAT_DOLBY_TRUEHD || format == AUDIO_FORMAT_DTS || format == AUDIO_FORMAT_DTS_HD || format == AUDIO_FORMAT_FLAC || format == AUDIO_FORMAT_ALAC || format == AUDIO_FORMAT_APE || format == AUDIO_FORMAT_DSD || format == AUDIO_FORMAT_VORBIS || format == AUDIO_FORMAT_WMA || format == AUDIO_FORMAT_WMA_PRO || format == AUDIO_FORMAT_APTX || format == AUDIO_FORMAT_IEC61937) return true; return false; } static inline bool is_mmap_usecase(audio_usecase_t uc_id) { return (uc_id == USECASE_AUDIO_RECORD_AFE_PROXY) || (uc_id == USECASE_AUDIO_PLAYBACK_AFE_PROXY); } int get_snd_card_state(struct audio_device *adev) { int snd_scard_state; if (!adev) return SND_CARD_STATE_OFFLINE; pthread_mutex_lock(&adev->snd_card_status.lock); snd_scard_state = adev->snd_card_status.state; pthread_mutex_unlock(&adev->snd_card_status.lock); return snd_scard_state; } static int set_snd_card_state(struct audio_device *adev, int snd_scard_state) { if (!adev) return -ENOSYS; pthread_mutex_lock(&adev->snd_card_status.lock); if (adev->snd_card_status.state != snd_scard_state) { adev->snd_card_status.state = snd_scard_state; platform_snd_card_update(adev->platform, snd_scard_state); } pthread_mutex_unlock(&adev->snd_card_status.lock); return 0; } static int enable_audio_route_for_voice_usecases(struct audio_device *adev, struct audio_usecase *uc_info) { struct listnode *node; struct audio_usecase *usecase; if (uc_info == NULL) return -EINVAL; /* Re-route all voice usecases on the shared backend other than the specified usecase to new snd devices */ list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if ((usecase->type == VOICE_CALL) && (usecase != uc_info)) enable_audio_route(adev, usecase); } return 0; } static void enable_asrc_mode(struct audio_device *adev) { ALOGV("%s", __func__); audio_route_apply_and_update_path(adev->audio_route, "asrc-mode"); adev->asrc_mode_enabled = true; } static void disable_asrc_mode(struct audio_device *adev) { ALOGV("%s", __func__); audio_route_reset_and_update_path(adev->audio_route, "asrc-mode"); adev->asrc_mode_enabled = false; } /* * - Enable ASRC mode for incoming mix path use case(Headphone backend)if Headphone * 44.1 or Native DSD backends are enabled for any of current use case. * e.g. 48-> + (Naitve DSD or Headphone 44.1) * - Disable current mix path use case(Headphone backend) and re-enable it with * ASRC mode for incoming Headphone 44.1 or Native DSD use case. * e.g. Naitve DSD or Headphone 44.1 -> + 48 */ static void check_and_set_asrc_mode(struct audio_device *adev, struct audio_usecase *uc_info, snd_device_t snd_device) { ALOGV("%s snd device %d", __func__, snd_device); int i, num_new_devices = 0; snd_device_t split_new_snd_devices[SND_DEVICE_OUT_END]; /* *Split snd device for new combo use case *e.g. Headphopne 44.1-> + Ringtone (Headphone + Speaker) */ if (platform_split_snd_device(adev->platform, snd_device, &num_new_devices, split_new_snd_devices) == 0) { for (i = 0; i < num_new_devices; i++) check_and_set_asrc_mode(adev, uc_info, split_new_snd_devices[i]); } else { int new_backend_idx = platform_get_backend_index(snd_device); if (((new_backend_idx == HEADPHONE_BACKEND) || (new_backend_idx == HEADPHONE_44_1_BACKEND) || (new_backend_idx == DSD_NATIVE_BACKEND)) && !adev->asrc_mode_enabled) { struct listnode *node = NULL; struct audio_usecase *uc = NULL; struct stream_out *curr_out = NULL; int usecase_backend_idx = DEFAULT_CODEC_BACKEND; int i, num_devices, ret = 0; snd_device_t split_snd_devices[SND_DEVICE_OUT_END]; list_for_each(node, &adev->usecase_list) { uc = node_to_item(node, struct audio_usecase, list); curr_out = (struct stream_out*) uc->stream.out; if (curr_out && PCM_PLAYBACK == uc->type && uc != uc_info) { /* *Split snd device for existing combo use case *e.g. Ringtone (Headphone + Speaker) + Headphopne 44.1 */ ret = platform_split_snd_device(adev->platform, uc->out_snd_device, &num_devices, split_snd_devices); if (ret < 0 || num_devices == 0) { ALOGV("%s: Unable to split uc->out_snd_device: %d",__func__, uc->out_snd_device); split_snd_devices[0] = uc->out_snd_device; num_devices = 1; } for (i = 0; i < num_devices; i++) { usecase_backend_idx = platform_get_backend_index(split_snd_devices[i]); ALOGD("%s:snd_dev %d usecase_backend_idx %d",__func__, split_snd_devices[i],usecase_backend_idx); if((new_backend_idx == HEADPHONE_BACKEND) && ((usecase_backend_idx == HEADPHONE_44_1_BACKEND) || (usecase_backend_idx == DSD_NATIVE_BACKEND))) { ALOGD("%s:DSD or native stream detected enabling asrcmode in hardware", __func__); enable_asrc_mode(adev); break; } else if(((new_backend_idx == HEADPHONE_44_1_BACKEND) || (new_backend_idx == DSD_NATIVE_BACKEND)) && (usecase_backend_idx == HEADPHONE_BACKEND)) { ALOGD("%s:48K stream detected, disabling and enabling it with asrcmode in hardware", __func__); disable_audio_route(adev, uc); disable_snd_device(adev, uc->out_snd_device); // Apply true-high-quality-mode if DSD or > 44.1KHz or >=24-bit if (new_backend_idx == DSD_NATIVE_BACKEND) audio_route_apply_and_update_path(adev->audio_route, "hph-true-highquality-mode"); else if ((new_backend_idx == HEADPHONE_44_1_BACKEND) && (curr_out->bit_width >= 24)) audio_route_apply_and_update_path(adev->audio_route, "hph-highquality-mode"); enable_asrc_mode(adev); enable_snd_device(adev, uc->out_snd_device); enable_audio_route(adev, uc); break; } } // reset split devices count num_devices = 0; } if (adev->asrc_mode_enabled) break; } } } } int pcm_ioctl(struct pcm *pcm, int request, ...) { va_list ap; void * arg; int pcm_fd = *(int*)pcm; va_start(ap, request); arg = va_arg(ap, void *); va_end(ap); return ioctl(pcm_fd, request, arg); } int enable_audio_route(struct audio_device *adev, struct audio_usecase *usecase) { snd_device_t snd_device; char mixer_path[MIXER_PATH_MAX_LENGTH]; struct stream_out *out = NULL; if (usecase == NULL) return -EINVAL; ALOGV("%s: enter: usecase(%d)", __func__, usecase->id); if (usecase->type == PCM_CAPTURE) snd_device = usecase->in_snd_device; else snd_device = usecase->out_snd_device; #ifdef DS1_DOLBY_DAP_ENABLED audio_extn_dolby_set_dmid(adev); audio_extn_dolby_set_endpoint(adev); #endif audio_extn_dolby_ds2_set_endpoint(adev); audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_BUSY); audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_BUSY); audio_extn_utils_send_app_type_cfg(adev, usecase); audio_extn_utils_send_audio_calibration(adev, usecase); if ((usecase->type == PCM_PLAYBACK) && is_offload_usecase(usecase->id)) { out = usecase->stream.out; if (out && out->compr) audio_extn_utils_compress_set_clk_rec_mode(usecase); } strlcpy(mixer_path, use_case_table[usecase->id], MIXER_PATH_MAX_LENGTH); platform_add_backend_name(mixer_path, snd_device, usecase); ALOGD("%s: apply mixer and update path: %s", __func__, mixer_path); audio_route_apply_and_update_path(adev->audio_route, mixer_path); ALOGV("%s: exit", __func__); return 0; } int disable_audio_route(struct audio_device *adev, struct audio_usecase *usecase) { snd_device_t snd_device; char mixer_path[MIXER_PATH_MAX_LENGTH]; if (usecase == NULL || usecase->id == USECASE_INVALID) return -EINVAL; ALOGV("%s: enter: usecase(%d)", __func__, usecase->id); if (usecase->type == PCM_CAPTURE) snd_device = usecase->in_snd_device; else snd_device = usecase->out_snd_device; strlcpy(mixer_path, use_case_table[usecase->id], MIXER_PATH_MAX_LENGTH); platform_add_backend_name(mixer_path, snd_device, usecase); ALOGD("%s: reset and update mixer path: %s", __func__, mixer_path); audio_route_reset_and_update_path(adev->audio_route, mixer_path); audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_FREE); audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_FREE); ALOGV("%s: exit", __func__); return 0; } int enable_snd_device(struct audio_device *adev, snd_device_t snd_device) { int i, num_devices = 0; snd_device_t new_snd_devices[SND_DEVICE_OUT_END]; char device_name[DEVICE_NAME_MAX_SIZE] = {0}; if (snd_device < SND_DEVICE_MIN || snd_device >= SND_DEVICE_MAX) { ALOGE("%s: Invalid sound device %d", __func__, snd_device); return -EINVAL; } adev->snd_dev_ref_cnt[snd_device]++; if(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0 ) { ALOGE("%s: Invalid sound device returned", __func__); return -EINVAL; } if (adev->snd_dev_ref_cnt[snd_device] > 1) { ALOGV("%s: snd_device(%d: %s) is already active", __func__, snd_device, device_name); return 0; } if (audio_extn_spkr_prot_is_enabled()) audio_extn_spkr_prot_calib_cancel(adev); if (platform_can_enable_spkr_prot_on_device(snd_device) && audio_extn_spkr_prot_is_enabled()) { if (platform_get_spkr_prot_acdb_id(snd_device) < 0) { adev->snd_dev_ref_cnt[snd_device]--; return -EINVAL; } audio_extn_dev_arbi_acquire(snd_device); if (audio_extn_spkr_prot_start_processing(snd_device)) { ALOGE("%s: spkr_start_processing failed", __func__); audio_extn_dev_arbi_release(snd_device); return -EINVAL; } } else if (platform_split_snd_device(adev->platform, snd_device, &num_devices, new_snd_devices) == 0) { for (i = 0; i < num_devices; i++) { enable_snd_device(adev, new_snd_devices[i]); } } else { ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name); if ((SND_DEVICE_OUT_BT_A2DP == snd_device) && (audio_extn_a2dp_start_playback() < 0)) { ALOGE(" fail to configure A2dp control path "); return -EINVAL; } /* due to the possibility of calibration overwrite between listen and audio, notify listen hal before audio calibration is sent */ audio_extn_sound_trigger_update_device_status(snd_device, ST_EVENT_SND_DEVICE_BUSY); audio_extn_listen_update_device_status(snd_device, LISTEN_EVENT_SND_DEVICE_BUSY); if (platform_get_snd_device_acdb_id(snd_device) < 0) { adev->snd_dev_ref_cnt[snd_device]--; audio_extn_sound_trigger_update_device_status(snd_device, ST_EVENT_SND_DEVICE_FREE); audio_extn_listen_update_device_status(snd_device, LISTEN_EVENT_SND_DEVICE_FREE); return -EINVAL; } audio_extn_dev_arbi_acquire(snd_device); audio_route_apply_and_update_path(adev->audio_route, device_name); if (SND_DEVICE_OUT_HEADPHONES == snd_device && !adev->native_playback_enabled && audio_is_true_native_stream_active(adev)) { ALOGD("%s: %d: napb: enabling native mode in hardware", __func__, __LINE__); audio_route_apply_and_update_path(adev->audio_route, "true-native-mode"); adev->native_playback_enabled = true; } } return 0; } int disable_snd_device(struct audio_device *adev, snd_device_t snd_device) { int i, num_devices = 0; snd_device_t new_snd_devices[SND_DEVICE_OUT_END]; char device_name[DEVICE_NAME_MAX_SIZE] = {0}; if (snd_device < SND_DEVICE_MIN || snd_device >= SND_DEVICE_MAX) { ALOGE("%s: Invalid sound device %d", __func__, snd_device); return -EINVAL; } if (adev->snd_dev_ref_cnt[snd_device] <= 0) { ALOGE("%s: device ref cnt is already 0", __func__); return -EINVAL; } adev->snd_dev_ref_cnt[snd_device]--; if(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0) { ALOGE("%s: Invalid sound device returned", __func__); return -EINVAL; } if (adev->snd_dev_ref_cnt[snd_device] == 0) { ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name); if (platform_can_enable_spkr_prot_on_device(snd_device) && audio_extn_spkr_prot_is_enabled()) { audio_extn_spkr_prot_stop_processing(snd_device); } else if (platform_split_snd_device(adev->platform, snd_device, &num_devices, new_snd_devices) == 0) { for (i = 0; i < num_devices; i++) { disable_snd_device(adev, new_snd_devices[i]); } } else { audio_route_reset_and_update_path(adev->audio_route, device_name); } if (SND_DEVICE_OUT_BT_A2DP == snd_device) audio_extn_a2dp_stop_playback(); if (snd_device == SND_DEVICE_OUT_HDMI || snd_device == SND_DEVICE_OUT_DISPLAY_PORT) adev->is_channel_status_set = false; else if (SND_DEVICE_OUT_HEADPHONES == snd_device && adev->native_playback_enabled) { ALOGD("%s: %d: napb: disabling native mode in hardware", __func__, __LINE__); audio_route_reset_and_update_path(adev->audio_route, "true-native-mode"); adev->native_playback_enabled = false; } else if (SND_DEVICE_OUT_HEADPHONES == snd_device && adev->asrc_mode_enabled) { ALOGD("%s: %d: disabling asrc mode in hardware", __func__, __LINE__); disable_asrc_mode(adev); audio_route_apply_and_update_path(adev->audio_route, "hph-lowpower-mode"); } audio_extn_dev_arbi_release(snd_device); audio_extn_sound_trigger_update_device_status(snd_device, ST_EVENT_SND_DEVICE_FREE); audio_extn_listen_update_device_status(snd_device, LISTEN_EVENT_SND_DEVICE_FREE); } return 0; } /* legend: uc - existing usecase new_uc - new usecase d1, d11, d2 - SND_DEVICE enums a1, a2 - corresponding ANDROID device enums B1, B2 - backend strings case 1 uc->dev d1 (a1) B1 new_uc->dev d1 (a1), d2 (a2) B1, B2 resolution: disable and enable uc->dev on d1 case 2 uc->dev d1 (a1) B1 new_uc->dev d11 (a1) B1 resolution: need to switch uc since d1 and d11 are related (e.g. speaker and voice-speaker) use ANDROID_DEVICE_OUT enums to match devices since SND_DEVICE enums may vary case 3 uc->dev d1 (a1) B1 new_uc->dev d2 (a2) B2 resolution: no need to switch uc case 4 uc->dev d1 (a1) B1 new_uc->dev d2 (a2) B1 resolution: disable enable uc-dev on d2 since backends match we cannot enable two streams on two different devices if they share the same backend. e.g. if offload is on speaker device using QUAD_MI2S backend and a low-latency stream is started on voice-handset using the same backend, offload must also be switched to voice-handset. case 5 uc->dev d1 (a1) B1 new_uc->dev d1 (a1), d2 (a2) B1 resolution: disable enable uc-dev on d2 since backends match we cannot enable two streams on two different devices if they share the same backend. case 6 uc->dev d1 (a1) B1 new_uc->dev d2 (a1) B2 resolution: no need to switch case 7 uc->dev d1 (a1), d2 (a2) B1, B2 new_uc->dev d1 (a1) B1 resolution: no need to switch */ static snd_device_t derive_playback_snd_device(void * platform, struct audio_usecase *uc, struct audio_usecase *new_uc, snd_device_t new_snd_device) { audio_devices_t a1, a2; snd_device_t d1 = uc->out_snd_device; snd_device_t d2 = new_snd_device; switch (uc->type) { case TRANSCODE_LOOPBACK : a1 = uc->stream.inout->out_config.devices; a2 = new_uc->stream.inout->out_config.devices; break; default : a1 = uc->stream.out->devices; a2 = new_uc->stream.out->devices; break; } // Treat as a special case when a1 and a2 are not disjoint if ((a1 != a2) && (a1 & a2)) { snd_device_t d3[2]; int num_devices = 0; int ret = platform_split_snd_device(platform, popcount(a1) > 1 ? d1 : d2, &num_devices, d3); if (ret < 0) { if (ret != -ENOSYS) { ALOGW("%s failed to split snd_device %d", __func__, popcount(a1) > 1 ? d1 : d2); } goto end; } // NB: case 7 is hypothetical and isn't a practical usecase yet. // But if it does happen, we need to give priority to d2 if // the combo devices active on the existing usecase share a backend. // This is because we cannot have a usecase active on a combo device // and a new usecase requests one device in this combo pair. if (platform_check_backends_match(d3[0], d3[1])) { return d2; // case 5 } else { return d1; // case 1 } } else { if (platform_check_backends_match(d1, d2)) { return d2; // case 2, 4 } else { return d1; // case 6, 3 } } end: return d2; // return whatever was calculated before. } static void check_usecases_codec_backend(struct audio_device *adev, struct audio_usecase *uc_info, snd_device_t snd_device) { struct listnode *node; struct audio_usecase *usecase; bool switch_device[AUDIO_USECASE_MAX]; snd_device_t uc_derive_snd_device; snd_device_t derive_snd_device[AUDIO_USECASE_MAX]; int i, num_uc_to_switch = 0; int status = 0; bool force_restart_session = false; /* * This function is to make sure that all the usecases that are active on * the hardware codec backend are always routed to any one device that is * handled by the hardware codec. * For example, if low-latency and deep-buffer usecases are currently active * on speaker and out_set_parameters(headset) is received on low-latency * output, then we have to make sure deep-buffer is also switched to headset, * because of the limitation that both the devices cannot be enabled * at the same time as they share the same backend. */ /* * This call is to check if we need to force routing for a particular stream * If there is a backend configuration change for the device when a * new stream starts, then ADM needs to be closed and re-opened with the new * configuraion. This call check if we need to re-route all the streams * associated with the backend. Touch tone + 24 bit + native playback. */ bool force_routing = platform_check_and_set_codec_backend_cfg(adev, uc_info, snd_device); /* For a2dp device reconfigure all active sessions * with new AFE encoder format based on a2dp state */ if ((SND_DEVICE_OUT_BT_A2DP == snd_device || SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP == snd_device) && audio_extn_a2dp_is_force_device_switch()) { force_routing = true; force_restart_session = true; } ALOGD("%s:becf: force routing %d", __func__, force_routing); /* Disable all the usecases on the shared backend other than the * specified usecase. */ for (i = 0; i < AUDIO_USECASE_MAX; i++) switch_device[i] = false; list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); ALOGD("%s:becf: (%d) check_usecases curr device: %s, usecase device:%s " "backends match %d",__func__, i, platform_get_snd_device_name(snd_device), platform_get_snd_device_name(usecase->out_snd_device), platform_check_backends_match(snd_device, usecase->out_snd_device)); if ((usecase->type != PCM_CAPTURE) && (usecase != uc_info)) { uc_derive_snd_device = derive_playback_snd_device(adev->platform, usecase, uc_info, snd_device); if (((uc_derive_snd_device != usecase->out_snd_device) || force_routing) && ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) || (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) || (usecase->devices & AUDIO_DEVICE_OUT_USB_DEVICE) || (usecase->devices & AUDIO_DEVICE_OUT_ALL_A2DP) || (usecase->devices & AUDIO_DEVICE_OUT_ALL_SCO)) && ((force_restart_session) || (platform_check_backends_match(snd_device, usecase->out_snd_device)))) { ALOGD("%s:becf: check_usecases (%s) is active on (%s) - disabling ..", __func__, use_case_table[usecase->id], platform_get_snd_device_name(usecase->out_snd_device)); disable_audio_route(adev, usecase); switch_device[usecase->id] = true; /* Enable existing usecase on derived playback device */ derive_snd_device[usecase->id] = uc_derive_snd_device; num_uc_to_switch++; } } } ALOGD("%s:becf: check_usecases num.of Usecases to switch %d", __func__, num_uc_to_switch); if (num_uc_to_switch) { /* All streams have been de-routed. Disable the device */ /* Make sure the previous devices to be disabled first and then enable the selected devices */ list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if (switch_device[usecase->id]) { disable_snd_device(adev, usecase->out_snd_device); } } list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if (switch_device[usecase->id]) { enable_snd_device(adev, derive_snd_device[usecase->id]); } } /* Re-route all the usecases on the shared backend other than the specified usecase to new snd devices */ list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); /* Update the out_snd_device only before enabling the audio route */ if (switch_device[usecase->id]) { usecase->out_snd_device = derive_snd_device[usecase->id]; if (usecase->type != VOICE_CALL) { ALOGD("%s:becf: enabling usecase (%s) on (%s)", __func__, use_case_table[usecase->id], platform_get_snd_device_name(usecase->out_snd_device)); /* Update voc calibration before enabling VoIP route */ if (usecase->type == VOIP_CALL) status = platform_switch_voice_call_device_post(adev->platform, usecase->out_snd_device, platform_get_input_snd_device(adev->platform, uc_info->devices)); enable_audio_route(adev, usecase); } } } } } static void check_usecases_capture_codec_backend(struct audio_device *adev, struct audio_usecase *uc_info, snd_device_t snd_device) { struct listnode *node; struct audio_usecase *usecase; bool switch_device[AUDIO_USECASE_MAX]; int i, num_uc_to_switch = 0; int backend_check_cond = AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND; int status = 0; bool force_routing = platform_check_and_set_capture_codec_backend_cfg(adev, uc_info, snd_device); ALOGD("%s:becf: force routing %d", __func__, force_routing); /* * Make sure out devices is checked against out codec backend device and * also in devices against in codec backend. Checking out device against in * codec backend or vice versa causes issues. */ if (uc_info->type == PCM_CAPTURE) backend_check_cond = AUDIO_DEVICE_IN_ALL_CODEC_BACKEND; /* * This function is to make sure that all the active capture usecases * are always routed to the same input sound device. * For example, if audio-record and voice-call usecases are currently * active on speaker(rx) and speaker-mic (tx) and out_set_parameters(earpiece) * is received for voice call then we have to make sure that audio-record * usecase is also switched to earpiece i.e. voice-dmic-ef, * because of the limitation that two devices cannot be enabled * at the same time if they share the same backend. */ for (i = 0; i < AUDIO_USECASE_MAX; i++) switch_device[i] = false; list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); /* * TODO: Enhance below condition to handle BT sco/USB multi recording */ if (usecase->type != PCM_PLAYBACK && usecase != uc_info && (usecase->in_snd_device != snd_device || force_routing) && ((uc_info->devices & backend_check_cond) && (((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND) || (usecase->type == VOIP_CALL))) && (usecase->id != USECASE_AUDIO_SPKR_CALIB_TX)) { ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..", __func__, use_case_table[usecase->id], platform_get_snd_device_name(usecase->in_snd_device)); disable_audio_route(adev, usecase); switch_device[usecase->id] = true; num_uc_to_switch++; } } if (num_uc_to_switch) { /* All streams have been de-routed. Disable the device */ /* Make sure the previous devices to be disabled first and then enable the selected devices */ list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if (switch_device[usecase->id]) { disable_snd_device(adev, usecase->in_snd_device); } } list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if (switch_device[usecase->id]) { enable_snd_device(adev, snd_device); } } /* Re-route all the usecases on the shared backend other than the specified usecase to new snd devices */ list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); /* Update the in_snd_device only before enabling the audio route */ if (switch_device[usecase->id] ) { usecase->in_snd_device = snd_device; if (usecase->type != VOICE_CALL) { /* Update voc calibration before enabling VoIP route */ if (usecase->type == VOIP_CALL) status = platform_switch_voice_call_device_post(adev->platform, platform_get_output_snd_device(adev->platform, uc_info->stream.out), usecase->in_snd_device); enable_audio_route(adev, usecase); } } } } } static void reset_hdmi_sink_caps(struct stream_out *out) { int i = 0; for (i = 0; i<= MAX_SUPPORTED_CHANNEL_MASKS; i++) { out->supported_channel_masks[i] = 0; } for (i = 0; i<= MAX_SUPPORTED_FORMATS; i++) { out->supported_formats[i] = 0; } for (i = 0; i<= MAX_SUPPORTED_SAMPLE_RATES; i++) { out->supported_sample_rates[i] = 0; } } /* must be called with hw device mutex locked */ static int read_hdmi_sink_caps(struct stream_out *out) { int ret = 0, i = 0, j = 0; int channels = platform_edid_get_max_channels(out->dev->platform); reset_hdmi_sink_caps(out); /* Cache ext disp type */ if (platform_get_ext_disp_type(adev->platform) <= 0) { ALOGE("%s: Failed to query disp type, ret:%d", __func__, ret); return -EINVAL; } switch (channels) { case 8: ALOGV("%s: HDMI supports 7.1 channels", __func__); out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_7POINT1; out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_6POINT1; case 6: ALOGV("%s: HDMI supports 5.1 channels", __func__); out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_5POINT1; out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_PENTA; out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_QUAD; out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_SURROUND; out->supported_channel_masks[i++] = AUDIO_CHANNEL_OUT_2POINT1; break; default: ALOGE("invalid/nonstandard channal count[%d]",channels); ret = -ENOSYS; break; } // check channel format caps i = 0; if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_AC3)) { ALOGV(":%s HDMI supports AC3/EAC3 formats", __func__); out->supported_formats[i++] = AUDIO_FORMAT_AC3; //Adding EAC3/EAC3_JOC formats if AC3 is supported by the sink. //EAC3/EAC3_JOC will be converted to AC3 for decoding if needed out->supported_formats[i++] = AUDIO_FORMAT_E_AC3; out->supported_formats[i++] = AUDIO_FORMAT_E_AC3_JOC; } if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_DOLBY_TRUEHD)) { ALOGV(":%s HDMI supports TRUE HD format", __func__); out->supported_formats[i++] = AUDIO_FORMAT_DOLBY_TRUEHD; } if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_DTS)) { ALOGV(":%s HDMI supports DTS format", __func__); out->supported_formats[i++] = AUDIO_FORMAT_DTS; } if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_DTS_HD)) { ALOGV(":%s HDMI supports DTS HD format", __func__); out->supported_formats[i++] = AUDIO_FORMAT_DTS_HD; } if (platform_is_edid_supported_format(out->dev->platform, AUDIO_FORMAT_IEC61937)) { ALOGV(":%s HDMI supports IEC61937 format", __func__); out->supported_formats[i++] = AUDIO_FORMAT_IEC61937; } // check sample rate caps i = 0; for (j = 0; j < MAX_SUPPORTED_SAMPLE_RATES; j++) { if (platform_is_edid_supported_sample_rate(out->dev->platform, out_hdmi_sample_rates[j])) { ALOGV(":%s HDMI supports sample rate:%d", __func__, out_hdmi_sample_rates[j]); out->supported_sample_rates[i++] = out_hdmi_sample_rates[j]; } } return ret; } audio_usecase_t get_usecase_id_from_usecase_type(const struct audio_device *adev, usecase_type_t type) { struct audio_usecase *usecase; struct listnode *node; list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if (usecase->type == type) { ALOGV("%s: usecase id %d", __func__, usecase->id); return usecase->id; } } return USECASE_INVALID; } struct audio_usecase *get_usecase_from_list(const struct audio_device *adev, audio_usecase_t uc_id) { struct audio_usecase *usecase; struct listnode *node; list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if (usecase->id == uc_id) return usecase; } return NULL; } struct stream_in *get_next_active_input(const struct audio_device *adev) { struct audio_usecase *usecase; struct listnode *node; list_for_each_reverse(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if (usecase->type == PCM_CAPTURE) return usecase->stream.in; } return NULL; } /* * is a true native playback active */ bool audio_is_true_native_stream_active(struct audio_device *adev) { bool active = false; int i = 0; struct listnode *node; if (NATIVE_AUDIO_MODE_TRUE_44_1 != platform_get_native_support()) { ALOGV("%s:napb: not in true mode or non hdphones device", __func__); active = false; goto exit; } list_for_each(node, &adev->usecase_list) { struct audio_usecase *uc; uc = node_to_item(node, struct audio_usecase, list); struct stream_out *curr_out = (struct stream_out*) uc->stream.out; if (curr_out && PCM_PLAYBACK == uc->type) { ALOGD("%s:napb: (%d) (%s)id (%d) sr %d bw " "(%d) device %s", __func__, i++, use_case_table[uc->id], uc->id, curr_out->sample_rate, curr_out->bit_width, platform_get_snd_device_name(uc->out_snd_device)); if (is_offload_usecase(uc->id) && (curr_out->sample_rate == OUTPUT_SAMPLING_RATE_44100)) { active = true; ALOGD("%s:napb:native stream detected", __func__); } } } exit: return active; } /* * if native DSD playback active */ bool audio_is_dsd_native_stream_active(struct audio_device *adev) { bool active = false; struct listnode *node = NULL; struct audio_usecase *uc = NULL; struct stream_out *curr_out = NULL; list_for_each(node, &adev->usecase_list) { uc = node_to_item(node, struct audio_usecase, list); curr_out = (struct stream_out*) uc->stream.out; if (curr_out && PCM_PLAYBACK == uc->type && (DSD_NATIVE_BACKEND == platform_get_backend_index(uc->out_snd_device))) { active = true; ALOGV("%s:DSD playback is active", __func__); break; } } return active; } static bool force_device_switch(struct audio_usecase *usecase) { bool ret = false; bool is_it_true_mode = false; if (is_offload_usecase(usecase->id) && (usecase->stream.out) && (usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) && (usecase->stream.out->devices == AUDIO_DEVICE_OUT_WIRED_HEADSET || usecase->stream.out->devices == AUDIO_DEVICE_OUT_WIRED_HEADPHONE)) { is_it_true_mode = (NATIVE_AUDIO_MODE_TRUE_44_1 == platform_get_native_support()? true : false); if ((is_it_true_mode && !adev->native_playback_enabled) || (!is_it_true_mode && adev->native_playback_enabled)){ ret = true; ALOGD("napb: time to toggle native mode"); } } // Force all a2dp output devices to reconfigure for proper AFE encode format //Also handle a case where in earlier a2dp start failed as A2DP stream was //in suspended state, hence try to trigger a retry when we again get a routing request. if((usecase->stream.out) && (usecase->stream.out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) && audio_extn_a2dp_is_force_device_switch()) { ALOGD("Force a2dp device switch to update new encoder config"); ret = true; } if (usecase->stream.out->stream_config_changed) { ALOGD("Force stream_config_changed to update iec61937 transmission config"); return true; } return ret; } bool is_btsco_device(snd_device_t out_snd_device, snd_device_t in_snd_device) { bool ret=false; if ((out_snd_device == SND_DEVICE_OUT_BT_SCO || out_snd_device == SND_DEVICE_OUT_BT_SCO_WB) || in_snd_device == SND_DEVICE_IN_BT_SCO_MIC_WB_NREC || in_snd_device == SND_DEVICE_IN_BT_SCO_MIC_WB || in_snd_device == SND_DEVICE_IN_BT_SCO_MIC_NREC || in_snd_device == SND_DEVICE_IN_BT_SCO_MIC) ret = true; return ret; } bool is_a2dp_device(snd_device_t out_snd_device) { bool ret=false; if (out_snd_device == SND_DEVICE_OUT_BT_A2DP) ret = true; return ret; } bool is_bt_soc_on(struct audio_device *adev) { struct mixer_ctl *ctl; char *mixer_ctl_name = "BT SOC status"; ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name); bool bt_soc_status = true; if (!ctl) { ALOGE("%s: Could not get ctl for mixer cmd - %s", __func__, mixer_ctl_name); /*This is to ensure we dont break targets which dont have the kernel change*/ return true; } bt_soc_status = mixer_ctl_get_value(ctl, 0); ALOGD("BT SOC status: %d",bt_soc_status); return bt_soc_status; } int out_standby_l(struct audio_stream *stream); 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; struct stream_out stream_out; audio_usecase_t hfp_ucid; int status = 0; ALOGD("%s for use case (%s)", __func__, use_case_table[uc_id]); usecase = get_usecase_from_list(adev, uc_id); if (usecase == NULL) { ALOGE("%s: Could not find the usecase(%d)", __func__, uc_id); return -EINVAL; } if ((usecase->type == VOICE_CALL) || (usecase->type == VOIP_CALL) || (usecase->type == PCM_HFP_CALL)) { if(usecase->stream.out == NULL) { ALOGE("%s: stream.out is NULL", __func__); return -EINVAL; } out_snd_device = platform_get_output_snd_device(adev->platform, usecase->stream.out); in_snd_device = platform_get_input_snd_device(adev->platform, usecase->stream.out->devices); usecase->devices = usecase->stream.out->devices; } else if (usecase->type == TRANSCODE_LOOPBACK ) { if (usecase->stream.inout == NULL) { ALOGE("%s: stream.inout is NULL", __func__); return -EINVAL; } stream_out.devices = usecase->stream.inout->out_config.devices; stream_out.sample_rate = usecase->stream.inout->out_config.sample_rate; stream_out.format = usecase->stream.inout->out_config.format; stream_out.channel_mask = usecase->stream.inout->out_config.channel_mask; out_snd_device = platform_get_output_snd_device(adev->platform, &stream_out); in_snd_device = platform_get_input_snd_device(adev->platform, AUDIO_DEVICE_NONE); usecase->devices = (out_snd_device | in_snd_device); } else { /* * If the voice call is active, use the sound devices of voice call usecase * so that it would not result any device switch. All the usecases will * be switched to new device when select_devices() is called for voice call * usecase. This is to avoid switching devices for voice call when * check_usecases_codec_backend() is called below. * choose voice call device only if the use case device is * also using the codec backend */ if (voice_is_in_call(adev) && adev->mode != AUDIO_MODE_NORMAL) { vc_usecase = get_usecase_from_list(adev, get_usecase_id_from_usecase_type(adev, VOICE_CALL)); if ((vc_usecase) && (((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) && (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) || ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) && (usecase->devices & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) || (usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL))) { in_snd_device = vc_usecase->in_snd_device; out_snd_device = vc_usecase->out_snd_device; } } else if (voice_extn_compress_voip_is_active(adev)) { bool out_snd_device_backend_match = true; voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL); if ((voip_usecase != NULL) && (usecase->type == PCM_PLAYBACK) && (usecase->stream.out != NULL)) { out_snd_device_backend_match = platform_check_backends_match( voip_usecase->out_snd_device, platform_get_output_snd_device( adev->platform, usecase->stream.out)); } if ((voip_usecase) && ((voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) && ((usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) || ((usecase->devices & ~AUDIO_DEVICE_BIT_IN) & AUDIO_DEVICE_IN_ALL_CODEC_BACKEND)) && out_snd_device_backend_match && (voip_usecase->stream.out != adev->primary_output))) { in_snd_device = voip_usecase->in_snd_device; out_snd_device = voip_usecase->out_snd_device; } } else if (audio_extn_hfp_is_active(adev)) { hfp_ucid = audio_extn_hfp_get_usecase(); hfp_usecase = get_usecase_from_list(adev, hfp_ucid); if ((hfp_usecase) && (hfp_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) { in_snd_device = hfp_usecase->in_snd_device; out_snd_device = hfp_usecase->out_snd_device; } } if (usecase->type == PCM_PLAYBACK) { if (usecase->stream.out == NULL) { ALOGE("%s: stream.out is NULL", __func__); return -EINVAL; } usecase->devices = usecase->stream.out->devices; in_snd_device = SND_DEVICE_NONE; if (out_snd_device == SND_DEVICE_NONE) { out_snd_device = platform_get_output_snd_device(adev->platform, usecase->stream.out); if (usecase->stream.out == adev->primary_output && adev->active_input && out_snd_device != usecase->out_snd_device) { select_devices(adev, adev->active_input->usecase); } } } else if (usecase->type == PCM_CAPTURE) { if (usecase->stream.in == NULL) { ALOGE("%s: stream.in is NULL", __func__); return -EINVAL; } usecase->devices = usecase->stream.in->device; out_snd_device = SND_DEVICE_NONE; if (in_snd_device == SND_DEVICE_NONE) { audio_devices_t out_device = AUDIO_DEVICE_NONE; if (adev->active_input && (adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION || (adev->mode == AUDIO_MODE_IN_COMMUNICATION && adev->active_input->source == AUDIO_SOURCE_MIC)) && adev->primary_output && !adev->primary_output->standby) { out_device = adev->primary_output->devices; platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE); } else if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) { out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX; } in_snd_device = platform_get_input_snd_device(adev->platform, out_device); } } } if (out_snd_device == usecase->out_snd_device && in_snd_device == usecase->in_snd_device) { if (!force_device_switch(usecase)) return 0; } if ((is_btsco_device(out_snd_device,in_snd_device) && !adev->bt_sco_on) || (is_a2dp_device(out_snd_device) && !audio_extn_a2dp_is_ready())) { ALOGD("SCO/A2DP is selected but they are not connected/ready hence dont route"); return 0; } ALOGD("%s: out_snd_device(%d: %s) in_snd_device(%d: %s)", __func__, out_snd_device, platform_get_snd_device_name(out_snd_device), in_snd_device, platform_get_snd_device_name(in_snd_device)); /* * Limitation: While in call, to do a device switch we need to disable * and enable both RX and TX devices though one of them is same as current * device. */ if ((usecase->type == VOICE_CALL) && (usecase->in_snd_device != SND_DEVICE_NONE) && (usecase->out_snd_device != SND_DEVICE_NONE)) { status = platform_switch_voice_call_device_pre(adev->platform); } if (((usecase->type == VOICE_CALL) || (usecase->type == VOIP_CALL)) && (usecase->out_snd_device != SND_DEVICE_NONE)) { /* Disable sidetone only if voice/voip call already exists */ if (voice_is_call_state_active(adev) || voice_extn_compress_voip_is_started(adev)) voice_set_sidetone(adev, usecase->out_snd_device, false); /* Disable aanc only if voice call exists */ if (voice_is_call_state_active(adev)) voice_check_and_update_aanc_path(adev, usecase->out_snd_device, false); } /* Disable current sound devices */ if (usecase->out_snd_device != SND_DEVICE_NONE) { disable_audio_route(adev, usecase); disable_snd_device(adev, usecase->out_snd_device); } if (usecase->in_snd_device != SND_DEVICE_NONE) { disable_audio_route(adev, usecase); disable_snd_device(adev, usecase->in_snd_device); } /* Applicable only on the targets that has external modem. * New device information should be sent to modem before enabling * the devices to reduce in-call device switch time. */ if ((usecase->type == VOICE_CALL) && (usecase->in_snd_device != SND_DEVICE_NONE) && (usecase->out_snd_device != SND_DEVICE_NONE)) { status = platform_switch_voice_call_enable_device_config(adev->platform, out_snd_device, in_snd_device); } /* Enable new sound devices */ if (out_snd_device != SND_DEVICE_NONE) { check_usecases_codec_backend(adev, usecase, out_snd_device); if (platform_check_codec_asrc_support(adev->platform)) check_and_set_asrc_mode(adev, usecase, out_snd_device); enable_snd_device(adev, out_snd_device); } if (in_snd_device != SND_DEVICE_NONE) { check_usecases_capture_codec_backend(adev, usecase, in_snd_device); enable_snd_device(adev, in_snd_device); } if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) { status = platform_switch_voice_call_device_post(adev->platform, out_snd_device, in_snd_device); enable_audio_route_for_voice_usecases(adev, usecase); } usecase->in_snd_device = in_snd_device; usecase->out_snd_device = out_snd_device; audio_extn_utils_update_stream_app_type_cfg_for_usecase(adev, usecase); if (usecase->type == PCM_PLAYBACK) { if ((24 == usecase->stream.out->bit_width) && (usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) { usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE; } else if ((out_snd_device == SND_DEVICE_OUT_HDMI || out_snd_device == SND_DEVICE_OUT_USB_HEADSET || out_snd_device == SND_DEVICE_OUT_DISPLAY_PORT) && (usecase->stream.out->sample_rate >= OUTPUT_SAMPLING_RATE_44100)) { /* * To best utlize DSP, check if the stream sample rate is supported/multiple of * configured device sample rate, if not update the COPP rate to be equal to the * device sample rate, else open COPP at stream sample rate */ platform_check_and_update_copp_sample_rate(adev->platform, out_snd_device, usecase->stream.out->sample_rate, &usecase->stream.out->app_type_cfg.sample_rate); } else if (((out_snd_device != SND_DEVICE_OUT_HEADPHONES_44_1 && !audio_is_true_native_stream_active(adev)) && usecase->stream.out->sample_rate == OUTPUT_SAMPLING_RATE_44100) || (usecase->stream.out->sample_rate < OUTPUT_SAMPLING_RATE_44100)) { usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE; } /* Notify device change info to effect clients registered */ pthread_mutex_unlock(&adev->lock); audio_extn_gef_notify_device_config( usecase->stream.out->devices, usecase->stream.out->channel_mask, usecase->stream.out->app_type_cfg.sample_rate, platform_get_snd_device_acdb_id(usecase->out_snd_device)); pthread_mutex_lock(&adev->lock); } enable_audio_route(adev, usecase); if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) { /* Enable aanc only if voice call exists */ if (voice_is_call_state_active(adev)) voice_check_and_update_aanc_path(adev, out_snd_device, true); /* Enable sidetone only if other voice/voip call already exists */ if (voice_is_call_state_active(adev) || voice_extn_compress_voip_is_started(adev)) voice_set_sidetone(adev, out_snd_device, true); } /* Applicable only on the targets that has external modem. * Enable device command should be sent to modem only after * enabling voice call mixer controls */ if (usecase->type == VOICE_CALL) status = platform_switch_voice_call_usecase_route_post(adev->platform, out_snd_device, in_snd_device); if (is_btsco_device(out_snd_device, in_snd_device) || is_a2dp_device(out_snd_device)) { if (usecase->type == VOIP_CALL) { if (adev->active_input != NULL && !adev->active_input->standby) { if (is_bt_soc_on(adev) == false){ ALOGD("BT SCO MIC disconnected while in connection"); if (adev->active_input->pcm != NULL) pcm_stop(adev->active_input->pcm); } } if ((usecase->stream.out != NULL) && (usecase->stream.out != adev->primary_output) && usecase->stream.out->started) { if (is_bt_soc_on(adev) == false) { ALOGD("BT SCO/A2DP disconnected while in connection"); out_standby_l(&usecase->stream.out->stream.common); } } } else if ((usecase->stream.out != NULL) && !(usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && usecase->stream.out->started) { if (is_bt_soc_on(adev) == false) { ALOGD("BT SCO/A2dp disconnected while in connection"); out_standby_l(&usecase->stream.out->stream.common); } } } ALOGD("%s: done",__func__); return status; } static int stop_input_stream(struct stream_in *in) { int ret = 0; struct audio_usecase *uc_info; struct audio_device *adev = in->dev; ALOGV("%s: enter: usecase(%d: %s)", __func__, in->usecase, use_case_table[in->usecase]); uc_info = get_usecase_from_list(adev, in->usecase); if (uc_info == NULL) { ALOGE("%s: Could not find the usecase (%d) in the list", __func__, in->usecase); return -EINVAL; } /* Close in-call recording streams */ voice_check_and_stop_incall_rec_usecase(adev, in); /* 1. Disable stream specific mixer controls */ disable_audio_route(adev, uc_info); /* 2. Disable the tx device */ disable_snd_device(adev, uc_info->in_snd_device); list_remove(&uc_info->list); free(uc_info); adev->active_input = get_next_active_input(adev); ALOGV("%s: exit: status(%d)", __func__, ret); return ret; } int start_input_stream(struct stream_in *in) { /* 1. Enable output device and stream routing controls */ int ret = 0; struct audio_usecase *uc_info; struct audio_device *adev = in->dev; int snd_card_status = get_snd_card_state(adev); int usecase = platform_update_usecase_from_source(in->source,in->usecase); if (get_usecase_from_list(adev, usecase) == NULL) in->usecase = usecase; ALOGD("%s: enter: stream(%p)usecase(%d: %s)", __func__, &in->stream, in->usecase, use_case_table[in->usecase]); if (SND_CARD_STATE_OFFLINE == snd_card_status) { ALOGE("%s: sound card is not active/SSR returning error", __func__); ret = -EIO; goto error_config; } if (audio_is_bluetooth_sco_device(in->device)) { if (!adev->bt_sco_on) { ALOGE("%s: SCO profile is not ready, return error", __func__); ret = -EIO; goto error_config; } } /* Check if source matches incall recording usecase criteria */ ret = voice_check_and_set_incall_rec_usecase(adev, in); if (ret) goto error_config; else ALOGV("%s: usecase(%d)", __func__, in->usecase); if (get_usecase_from_list(adev, in->usecase) != NULL) { ALOGE("%s: use case assigned already in use, stream(%p)usecase(%d: %s)", __func__, &in->stream, in->usecase, use_case_table[in->usecase]); return -EINVAL; } in->pcm_device_id = platform_get_pcm_device_id(in->usecase, PCM_CAPTURE); if (in->pcm_device_id < 0) { ALOGE("%s: Could not find PCM device id for the usecase(%d)", __func__, in->usecase); ret = -EINVAL; goto error_config; } adev->active_input = in; uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase)); if (!uc_info) { ret = -ENOMEM; goto error_config; } uc_info->id = in->usecase; uc_info->type = PCM_CAPTURE; uc_info->stream.in = in; uc_info->devices = in->device; uc_info->in_snd_device = SND_DEVICE_NONE; uc_info->out_snd_device = SND_DEVICE_NONE; list_add_tail(&adev->usecase_list, &uc_info->list); audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0, adev->perf_lock_opts, adev->perf_lock_opts_size); select_devices(adev, in->usecase); ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d format %d", __func__, adev->snd_card, in->pcm_device_id, in->config.channels, in->config.format); if (audio_extn_cin_attached_usecase(in->usecase)) { ret = audio_extn_cin_start_input_stream(in); if (ret) goto error_open; else goto done_open; } unsigned int flags = PCM_IN; unsigned int pcm_open_retry_count = 0; if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) { flags |= PCM_MMAP | PCM_NOIRQ; pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT; } else if (in->realtime) { flags |= PCM_MMAP | PCM_NOIRQ; } while (1) { in->pcm = pcm_open(adev->snd_card, in->pcm_device_id, flags, &in->config); if (in->pcm == NULL || !pcm_is_ready(in->pcm)) { ALOGE("%s: %s", __func__, pcm_get_error(in->pcm)); if (in->pcm != NULL) { pcm_close(in->pcm); in->pcm = NULL; } if (pcm_open_retry_count-- == 0) { ret = -EIO; goto error_open; } usleep(PROXY_OPEN_WAIT_TIME * 1000); continue; } break; } ALOGV("%s: pcm_prepare", __func__); ret = pcm_prepare(in->pcm); if (ret < 0) { ALOGE("%s: pcm_prepare returned %d", __func__, ret); pcm_close(in->pcm); in->pcm = NULL; goto error_open; } register_in_stream(in); if (in->realtime) { ret = pcm_start(in->pcm); if (ret < 0) goto error_open; } done_open: audio_extn_perf_lock_release(&adev->perf_lock_handle); ALOGD("%s: exit", __func__); return ret; error_open: audio_extn_perf_lock_release(&adev->perf_lock_handle); stop_input_stream(in); error_config: adev->active_input = get_next_active_input(adev); /* * sleep 50ms to allow sufficient time for kernel * drivers to recover incases like SSR. */ usleep(50000); ALOGD("%s: exit: status(%d)", __func__, ret); return ret; } void lock_input_stream(struct stream_in *in) { pthread_mutex_lock(&in->pre_lock); pthread_mutex_lock(&in->lock); pthread_mutex_unlock(&in->pre_lock); } void lock_output_stream(struct stream_out *out) { pthread_mutex_lock(&out->pre_lock); pthread_mutex_lock(&out->lock); pthread_mutex_unlock(&out->pre_lock); } /* must be called with out->lock locked */ static int send_offload_cmd_l(struct stream_out* out, int command) { struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd)); if (!cmd) { ALOGE("failed to allocate mem for command 0x%x", command); return -ENOMEM; } ALOGVV("%s %d", __func__, command); cmd->cmd = command; list_add_tail(&out->offload_cmd_list, &cmd->node); pthread_cond_signal(&out->offload_cond); return 0; } /* must be called iwth out->lock locked */ static void stop_compressed_output_l(struct stream_out *out) { out->offload_state = OFFLOAD_STATE_IDLE; out->playback_started = 0; out->send_new_metadata = 1; if (out->compr != NULL) { compress_stop(out->compr); while (out->offload_thread_blocked) { pthread_cond_wait(&out->cond, &out->lock); } } } bool is_interactive_usecase(audio_usecase_t uc_id) { unsigned int i; for (i = 0; i < sizeof(interactive_usecases)/sizeof(interactive_usecases[0]); i++) { if (uc_id == interactive_usecases[i]) return true; } return false; } static audio_usecase_t get_interactive_usecase(struct audio_device *adev) { audio_usecase_t ret_uc = USECASE_INVALID; unsigned int intract_uc_index; unsigned int num_usecase = sizeof(interactive_usecases)/sizeof(interactive_usecases[0]); ALOGV("%s: num_usecase: %d", __func__, num_usecase); for (intract_uc_index = 0; intract_uc_index < num_usecase; intract_uc_index++) { if (!(adev->interactive_usecase_state & (0x1 << intract_uc_index))) { adev->interactive_usecase_state |= 0x1 << intract_uc_index; ret_uc = interactive_usecases[intract_uc_index]; break; } } ALOGV("%s: Interactive usecase is %d", __func__, ret_uc); return ret_uc; } static void free_interactive_usecase(struct audio_device *adev, audio_usecase_t uc_id) { unsigned int interact_uc_index; unsigned int num_usecase = sizeof(interactive_usecases)/sizeof(interactive_usecases[0]); for (interact_uc_index = 0; interact_uc_index < num_usecase; interact_uc_index++) { if (interactive_usecases[interact_uc_index] == uc_id) { adev->interactive_usecase_state &= ~(0x1 << interact_uc_index); break; } } ALOGV("%s: free Interactive usecase %d", __func__, uc_id); } bool is_offload_usecase(audio_usecase_t uc_id) { unsigned int i; for (i = 0; i < sizeof(offload_usecases)/sizeof(offload_usecases[0]); i++) { if (uc_id == offload_usecases[i]) return true; } return false; } static audio_usecase_t get_offload_usecase(struct audio_device *adev, bool is_direct_pcm) { audio_usecase_t ret_uc = USECASE_INVALID; unsigned int offload_uc_index; unsigned int num_usecase = sizeof(offload_usecases)/sizeof(offload_usecases[0]); if (!adev->multi_offload_enable) { if (is_direct_pcm) ret_uc = USECASE_AUDIO_PLAYBACK_OFFLOAD2; else ret_uc = USECASE_AUDIO_PLAYBACK_OFFLOAD; pthread_mutex_lock(&adev->lock); if (get_usecase_from_list(adev, ret_uc) != NULL) ret_uc = USECASE_INVALID; pthread_mutex_unlock(&adev->lock); return ret_uc; } ALOGV("%s: num_usecase: %d", __func__, num_usecase); for (offload_uc_index = 0; offload_uc_index < num_usecase; offload_uc_index++) { if (!(adev->offload_usecases_state & (0x1 << offload_uc_index))) { adev->offload_usecases_state |= 0x1 << offload_uc_index; ret_uc = offload_usecases[offload_uc_index]; break; } } ALOGV("%s: offload usecase is %d", __func__, ret_uc); return ret_uc; } static void free_offload_usecase(struct audio_device *adev, audio_usecase_t uc_id) { unsigned int offload_uc_index; unsigned int num_usecase = sizeof(offload_usecases)/sizeof(offload_usecases[0]); if (!adev->multi_offload_enable) return; for (offload_uc_index = 0; offload_uc_index < num_usecase; offload_uc_index++) { if (offload_usecases[offload_uc_index] == uc_id) { adev->offload_usecases_state &= ~(0x1 << offload_uc_index); break; } } ALOGV("%s: free offload usecase %d", __func__, uc_id); } static void *offload_thread_loop(void *context) { struct stream_out *out = (struct stream_out *) context; struct listnode *item; int ret = 0; setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO); set_sched_policy(0, SP_FOREGROUND); prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0); ALOGV("%s", __func__); lock_output_stream(out); for (;;) { struct offload_cmd *cmd = NULL; stream_callback_event_t event; bool send_callback = false; ALOGVV("%s offload_cmd_list %d out->offload_state %d", __func__, list_empty(&out->offload_cmd_list), out->offload_state); if (list_empty(&out->offload_cmd_list)) { ALOGV("%s SLEEPING", __func__); pthread_cond_wait(&out->offload_cond, &out->lock); ALOGV("%s RUNNING", __func__); continue; } item = list_head(&out->offload_cmd_list); cmd = node_to_item(item, struct offload_cmd, node); list_remove(item); ALOGVV("%s STATE %d CMD %d out->compr %p", __func__, out->offload_state, cmd->cmd, out->compr); if (cmd->cmd == OFFLOAD_CMD_EXIT) { free(cmd); break; } if (out->compr == NULL) { ALOGE("%s: Compress handle is NULL", __func__); free(cmd); pthread_cond_signal(&out->cond); continue; } out->offload_thread_blocked = true; pthread_mutex_unlock(&out->lock); send_callback = false; switch(cmd->cmd) { case OFFLOAD_CMD_WAIT_FOR_BUFFER: ALOGD("copl(%p):calling compress_wait", out); compress_wait(out->compr, -1); ALOGD("copl(%p):out of compress_wait", out); send_callback = true; event = STREAM_CBK_EVENT_WRITE_READY; break; case OFFLOAD_CMD_PARTIAL_DRAIN: ret = compress_next_track(out->compr); if(ret == 0) { ALOGD("copl(%p):calling compress_partial_drain", out); ret = compress_partial_drain(out->compr); ALOGD("copl(%p):out of compress_partial_drain", out); if (ret < 0) ret = -errno; } else if (ret == -ETIMEDOUT) compress_drain(out->compr); else ALOGE("%s: Next track returned error %d",__func__, ret); if (ret != -ENETRESET) { send_callback = true; pthread_mutex_lock(&out->lock); out->send_new_metadata = 1; out->send_next_track_params = true; pthread_mutex_unlock(&out->lock); event = STREAM_CBK_EVENT_DRAIN_READY; ALOGV("copl(%p):send drain callback, ret %d", out, ret); } else ALOGE("%s: Block drain ready event during SSR", __func__); break; case OFFLOAD_CMD_DRAIN: ALOGD("copl(%p):calling compress_drain", out); compress_drain(out->compr); ALOGD("copl(%p):calling compress_drain", out); send_callback = true; event = STREAM_CBK_EVENT_DRAIN_READY; break; default: ALOGE("%s unknown command received: %d", __func__, cmd->cmd); break; } lock_output_stream(out); out->offload_thread_blocked = false; pthread_cond_signal(&out->cond); if (send_callback && out->client_callback) { ALOGVV("%s: sending client_callback event %d", __func__, event); out->client_callback(event, NULL, out->client_cookie); } free(cmd); } pthread_cond_signal(&out->cond); while (!list_empty(&out->offload_cmd_list)) { item = list_head(&out->offload_cmd_list); list_remove(item); free(node_to_item(item, struct offload_cmd, node)); } pthread_mutex_unlock(&out->lock); return NULL; } static int create_offload_callback_thread(struct stream_out *out) { pthread_cond_init(&out->offload_cond, (const pthread_condattr_t *) NULL); list_init(&out->offload_cmd_list); pthread_create(&out->offload_thread, (const pthread_attr_t *) NULL, offload_thread_loop, out); return 0; } static int destroy_offload_callback_thread(struct stream_out *out) { lock_output_stream(out); stop_compressed_output_l(out); send_offload_cmd_l(out, OFFLOAD_CMD_EXIT); pthread_mutex_unlock(&out->lock); pthread_join(out->offload_thread, (void **) NULL); pthread_cond_destroy(&out->offload_cond); return 0; } static int stop_output_stream(struct stream_out *out) { int ret = 0; struct audio_usecase *uc_info; struct audio_device *adev = out->dev; ALOGV("%s: enter: usecase(%d: %s)", __func__, out->usecase, use_case_table[out->usecase]); uc_info = get_usecase_from_list(adev, out->usecase); if (uc_info == NULL) { ALOGE("%s: Could not find the usecase (%d) in the list", __func__, out->usecase); return -EINVAL; } if (is_offload_usecase(out->usecase) && !(audio_extn_passthru_is_passthrough_stream(out))) { if (adev->visualizer_stop_output != NULL) adev->visualizer_stop_output(out->handle, out->pcm_device_id); audio_extn_dts_remove_state_notifier_node(out->usecase); if (adev->offload_effects_stop_output != NULL) adev->offload_effects_stop_output(out->handle, out->pcm_device_id); } /* 1. Get and set stream specific mixer controls */ disable_audio_route(adev, uc_info); /* 2. Disable the rx device */ disable_snd_device(adev, uc_info->out_snd_device); list_remove(&uc_info->list); free(uc_info); out->started = 0; if (is_offload_usecase(out->usecase) && (audio_extn_passthru_is_passthrough_stream(out))) { ALOGV("Disable passthrough , reset mixer to pcm"); /* NO_PASSTHROUGH */ out->compr_config.codec->compr_passthr = 0; audio_extn_passthru_on_stop(out); audio_extn_dolby_set_dap_bypass(adev, DAP_STATE_ON); } /* Must be called after removing the usecase from list */ if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) audio_extn_keep_alive_start(); if (out->ip_hdlr_handle) { ret = audio_extn_ip_hdlr_intf_close(out->ip_hdlr_handle, true, out); if (ret < 0) ALOGE("%s: audio_extn_ip_hdlr_intf_close failed %d",__func__, ret); } ALOGV("%s: exit: status(%d)", __func__, ret); return ret; } int start_output_stream(struct stream_out *out) { int ret = 0; struct audio_usecase *uc_info; struct audio_device *adev = out->dev; int snd_card_status = get_snd_card_state(adev); char mixer_ctl_name[128]; struct mixer_ctl *ctl = NULL; char* perf_mode[] = {"ULL", "ULL_PP", "LL"}; if ((out->usecase < 0) || (out->usecase >= AUDIO_USECASE_MAX)) { ret = -EINVAL; goto error_config; } ALOGD("%s: enter: stream(%p)usecase(%d: %s) devices(%#x)", __func__, &out->stream, out->usecase, use_case_table[out->usecase], out->devices); if (SND_CARD_STATE_OFFLINE == snd_card_status) { ALOGE("%s: sound card is not active/SSR returning error", __func__); ret = -EIO; goto error_config; } if (out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) { if (!audio_extn_a2dp_is_ready()) { if (out->devices & AUDIO_DEVICE_OUT_SPEAKER) { //combo usecase just by pass a2dp ALOGW("%s: A2DP profile is not ready, route it to speaker", __func__); out->devices = AUDIO_DEVICE_OUT_SPEAKER; } else { ALOGE("%s: A2DP profile is not ready, return error", __func__); ret = -EAGAIN; goto error_config; } } } if (out->devices & AUDIO_DEVICE_OUT_ALL_SCO) { if (!adev->bt_sco_on) { if (out->devices & AUDIO_DEVICE_OUT_SPEAKER) { //combo usecase just by pass a2dp ALOGW("%s: SCO is not connected, route it to speaker", __func__); out->devices = AUDIO_DEVICE_OUT_SPEAKER; } else { ALOGE("%s: SCO profile is not ready, return error", __func__); ret = -EAGAIN; goto error_config; } } } out->pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK); if (out->pcm_device_id < 0) { ALOGE("%s: Invalid PCM device id(%d) for the usecase(%d)", __func__, out->pcm_device_id, out->usecase); ret = -EINVAL; goto error_open; } uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase)); if (!uc_info) { ret = -ENOMEM; goto error_config; } uc_info->id = out->usecase; uc_info->type = PCM_PLAYBACK; uc_info->stream.out = out; uc_info->devices = out->devices; uc_info->in_snd_device = SND_DEVICE_NONE; uc_info->out_snd_device = SND_DEVICE_NONE; list_add_tail(&adev->usecase_list, &uc_info->list); audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0, adev->perf_lock_opts, adev->perf_lock_opts_size); if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { audio_extn_keep_alive_stop(); if (audio_extn_passthru_is_enabled() && audio_extn_passthru_is_passthrough_stream(out)) { audio_extn_passthru_on_start(out); } } select_devices(adev, out->usecase); ALOGV("%s: Opening PCM device card_id(%d) device_id(%d) format(%#x)", __func__, adev->snd_card, out->pcm_device_id, out->config.format); if (!is_offload_usecase(out->usecase)) { unsigned int flags = PCM_OUT; unsigned int pcm_open_retry_count = 0; if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) { flags |= PCM_MMAP | PCM_NOIRQ; pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT; } else if (out->realtime) { flags |= PCM_MMAP | PCM_NOIRQ; } else flags |= PCM_MONOTONIC; if ((adev->vr_audio_mode_enabled) && (out->flags & AUDIO_OUTPUT_FLAG_RAW)) { snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "PCM_Dev %d Topology", out->pcm_device_id); ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name); if (!ctl) { ALOGI("%s: Could not get ctl for mixer cmd might be ULL - %s", __func__, mixer_ctl_name); } else { //if success use ULLPP ALOGI("%s: mixer ctrl %s succeeded setting up ULL for %d", __func__, mixer_ctl_name, out->pcm_device_id); //There is a still a possibility that some sessions // that request for FAST|RAW when 3D audio is active //can go through ULLPP. Ideally we expects apps to //listen to audio focus and stop concurrent playback //Also, we will look for mode flag (voice_in_communication) //before enabling the realtime flag. mixer_ctl_set_enum_by_string(ctl, perf_mode[1]); } } while (1) { out->pcm = pcm_open(adev->snd_card, out->pcm_device_id, flags, &out->config); if (out->pcm == NULL || !pcm_is_ready(out->pcm)) { ALOGE("%s: %s", __func__, pcm_get_error(out->pcm)); if (out->pcm != NULL) { pcm_close(out->pcm); out->pcm = NULL; } if (pcm_open_retry_count-- == 0) { ret = -EIO; goto error_open; } usleep(PROXY_OPEN_WAIT_TIME * 1000); continue; } break; } platform_set_stream_channel_map(adev->platform, out->channel_mask, out->pcm_device_id, &out->channel_map_param.channel_map[0]); 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 { platform_set_stream_channel_map(adev->platform, out->channel_mask, out->pcm_device_id, &out->channel_map_param.channel_map[0]); out->pcm = NULL; out->compr = compress_open(adev->snd_card, out->pcm_device_id, COMPRESS_IN, &out->compr_config); if (out->compr && !is_compress_ready(out->compr)) { ALOGE("%s: %s", __func__, compress_get_error(out->compr)); compress_close(out->compr); out->compr = NULL; ret = -EIO; goto error_open; } /* compress_open sends params of the track, so reset the flag here */ out->is_compr_metadata_avail = false; if (out->client_callback) compress_nonblock(out->compr, out->non_blocking); /* Since small bufs uses blocking writes, a write will be blocked for the default max poll time (20s) in the event of an SSR. Reduce the poll time to observe and deal with SSR faster. */ if (!out->non_blocking) { compress_set_max_poll_wait(out->compr, 1000); } audio_extn_utils_compress_set_render_mode(out); audio_extn_utils_compress_set_clk_rec_mode(uc_info); audio_extn_dts_create_state_notifier_node(out->usecase); audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate, popcount(out->channel_mask), out->playback_started); #ifdef DS1_DOLBY_DDP_ENABLED if (audio_extn_utils_is_dolby_format(out->format)) audio_extn_dolby_send_ddp_endp_params(adev); #endif if (!(audio_extn_passthru_is_passthrough_stream(out)) && (out->sample_rate != 176400 && out->sample_rate <= 192000)) { if (adev->visualizer_start_output != NULL) adev->visualizer_start_output(out->handle, out->pcm_device_id); if (adev->offload_effects_start_output != NULL) adev->offload_effects_start_output(out->handle, out->pcm_device_id, adev->mixer); audio_extn_check_and_set_dts_hpx_state(adev); } } if (ret == 0) { register_out_stream(out); if (out->realtime) { ret = pcm_start(out->pcm); if (ret < 0) goto error_open; } } audio_extn_perf_lock_release(&adev->perf_lock_handle); ALOGD("%s: exit", __func__); if (out->ip_hdlr_handle) { ret = audio_extn_ip_hdlr_intf_open(out->ip_hdlr_handle, true, out, out->usecase); if (ret < 0) ALOGE("%s: audio_extn_ip_hdlr_intf_open failed %d",__func__, ret); } return ret; error_open: audio_extn_perf_lock_release(&adev->perf_lock_handle); stop_output_stream(out); error_config: /* * sleep 50ms to allow sufficient time for kernel * drivers to recover incases like SSR. */ usleep(50000); return ret; } static int check_input_parameters(uint32_t sample_rate, audio_format_t format, int channel_count) { int ret = 0; if (((format != AUDIO_FORMAT_PCM_16_BIT) && (format != AUDIO_FORMAT_PCM_8_24_BIT) && (format != AUDIO_FORMAT_PCM_24_BIT_PACKED) && (format != AUDIO_FORMAT_PCM_32_BIT) && (format != AUDIO_FORMAT_PCM_FLOAT)) && !voice_extn_compress_voip_is_format_supported(format) && !audio_extn_compr_cap_format_supported(format)) ret = -EINVAL; switch (channel_count) { case 1: case 2: case 3: case 4: case 6: break; default: ret = -EINVAL; } switch (sample_rate) { case 8000: case 11025: case 12000: case 16000: case 22050: case 24000: case 32000: case 44100: case 48000: case 96000: case 192000: break; default: ret = -EINVAL; } return ret; } static size_t get_input_buffer_size(uint32_t sample_rate, audio_format_t format, int channel_count, bool is_low_latency) { size_t size = 0; if (check_input_parameters(sample_rate, format, channel_count) != 0) return 0; size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000; if (is_low_latency) size = configured_low_latency_capture_period_size; size *= audio_bytes_per_sample(format) * channel_count; /* make sure the size is multiple of 32 bytes * At 48 kHz mono 16-bit PCM: * 5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15) * 3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10) */ size += 0x1f; size &= ~0x1f; return size; } static size_t get_output_period_size(uint32_t sample_rate, audio_format_t format, int channel_count, int duration /*in millisecs*/) { size_t size = 0; uint32_t bytes_per_sample = audio_bytes_per_sample(format); if ((duration == 0) || (sample_rate == 0) || (bytes_per_sample == 0) || (channel_count == 0)) { ALOGW("Invalid config duration %d sr %d bps %d ch %d", duration, sample_rate, bytes_per_sample, channel_count); return -EINVAL; } size = (sample_rate * duration * bytes_per_sample * channel_count) / 1000; /* * To have same PCM samples for all channels, the buffer size requires to * be multiple of (number of channels * bytes per sample) * For writes to succeed, the buffer must be written at address which is multiple of 32 */ size = ALIGN(size, (bytes_per_sample * channel_count * 32)); return (size/(channel_count * bytes_per_sample)); } static uint64_t get_actual_pcm_frames_rendered(struct stream_out *out) { uint64_t actual_frames_rendered = 0; size_t kernel_buffer_size = out->compr_config.fragment_size * out->compr_config.fragments; /* This adjustment accounts for buffering after app processor. * It is based on estimated DSP latency per use case, rather than exact. */ int64_t platform_latency = platform_render_latency(out->usecase) * out->sample_rate / 1000000LL; /* not querying actual state of buffering in kernel as it would involve an ioctl call * which then needs protection, this causes delay in TS query for pcm_offload usecase * hence only estimate. */ int64_t signed_frames = out->written - kernel_buffer_size; signed_frames = signed_frames / (audio_bytes_per_sample(out->format) * popcount(out->channel_mask)) - platform_latency; if (signed_frames > 0) actual_frames_rendered = signed_frames; ALOGVV("%s signed frames %lld out_written %lld kernel_buffer_size %d" "bytes/sample %zu channel count %d", __func__,(long long int)signed_frames, (long long int)out->written, (int)kernel_buffer_size, audio_bytes_per_sample(out->compr_config.codec->format), popcount(out->channel_mask)); return actual_frames_rendered; } static uint32_t out_get_sample_rate(const struct audio_stream *stream) { struct stream_out *out = (struct stream_out *)stream; return out->sample_rate; } static int out_set_sample_rate(struct audio_stream *stream __unused, uint32_t rate __unused) { return -ENOSYS; } static size_t out_get_buffer_size(const struct audio_stream *stream) { struct stream_out *out = (struct stream_out *)stream; if (is_interactive_usecase(out->usecase)) { return out->config.period_size * out->config.period_count; } else if (is_offload_usecase(out->usecase)) { if (out->flags & AUDIO_OUTPUT_FLAG_TIMESTAMP) return out->compr_config.fragment_size - sizeof(struct snd_codec_metadata); else return out->compr_config.fragment_size; } else if(out->usecase == USECASE_COMPRESS_VOIP_CALL) return voice_extn_compress_voip_out_get_buffer_size(out); else if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) return out->hal_fragment_size; return out->config.period_size * out->af_period_multiplier * audio_stream_out_frame_size((const struct audio_stream_out *)stream); } static uint32_t out_get_channels(const struct audio_stream *stream) { struct stream_out *out = (struct stream_out *)stream; return out->channel_mask; } static audio_format_t out_get_format(const struct audio_stream *stream) { struct stream_out *out = (struct stream_out *)stream; return out->format; } static int out_set_format(struct audio_stream *stream __unused, audio_format_t format __unused) { return -ENOSYS; } static int out_standby(struct audio_stream *stream) { struct stream_out *out = (struct stream_out *)stream; struct audio_device *adev = out->dev; ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__, stream, out->usecase, use_case_table[out->usecase]); lock_output_stream(out); if (!out->standby) { if (adev->adm_deregister_stream) adev->adm_deregister_stream(adev->adm_data, out->handle); if (is_offload_usecase(out->usecase)) stop_compressed_output_l(out); pthread_mutex_lock(&adev->lock); out->standby = true; if (out->usecase == USECASE_COMPRESS_VOIP_CALL) { voice_extn_compress_voip_close_output_stream(stream); out->started = 0; pthread_mutex_unlock(&adev->lock); pthread_mutex_unlock(&out->lock); ALOGD("VOIP output entered standby"); return 0; } else if (!is_offload_usecase(out->usecase)) { if (out->pcm) { pcm_close(out->pcm); out->pcm = NULL; } } else { ALOGD("copl(%p):standby", out); out->send_next_track_params = false; out->is_compr_metadata_avail = false; out->gapless_mdata.encoder_delay = 0; out->gapless_mdata.encoder_padding = 0; if (out->compr != NULL) { compress_close(out->compr); out->compr = NULL; } } stop_output_stream(out); pthread_mutex_unlock(&adev->lock); } pthread_mutex_unlock(&out->lock); ALOGD("%s: exit", __func__); return 0; } /* *standby implementation without locks, assumes that the callee already *has taken adev and out lock. */ int out_standby_l(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->standby) { if (adev->adm_deregister_stream) adev->adm_deregister_stream(adev->adm_data, out->handle); if (is_offload_usecase(out->usecase)) stop_compressed_output_l(out); out->standby = true; if (out->usecase == USECASE_COMPRESS_VOIP_CALL) { voice_extn_compress_voip_close_output_stream(stream); out->started = 0; ALOGD("VOIP output entered standby"); return 0; } else if (!is_offload_usecase(out->usecase)) { if (out->pcm) { pcm_close(out->pcm); out->pcm = NULL; } } else { ALOGD("copl(%p):standby", out); out->send_next_track_params = false; out->is_compr_metadata_avail = false; out->gapless_mdata.encoder_delay = 0; out->gapless_mdata.encoder_padding = 0; if (out->compr != NULL) { compress_close(out->compr); out->compr = NULL; } } stop_output_stream(out); } ALOGD("%s: exit", __func__); return 0; } static int out_dump(const struct audio_stream *stream __unused, int fd __unused) { return 0; } static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms) { int ret = 0; char value[32]; if (!out || !parms) { ALOGE("%s: return invalid ",__func__); return -EINVAL; } ret = audio_extn_parse_compress_metadata(out, parms); ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value)); if (ret >= 0) { out->gapless_mdata.encoder_delay = atoi(value); //whats a good limit check? } ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value)); if (ret >= 0) { out->gapless_mdata.encoder_padding = atoi(value); } ALOGV("%s new encoder delay %u and padding %u", __func__, out->gapless_mdata.encoder_delay, out->gapless_mdata.encoder_padding); return 0; } static bool output_drives_call(struct audio_device *adev, struct stream_out *out) { return out == adev->primary_output || out == adev->voice_tx_output; } static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) { struct stream_out *out = (struct stream_out *)stream; struct audio_device *adev = out->dev; struct str_parms *parms; char value[32]; int ret = 0, val = 0, err; ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s", __func__, out->usecase, use_case_table[out->usecase], kvpairs); parms = str_parms_create_str(kvpairs); if (!parms) goto error; err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); if (err >= 0) { val = atoi(value); lock_output_stream(out); pthread_mutex_lock(&adev->lock); /* * When HDMI cable is unplugged the music playback is paused and * the policy manager sends routing=0. But the audioflinger continues * to write data until standby time (3sec). As the HDMI core is * turned off, the write gets blocked. * Avoid this by routing audio to speaker until standby. */ if ((out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL) && (val == AUDIO_DEVICE_NONE) && !audio_extn_passthru_is_passthrough_stream(out) && (platform_get_edid_info(adev->platform) != 0) /* HDMI disconnected */) { val = AUDIO_DEVICE_OUT_SPEAKER; } /* * When A2DP is disconnected the * music playback is paused and the policy manager sends routing=0 * But the audioflingercontinues to write data until standby time * (3sec). As BT is turned off, the write gets blocked. * Avoid this by routing audio to speaker until standby. */ if ((out->devices & AUDIO_DEVICE_OUT_ALL_A2DP) && (val == AUDIO_DEVICE_NONE)) { val = AUDIO_DEVICE_OUT_SPEAKER; } /* To avoid a2dp to sco overlapping / BT device improper state * check with BT lib about a2dp streaming support before routing */ if (val & AUDIO_DEVICE_OUT_ALL_A2DP) { if (!audio_extn_a2dp_is_ready()) { if (val & AUDIO_DEVICE_OUT_SPEAKER) { //combo usecase just by pass a2dp ALOGW("%s: A2DP profile is not ready,routing to speaker only", __func__); val = AUDIO_DEVICE_OUT_SPEAKER; } else { ALOGE("%s: A2DP profile is not ready,ignoring routing request", __func__); /* update device to a2dp and don't route as BT returned error * However it is still possible a2dp routing called because * of current active device disconnection (like wired headset) */ out->devices = val; pthread_mutex_unlock(&out->lock); pthread_mutex_unlock(&adev->lock); goto error; } } } /* * select_devices() call below switches all the usecases on the same * backend to the new device. Refer to check_usecases_codec_backend() in * the select_devices(). But how do we undo this? * * For example, music playback is active on headset (deep-buffer usecase) * and if we go to ringtones and select a ringtone, low-latency usecase * will be started on headset+speaker. As we can't enable headset+speaker * and headset devices at the same time, select_devices() switches the music * playback to headset+speaker while starting low-lateny usecase for ringtone. * So when the ringtone playback is completed, how do we undo the same? * * We are relying on the out_set_parameters() call on deep-buffer output, * once the ringtone playback is ended. * NOTE: We should not check if the current devices are same as new devices. * Because select_devices() must be called to switch back the music * playback to headset. */ if (val != 0) { audio_devices_t new_dev = val; bool same_dev = out->devices == new_dev; out->devices = new_dev; if (output_drives_call(adev, out)) { if(!voice_is_in_call(adev)) { if (adev->mode == AUDIO_MODE_IN_CALL) { adev->current_call_output = out; ret = voice_start_call(adev); } } else { adev->current_call_output = out; voice_update_devices_for_all_voice_usecases(adev); } } if (!out->standby) { if (!same_dev) { ALOGV("update routing change"); audio_extn_perf_lock_acquire(&adev->perf_lock_handle, 0, adev->perf_lock_opts, adev->perf_lock_opts_size); if (adev->adm_on_routing_change) adev->adm_on_routing_change(adev->adm_data, out->handle); } select_devices(adev, out->usecase); if (!same_dev) audio_extn_perf_lock_release(&adev->perf_lock_handle); } } pthread_mutex_unlock(&adev->lock); pthread_mutex_unlock(&out->lock); } if (out == adev->primary_output) { pthread_mutex_lock(&adev->lock); audio_extn_set_parameters(adev, parms); pthread_mutex_unlock(&adev->lock); } if (is_offload_usecase(out->usecase)) { lock_output_stream(out); parse_compress_metadata(out, parms); audio_extn_dts_create_state_notifier_node(out->usecase); audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate, popcount(out->channel_mask), out->playback_started); pthread_mutex_unlock(&out->lock); } err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_PROFILE, value, sizeof(value)); if (err >= 0) { strlcpy(out->profile, value, sizeof(out->profile)); ALOGV("updating stream profile with value '%s'", out->profile); lock_output_stream(out); audio_extn_utils_update_stream_output_app_type_cfg(adev->platform, &adev->streams_output_cfg_list, out->devices, out->flags, out->format, out->sample_rate, out->bit_width, out->channel_mask, out->profile, &out->app_type_cfg); pthread_mutex_unlock(&out->lock); } str_parms_destroy(parms); error: ALOGV("%s: exit: code(%d)", __func__, ret); return ret; } static char* out_get_parameters(const struct audio_stream *stream, const char *keys) { struct stream_out *out = (struct stream_out *)stream; struct str_parms *query = str_parms_create_str(keys); char *str = (char*) NULL; char value[256]; struct str_parms *reply = str_parms_create(); size_t i, j; int ret; bool first = true; if (!query || !reply) { if (reply) { str_parms_destroy(reply); } if (query) { str_parms_destroy(query); } ALOGE("out_get_parameters: failed to allocate mem for query or reply"); return NULL; } ALOGV("%s: enter: keys - %s", __func__, keys); ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value)); if (ret >= 0) { value[0] = '\0'; i = 0; while (out->supported_channel_masks[i] != 0) { for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) { if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) { if (!first) { strlcat(value, "|", sizeof(value)); } strlcat(value, out_channels_name_to_enum_table[j].name, sizeof(value)); first = false; break; } } i++; } str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value); str = str_parms_to_str(reply); } else { voice_extn_out_get_parameters(out, query, reply); str = str_parms_to_str(reply); if (str && !strncmp(str, "", sizeof(""))) { free(str); str = strdup(keys); } } ret = str_parms_get_str(query, "is_direct_pcm_track", value, sizeof(value)); if (ret >= 0) { value[0] = '\0'; if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) { ALOGV("in direct_pcm"); strlcat(value, "true", sizeof(value )); } else { ALOGV("not in direct_pcm"); strlcat(value, "false", sizeof(value)); } str_parms_add_str(reply, "is_direct_pcm_track", value); if (str) free(str); str = str_parms_to_str(reply); } ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value, sizeof(value)); if (ret >= 0) { value[0] = '\0'; i = 0; first = true; while (out->supported_formats[i] != 0) { for (j = 0; j < ARRAY_SIZE(out_formats_name_to_enum_table); j++) { if (out_formats_name_to_enum_table[j].value == out->supported_formats[i]) { if (!first) { strlcat(value, "|", sizeof(value)); } strlcat(value, out_formats_name_to_enum_table[j].name, sizeof(value)); first = false; break; } } i++; } str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_FORMATS, value); if (str) free(str); str = str_parms_to_str(reply); } ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES, value, sizeof(value)); if (ret >= 0) { value[0] = '\0'; i = 0; first = true; while (out->supported_sample_rates[i] != 0) { for (j = 0; j < ARRAY_SIZE(out_hdmi_sample_rates_name_to_enum_table); j++) { if (out_hdmi_sample_rates_name_to_enum_table[j].value == out->supported_sample_rates[i]) { if (!first) { strlcat(value, "|", sizeof(value)); } strlcat(value, out_hdmi_sample_rates_name_to_enum_table[j].name, sizeof(value)); first = false; break; } } i++; } str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES, value); if (str) free(str); str = str_parms_to_str(reply); } str_parms_destroy(query); str_parms_destroy(reply); ALOGV("%s: exit: returns - %s", __func__, str); return str; } static uint32_t out_get_latency(const struct audio_stream_out *stream) { uint32_t period_ms; struct stream_out *out = (struct stream_out *)stream; uint32_t latency = 0; if (is_offload_usecase(out->usecase)) { lock_output_stream(out); latency = audio_extn_utils_compress_get_dsp_latency(out); pthread_mutex_unlock(&out->lock); } else if (out->realtime) { // since the buffer won't be filled up faster than realtime, // return a smaller number if (out->config.rate) period_ms = (out->af_period_multiplier * out->config.period_size * 1000) / (out->config.rate); else period_ms = 0; latency = period_ms + platform_render_latency(out->usecase)/1000; } else { latency = (out->config.period_count * out->config.period_size * 1000) / (out->config.rate); } if ((AUDIO_DEVICE_OUT_BLUETOOTH_A2DP == out->devices) && !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) latency += audio_extn_a2dp_get_encoder_latency(); ALOGV("%s: Latency %d", __func__, latency); return latency; } static float AmpToDb(float amplification) { float db = DSD_VOLUME_MIN_DB; if (amplification > 0) { db = 20 * log10(amplification); if(db < DSD_VOLUME_MIN_DB) return DSD_VOLUME_MIN_DB; } return db; } static int out_set_volume(struct audio_stream_out *stream, float left, float right) { struct stream_out *out = (struct stream_out *)stream; int volume[2]; if (out->usecase == USECASE_AUDIO_PLAYBACK_MULTI_CH) { /* only take left channel into account: the API is for stereo anyway */ out->muted = (left == 0.0f); return 0; } else if (is_offload_usecase(out->usecase)) { if (audio_extn_passthru_is_passthrough_stream(out)) { /* * Set mute or umute on HDMI passthrough stream. * Only take left channel into account. * Mute is 0 and unmute 1 */ audio_extn_passthru_set_volume(out, (left == 0.0f)); } else if (out->format == AUDIO_FORMAT_DSD){ char mixer_ctl_name[128] = "DSD Volume"; struct audio_device *adev = out->dev; struct mixer_ctl *ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name); if (!ctl) { ALOGE("%s: Could not get ctl for mixer cmd - %s", __func__, mixer_ctl_name); return -EINVAL; } volume[0] = (int)(AmpToDb(left)); volume[1] = (int)(AmpToDb(right)); mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0])); return 0; } else { char mixer_ctl_name[128]; struct audio_device *adev = out->dev; struct mixer_ctl *ctl; int pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK); snprintf(mixer_ctl_name, sizeof(mixer_ctl_name), "Compress Playback %d Volume", pcm_device_id); ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name); if (!ctl) { ALOGE("%s: Could not get ctl for mixer cmd - %s", __func__, mixer_ctl_name); return -EINVAL; } volume[0] = (int)(left * COMPRESS_PLAYBACK_VOLUME_MAX); volume[1] = (int)(right * COMPRESS_PLAYBACK_VOLUME_MAX); mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0])); return 0; } } return -ENOSYS; } static ssize_t out_write(struct audio_stream_out *stream, const void *buffer, size_t bytes) { struct stream_out *out = (struct stream_out *)stream; struct audio_device *adev = out->dev; int snd_scard_state = get_snd_card_state(adev); ssize_t ret = 0; lock_output_stream(out); if (SND_CARD_STATE_OFFLINE == snd_scard_state) { if ((!(out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) && is_offload_usecase(out->usecase)) { /*during SSR for compress usecase we should return error to flinger*/ ALOGD(" copl %s: sound card is not active/SSR state", __func__); pthread_mutex_unlock(&out->lock); return -ENETRESET; } else { /* increase written size during SSR to avoid mismatch * with the written frames count in AF */ // bytes per frame size_t bpf = audio_bytes_per_sample(out->format) * audio_channel_count_from_out_mask(out->channel_mask); if (bpf != 0) out->written += bytes / bpf; ALOGD(" %s: sound card is not active/SSR state", __func__); ret= -EIO; goto exit; } } if (audio_extn_passthru_should_drop_data(out)) { ALOGV(" %s : Drop data as compress passthrough session is going on", __func__); if ((audio_bytes_per_sample(out->format) != 0) && (out->config.channels != 0)) out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format)); ret = -EIO; goto exit; } if ((out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) && !out->is_iec61937_info_available) { if (!audio_extn_passthru_is_passthrough_stream(out)) { out->is_iec61937_info_available = true; } else if (audio_extn_passthru_is_enabled()) { audio_extn_passthru_update_stream_configuration(adev, out, buffer, bytes); out->is_iec61937_info_available = true; if((out->format == AUDIO_FORMAT_DTS) || (out->format == AUDIO_FORMAT_DTS_HD)) { ret = audio_extn_passthru_update_dts_stream_configuration(out, buffer, bytes); if (ret) { if (ret != -ENOSYS) { out->is_iec61937_info_available = false; ALOGD("iec61937 transmission info not yet updated retry"); } } else if (!out->standby) { /* if stream has started and after that there is * stream config change (iec transmission config) * then trigger select_device to update backend configuration. */ out->stream_config_changed = true; pthread_mutex_lock(&adev->lock); select_devices(adev, out->usecase); if (!audio_extn_passthru_is_supported_backend_edid_cfg(adev, out)) { ret = -EINVAL; goto exit; } pthread_mutex_unlock(&adev->lock); out->stream_config_changed = false; out->is_iec61937_info_available = true; } } } } 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; } out->started = 1; if (last_known_cal_step != -1) { ALOGD("%s: retry previous failed cal level set", __func__); audio_hw_send_gain_dep_calibration(last_known_cal_step); last_known_cal_step = -1; } if ((out->is_iec61937_info_available == true) && (audio_extn_passthru_is_passthrough_stream(out))&& (!audio_extn_passthru_is_supported_backend_edid_cfg(adev, out))) { ret = -EINVAL; goto exit; } } if (adev->is_channel_status_set == false && (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)){ audio_utils_set_hdmi_channel_status(out, (void *)buffer, bytes); adev->is_channel_status_set = true; } if (is_offload_usecase(out->usecase)) { ALOGVV("copl(%p): writing buffer (%zu bytes) to compress device", out, bytes); if (out->send_new_metadata) { ALOGD("copl(%p):send new gapless metadata", out); compress_set_gapless_metadata(out->compr, &out->gapless_mdata); out->send_new_metadata = 0; if (out->send_next_track_params && out->is_compr_metadata_avail) { ALOGD("copl(%p):send next track params in gapless", out); compress_set_next_track_param(out->compr, &(out->compr_config.codec->options)); out->send_next_track_params = false; out->is_compr_metadata_avail = false; } } if ((out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) && (out->convert_buffer) != NULL) { if ((bytes > out->hal_fragment_size)) { ALOGW("Error written bytes %zu > %d (fragment_size)", bytes, out->hal_fragment_size); pthread_mutex_unlock(&out->lock); return -EINVAL; } else { audio_format_t dst_format = out->hal_op_format; audio_format_t src_format = out->hal_ip_format; uint32_t frames = bytes / format_to_bitwidth_table[src_format]; uint32_t bytes_to_write = frames * format_to_bitwidth_table[dst_format]; memcpy_by_audio_format(out->convert_buffer, dst_format, buffer, src_format, frames); ret = compress_write(out->compr, out->convert_buffer, bytes_to_write); /*Convert written bytes in audio flinger format*/ if (ret > 0) ret = ((ret * format_to_bitwidth_table[out->format]) / format_to_bitwidth_table[dst_format]); } } else ret = compress_write(out->compr, buffer, bytes); if (ret < 0) ret = -errno; ALOGVV("%s: writing buffer (%zu bytes) to compress device returned %zd", __func__, bytes, ret); /*msg to cb thread only if non blocking write is enabled*/ if (ret >= 0 && ret < (ssize_t)bytes && out->non_blocking) { ALOGD("No space available in compress driver, post msg to cb thread"); send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER); } else if (-ENETRESET == ret) { ALOGE("copl %s: received sound card offline state on compress write", __func__); set_snd_card_state(adev,SND_CARD_STATE_OFFLINE); pthread_mutex_unlock(&out->lock); out_standby(&out->stream.common); return ret; } if ( ret == (ssize_t)bytes && !out->non_blocking) out->written += bytes; /* Call compr start only when non-zero bytes of data is there to be rendered */ if (!out->playback_started && ret > 0) { int status = compress_start(out->compr); if (status < 0) { ret = status; ALOGE("%s: compr start failed with err %d", __func__, errno); goto exit; } audio_extn_dts_eagle_fade(adev, true, out); out->playback_started = 1; out->offload_state = OFFLOAD_STATE_PLAYING; audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate, popcount(out->channel_mask), out->playback_started); } pthread_mutex_unlock(&out->lock); return ret; } else { if (out->pcm) { if (out->muted) memset((void *)buffer, 0, bytes); ALOGVV("%s: writing buffer (%zu bytes) to pcm device", __func__, bytes); long ns = 0; if (out->config.rate) ns = pcm_bytes_to_frames(out->pcm, bytes)*1000000000LL/ out->config.rate; bool use_mmap = is_mmap_usecase(out->usecase) || out->realtime; request_out_focus(out, ns); if (use_mmap) ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes); else if (out->hal_op_format != out->hal_ip_format && out->convert_buffer != NULL) { memcpy_by_audio_format(out->convert_buffer, out->hal_op_format, buffer, out->hal_ip_format, out->config.period_size * out->config.channels); ret = pcm_write(out->pcm, out->convert_buffer, (out->config.period_size * out->config.channels * format_to_bitwidth_table[out->hal_op_format])); } else { /* * To avoid underrun in DSP when the application is not pumping * data at required rate, check for the no. of bytes and ignore * pcm_write if it is less than actual buffer size. * It is a work around to a change in compress VOIP driver. */ if ((out->flags & AUDIO_OUTPUT_FLAG_VOIP_RX) && bytes < (out->config.period_size * out->config.channels * audio_bytes_per_sample(out->format))) { size_t voip_buf_size = out->config.period_size * out->config.channels * audio_bytes_per_sample(out->format); ALOGE("%s:VOIP underrun: bytes received %zu, required:%zu\n", __func__, bytes, voip_buf_size); usleep(((uint64_t)voip_buf_size - bytes) * 1000000 / audio_stream_out_frame_size(stream) / out_get_sample_rate(&out->stream.common)); ret = 0; } else ret = pcm_write(out->pcm, (void *)buffer, bytes); } release_out_focus(out); if (ret < 0) ret = -errno; else if (ret == 0 && (audio_bytes_per_sample(out->format) != 0)) out->written += bytes / (out->config.channels * audio_bytes_per_sample(out->format)); else ret = -EINVAL; } } exit: /* ToDo: There may be a corner case when SSR happens back to back during start/stop. Need to post different error to handle that. */ if (-ENETRESET == ret) { set_snd_card_state(adev,SND_CARD_STATE_OFFLINE); } pthread_mutex_unlock(&out->lock); if (ret != 0) { if (out->pcm) ALOGE("%s: error %d, %s", __func__, (int)ret, pcm_get_error(out->pcm)); if (out->usecase == USECASE_COMPRESS_VOIP_CALL) { pthread_mutex_lock(&adev->lock); voice_extn_compress_voip_close_output_stream(&out->stream.common); out->started = 0; pthread_mutex_unlock(&adev->lock); out->standby = true; } out_standby(&out->stream.common); if (!(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD)) usleep((uint64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) / out_get_sample_rate(&out->stream.common)); if (audio_extn_passthru_is_passthrough_stream(out)) { ALOGE("%s: write error, ret = %d", __func__, ret); return ret; } } return bytes; } static int out_get_render_position(const struct audio_stream_out *stream, uint32_t *dsp_frames) { struct stream_out *out = (struct stream_out *)stream; struct audio_device *adev = out->dev; if (dsp_frames == NULL) return -EINVAL; *dsp_frames = 0; if (is_offload_usecase(out->usecase)) { ssize_t ret = 0; /* Below piece of code is not guarded against any lock beacuse audioFliner serializes * this operation and adev_close_output_stream(where out gets reset). */ if (!out->non_blocking && (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) { *dsp_frames = get_actual_pcm_frames_rendered(out); ALOGVV("dsp_frames %d sampleRate %d",(int)*dsp_frames,out->sample_rate); return 0; } lock_output_stream(out); if (out->compr != NULL && out->non_blocking) { ret = compress_get_tstamp(out->compr, (unsigned long *)dsp_frames, &out->sample_rate); if (ret < 0) ret = -errno; ALOGVV("%s rendered frames %d sample_rate %d", __func__, *dsp_frames, out->sample_rate); } pthread_mutex_unlock(&out->lock); if (-ENETRESET == ret) { ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver"); set_snd_card_state(adev,SND_CARD_STATE_OFFLINE); return -EINVAL; } else if(ret < 0) { ALOGE(" ERROR: Unable to get time stamp from compress driver"); return -EINVAL; } else if (get_snd_card_state(adev) == SND_CARD_STATE_OFFLINE){ /* * Handle corner case where compress session is closed during SSR * and timestamp is queried */ ALOGE(" ERROR: sound card not active, return error"); return -EINVAL; } else { return 0; } } else if (audio_is_linear_pcm(out->format)) { *dsp_frames = out->written; return 0; } else return -EINVAL; } static int out_add_audio_effect(const struct audio_stream *stream __unused, effect_handle_t effect __unused) { return 0; } static int out_remove_audio_effect(const struct audio_stream *stream __unused, effect_handle_t effect __unused) { return 0; } static int out_get_next_write_timestamp(const struct audio_stream_out *stream __unused, int64_t *timestamp __unused) { return -EINVAL; } static int out_get_presentation_position(const struct audio_stream_out *stream, uint64_t *frames, struct timespec *timestamp) { struct stream_out *out = (struct stream_out *)stream; int ret = -1; unsigned long dsp_frames; /* below piece of code is not guarded against any lock because audioFliner serializes * this operation and adev_close_output_stream( where out gets reset). */ if (is_offload_usecase(out->usecase) && !out->non_blocking && (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) { *frames = get_actual_pcm_frames_rendered(out); /* this is the best we can do */ clock_gettime(CLOCK_MONOTONIC, timestamp); ALOGVV("frames %lld playedat %lld",(long long int)*frames, timestamp->tv_sec * 1000000LL + timestamp->tv_nsec / 1000); return 0; } lock_output_stream(out); if (is_offload_usecase(out->usecase) && out->compr != NULL && out->non_blocking) { ret = compress_get_tstamp(out->compr, &dsp_frames, &out->sample_rate); ALOGVV("%s rendered frames %ld sample_rate %d", __func__, dsp_frames, out->sample_rate); *frames = dsp_frames; if (ret < 0) ret = -errno; if (-ENETRESET == ret) { ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver"); set_snd_card_state(adev,SND_CARD_STATE_OFFLINE); ret = -EINVAL; } else ret = 0; /* this is the best we can do */ clock_gettime(CLOCK_MONOTONIC, timestamp); } else { if (out->pcm) { unsigned int avail; if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) { size_t kernel_buffer_size = out->config.period_size * out->config.period_count; int64_t signed_frames = out->written - kernel_buffer_size + avail; // This adjustment accounts for buffering after app processor. // It is based on estimated DSP latency per use case, rather than exact. signed_frames -= (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL); // It would be unusual for this value to be negative, but check just in case ... if (signed_frames >= 0) { *frames = signed_frames; ret = 0; } } } else if (adev->snd_card_status.state == SND_CARD_STATE_OFFLINE) { *frames = out->written; clock_gettime(CLOCK_MONOTONIC, timestamp); ret = 0; } } pthread_mutex_unlock(&out->lock); return ret; } static int out_set_callback(struct audio_stream_out *stream, stream_callback_t callback, void *cookie) { struct stream_out *out = (struct stream_out *)stream; int ret; ALOGV("%s", __func__); lock_output_stream(out); out->client_callback = callback; out->client_cookie = cookie; if (out->adsp_hdlr_stream_handle) { ret = audio_extn_adsp_hdlr_stream_set_callback( out->adsp_hdlr_stream_handle, callback, cookie); if (ret) ALOGW("%s:adsp hdlr callback registration failed %d", __func__, ret); } pthread_mutex_unlock(&out->lock); return 0; } static int out_pause(struct audio_stream_out* stream) { struct stream_out *out = (struct stream_out *)stream; int status = -ENOSYS; ALOGV("%s", __func__); if (is_offload_usecase(out->usecase)) { ALOGD("copl(%p):pause compress driver", out); lock_output_stream(out); if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) { struct audio_device *adev = out->dev; int snd_scard_state = get_snd_card_state(adev); if (SND_CARD_STATE_ONLINE == snd_scard_state) status = compress_pause(out->compr); out->offload_state = OFFLOAD_STATE_PAUSED; if (audio_extn_passthru_is_active()) { ALOGV("offload use case, pause passthru"); audio_extn_passthru_on_pause(out); } audio_extn_dts_eagle_fade(adev, false, out); audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate, popcount(out->channel_mask), 0); } pthread_mutex_unlock(&out->lock); } return status; } static int out_resume(struct audio_stream_out* stream) { struct stream_out *out = (struct stream_out *)stream; int status = -ENOSYS; ALOGV("%s", __func__); if (is_offload_usecase(out->usecase)) { ALOGD("copl(%p):resume compress driver", out); status = 0; lock_output_stream(out); if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) { struct audio_device *adev = out->dev; int snd_scard_state = get_snd_card_state(adev); if (SND_CARD_STATE_ONLINE == snd_scard_state) { if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { pthread_mutex_lock(&out->dev->lock); ALOGV("offload resume, check and set hdmi backend again"); pthread_mutex_unlock(&out->dev->lock); } status = compress_resume(out->compr); } if (!status) { out->offload_state = OFFLOAD_STATE_PLAYING; } audio_extn_dts_eagle_fade(adev, true, out); audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate, popcount(out->channel_mask), 1); } pthread_mutex_unlock(&out->lock); } return status; } static int out_drain(struct audio_stream_out* stream, audio_drain_type_t type ) { struct stream_out *out = (struct stream_out *)stream; int status = -ENOSYS; ALOGV("%s", __func__); if (is_offload_usecase(out->usecase)) { lock_output_stream(out); if (type == AUDIO_DRAIN_EARLY_NOTIFY) status = send_offload_cmd_l(out, OFFLOAD_CMD_PARTIAL_DRAIN); else status = send_offload_cmd_l(out, OFFLOAD_CMD_DRAIN); pthread_mutex_unlock(&out->lock); } return status; } static int out_flush(struct audio_stream_out* stream) { struct stream_out *out = (struct stream_out *)stream; ALOGV("%s", __func__); if (is_offload_usecase(out->usecase)) { ALOGD("copl(%p):calling compress flush", out); lock_output_stream(out); if (out->offload_state == OFFLOAD_STATE_PAUSED) { stop_compressed_output_l(out); out->written = 0; } else { ALOGW("%s called in invalid state %d", __func__, out->offload_state); } pthread_mutex_unlock(&out->lock); ALOGD("copl(%p):out of compress flush", out); return 0; } return -ENOSYS; } /** audio_stream_in implementation **/ static uint32_t in_get_sample_rate(const struct audio_stream *stream) { struct stream_in *in = (struct stream_in *)stream; return in->config.rate; } static int in_set_sample_rate(struct audio_stream *stream __unused, uint32_t rate __unused) { return -ENOSYS; } static size_t in_get_buffer_size(const struct audio_stream *stream) { struct stream_in *in = (struct stream_in *)stream; if(in->usecase == USECASE_COMPRESS_VOIP_CALL) return voice_extn_compress_voip_in_get_buffer_size(in); else if(audio_extn_compr_cap_usecase_supported(in->usecase)) return audio_extn_compr_cap_get_buffer_size(in->config.format); else if(audio_extn_cin_attached_usecase(in->usecase)) return audio_extn_cin_get_buffer_size(in); return in->config.period_size * in->af_period_multiplier * audio_stream_in_frame_size((const struct audio_stream_in *)stream); } static uint32_t in_get_channels(const struct audio_stream *stream) { struct stream_in *in = (struct stream_in *)stream; return in->channel_mask; } static audio_format_t in_get_format(const struct audio_stream *stream) { struct stream_in *in = (struct stream_in *)stream; return in->format; } static int in_set_format(struct audio_stream *stream __unused, audio_format_t format __unused) { return -ENOSYS; } static int in_standby(struct audio_stream *stream) { struct stream_in *in = (struct stream_in *)stream; struct audio_device *adev = in->dev; int status = 0; ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__, stream, in->usecase, use_case_table[in->usecase]); lock_input_stream(in); if (!in->standby && in->is_st_session) { ALOGD("%s: sound trigger pcm stop lab", __func__); audio_extn_sound_trigger_stop_lab(in); in->standby = 1; } if (!in->standby) { if (adev->adm_deregister_stream) adev->adm_deregister_stream(adev->adm_data, in->capture_handle); pthread_mutex_lock(&adev->lock); in->standby = true; if (in->usecase == USECASE_COMPRESS_VOIP_CALL) { voice_extn_compress_voip_close_input_stream(stream); ALOGD("VOIP input entered standby"); } else { if (audio_extn_cin_attached_usecase(in->usecase)) audio_extn_cin_stop_input_stream(in); if (in->pcm) { pcm_close(in->pcm); in->pcm = NULL; } status = stop_input_stream(in); } pthread_mutex_unlock(&adev->lock); } pthread_mutex_unlock(&in->lock); ALOGV("%s: exit: status(%d)", __func__, status); return status; } static int in_dump(const struct audio_stream *stream __unused, int fd __unused) { return 0; } static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) { struct stream_in *in = (struct stream_in *)stream; struct audio_device *adev = in->dev; struct str_parms *parms; char value[32]; int ret = 0, val = 0, err; ALOGD("%s: enter: kvpairs=%s", __func__, kvpairs); parms = str_parms_create_str(kvpairs); if (!parms) goto error; lock_input_stream(in); pthread_mutex_lock(&adev->lock); err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value)); if (err >= 0) { val = atoi(value); /* no audio source uses val == 0 */ if ((in->source != val) && (val != 0)) { in->source = val; if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) && (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) && (voice_extn_compress_voip_is_format_supported(in->format)) && (in->config.rate == 8000 || in->config.rate == 16000 || in->config.rate == 32000 || in->config.rate == 48000 ) && (audio_channel_count_from_in_mask(in->channel_mask) == 1)) { err = voice_extn_compress_voip_open_input_stream(in); if (err != 0) { ALOGE("%s: Compress voip input cannot be opened, error:%d", __func__, err); } } } } err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); if (err >= 0) { val = atoi(value); if (((int)in->device != val) && (val != 0)) { in->device = val; /* If recording is in progress, change the tx device to new device */ if (!in->standby && !in->is_st_session) { ALOGV("update input routing change"); if (adev->adm_on_routing_change) adev->adm_on_routing_change(adev->adm_data, in->capture_handle); ret = select_devices(adev, in->usecase); } } } err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_PROFILE, value, sizeof(value)); if (err >= 0) { strlcpy(in->profile, value, sizeof(in->profile)); ALOGV("updating stream profile with value '%s'", in->profile); audio_extn_utils_update_stream_input_app_type_cfg(adev->platform, &adev->streams_input_cfg_list, in->device, in->flags, in->format, in->sample_rate, in->bit_width, in->profile, &in->app_type_cfg); } pthread_mutex_unlock(&adev->lock); pthread_mutex_unlock(&in->lock); str_parms_destroy(parms); error: ALOGV("%s: exit: status(%d)", __func__, ret); return ret; } static char* in_get_parameters(const struct audio_stream *stream, const char *keys) { struct stream_in *in = (struct stream_in *)stream; struct str_parms *query = str_parms_create_str(keys); char *str; struct str_parms *reply = str_parms_create(); if (!query || !reply) { if (reply) { str_parms_destroy(reply); } if (query) { str_parms_destroy(query); } ALOGE("in_get_parameters: failed to create query or reply"); return NULL; } ALOGV("%s: enter: keys - %s", __func__, keys); voice_extn_in_get_parameters(in, query, reply); str = str_parms_to_str(reply); str_parms_destroy(query); str_parms_destroy(reply); ALOGV("%s: exit: returns - %s", __func__, str); return str; } static int in_set_gain(struct audio_stream_in *stream __unused, float gain __unused) { return 0; } static ssize_t in_read(struct audio_stream_in *stream, void *buffer, size_t bytes) { struct stream_in *in = (struct stream_in *)stream; struct audio_device *adev = in->dev; int ret = -1; int snd_scard_state = get_snd_card_state(adev); size_t bytes_read = 0; lock_input_stream(in); if (in->is_st_session) { ALOGVV(" %s: reading on st session bytes=%zu", __func__, bytes); /* Read from sound trigger HAL */ audio_extn_sound_trigger_read(in, buffer, bytes); pthread_mutex_unlock(&in->lock); return bytes; } if (SND_CARD_STATE_OFFLINE == snd_scard_state) { ALOGD(" %s: sound card is not active/SSR state", __func__); ret= -EIO;; goto exit; } if (in->standby) { pthread_mutex_lock(&adev->lock); if (in->usecase == USECASE_COMPRESS_VOIP_CALL) ret = voice_extn_compress_voip_start_input_stream(in); else ret = start_input_stream(in); pthread_mutex_unlock(&adev->lock); if (ret != 0) { goto exit; } in->standby = 0; } // what's the duration requested by the client? long ns = 0; if (in->pcm && in->config.rate) ns = pcm_bytes_to_frames(in->pcm, bytes)*1000000000LL/ in->config.rate; request_in_focus(in, ns); bool use_mmap = is_mmap_usecase(in->usecase) || in->realtime; if (audio_extn_cin_attached_usecase(in->usecase)) { ret = audio_extn_cin_read(in, buffer, bytes, &bytes_read); } else if (in->pcm) { if (audio_extn_ssr_get_stream() == in) { ret = audio_extn_ssr_read(stream, buffer, bytes); } else if (audio_extn_compr_cap_usecase_supported(in->usecase)) { ret = audio_extn_compr_cap_read(in, buffer, bytes); } else if (use_mmap) { ret = pcm_mmap_read(in->pcm, buffer, bytes); } else { ret = pcm_read(in->pcm, buffer, bytes); /* data from DSP comes in 24_8 format, convert it to 8_24 */ if (!ret && bytes > 0 && (in->format == AUDIO_FORMAT_PCM_8_24_BIT)) { if (audio_extn_utils_convert_format_24_8_to_8_24(buffer, bytes) != bytes) { ret = -EINVAL; goto exit; } } else if (ret < 0) { ret = -errno; } } /* bytes read is always set to bytes for non compress usecases */ bytes_read = bytes; } release_in_focus(in); /* * Instead of writing zeroes here, we could trust the hardware * to always provide zeroes when muted. */ if (ret == 0 && voice_get_mic_mute(adev) && !voice_is_in_call_rec_stream(in) && in->usecase != USECASE_AUDIO_RECORD_AFE_PROXY) memset(buffer, 0, bytes); exit: /* ToDo: There may be a corner case when SSR happens back to back during start/stop. Need to post different error to handle that. */ if (-ENETRESET == ret) set_snd_card_state(adev,SND_CARD_STATE_OFFLINE); pthread_mutex_unlock(&in->lock); if (ret != 0) { if (in->usecase == USECASE_COMPRESS_VOIP_CALL) { pthread_mutex_lock(&adev->lock); voice_extn_compress_voip_close_input_stream(&in->stream.common); pthread_mutex_unlock(&adev->lock); in->standby = true; } if (!audio_extn_cin_attached_usecase(in->usecase)) { bytes_read = bytes; memset(buffer, 0, bytes); } in_standby(&in->stream.common); ALOGV("%s: read failed status %d- sleeping for buffer duration", __func__, ret); usleep((uint64_t)bytes * 1000000 / audio_stream_in_frame_size(stream) / in_get_sample_rate(&in->stream.common)); } return bytes_read; } static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream __unused) { return 0; } static int add_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect, bool enable) { struct stream_in *in = (struct stream_in *)stream; int status = 0; effect_descriptor_t desc; status = (*effect)->get_descriptor(effect, &desc); if (status != 0) return status; lock_input_stream(in); pthread_mutex_lock(&in->dev->lock); if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) && in->enable_aec != enable && (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0)) { in->enable_aec = enable; if (!in->standby) select_devices(in->dev, in->usecase); } if (in->enable_ns != enable && (memcmp(&desc.type, FX_IID_NS, sizeof(effect_uuid_t)) == 0)) { in->enable_ns = enable; if (!in->standby) select_devices(in->dev, in->usecase); } pthread_mutex_unlock(&in->dev->lock); pthread_mutex_unlock(&in->lock); return 0; } static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) { ALOGV("%s: effect %p", __func__, effect); return add_remove_audio_effect(stream, effect, true); } static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) { ALOGV("%s: effect %p", __func__, effect); return add_remove_audio_effect(stream, effect, false); } int adev_open_output_stream(struct audio_hw_device *dev, audio_io_handle_t handle, audio_devices_t devices, audio_output_flags_t flags, struct audio_config *config, struct audio_stream_out **stream_out, const char *address __unused) { struct audio_device *adev = (struct audio_device *)dev; struct stream_out *out; int ret = 0; audio_format_t format; struct adsp_hdlr_stream_cfg hdlr_stream_cfg; bool is_direct_passthough = false; *stream_out = NULL; if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) && (SND_CARD_STATE_OFFLINE == get_snd_card_state(adev))) { ALOGE("sound card is not active rejecting compress output open request"); return -EINVAL; } out = (struct stream_out *)calloc(1, sizeof(struct stream_out)); ALOGD("%s: enter: format(%#x) sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)\ stream_handle(%p)", __func__, config->format, config->sample_rate, config->channel_mask, devices, flags, &out->stream); if (!out) { return -ENOMEM; } pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL); pthread_mutex_init(&out->pre_lock, (const pthread_mutexattr_t *) NULL); pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL); if (devices == AUDIO_DEVICE_NONE) devices = AUDIO_DEVICE_OUT_SPEAKER; out->flags = flags; out->devices = devices; out->dev = adev; format = out->format = config->format; out->sample_rate = config->sample_rate; out->channel_mask = config->channel_mask; out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO; out->handle = handle; out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH; out->non_blocking = 0; out->convert_buffer = NULL; out->started = 0; if ((flags & AUDIO_OUTPUT_FLAG_BD) && (property_get_bool("audio.matrix.limiter.enable", false))) platform_set_device_params(out, DEVICE_PARAM_LIMITER_ID, 1); if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL && (flags & AUDIO_OUTPUT_FLAG_DIRECT)) { pthread_mutex_lock(&adev->lock); ALOGV("AUDIO_DEVICE_OUT_AUX_DIGITAL and DIRECT|OFFLOAD, check hdmi caps"); ret = read_hdmi_sink_caps(out); pthread_mutex_unlock(&adev->lock); if (ret != 0) { if (ret == -ENOSYS) { /* ignore and go with default */ ret = 0; } else { ALOGE("error reading hdmi sink caps"); goto error_open; } } } /* Init use case and pcm_config */ if ((out->flags & AUDIO_OUTPUT_FLAG_DIRECT) && !(out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD || (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) && (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) { /* * Do not handle stereo output in Multi-channel cases * Stereo case is handled in normal playback path */ if (out->supported_channel_masks[0] == AUDIO_CHANNEL_OUT_STEREO) ret = AUDIO_CHANNEL_OUT_STEREO; } 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; if (config->format == 0) config->format = AUDIO_FORMAT_PCM_16_BIT; out->channel_mask = config->channel_mask; out->sample_rate = config->sample_rate; out->format = config->format; 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 || voice_extn_compress_voip_is_active(out->dev)) && (out->flags == (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_VOIP_RX)) && (voice_extn_compress_voip_is_config_supported(config))) { ret = voice_extn_compress_voip_open_output_stream(out); if (ret != 0) { ALOGE("%s: Compress voip output cannot be opened, error:%d", __func__, ret); goto error_open; } } else if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) || (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM)) { if (config->offload_info.version != AUDIO_INFO_INITIALIZER.version || config->offload_info.size != AUDIO_INFO_INITIALIZER.size) { ALOGE("%s: Unsupported Offload information", __func__); ret = -EINVAL; goto error_open; } if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { if(config->offload_info.format == 0) config->offload_info.format = out->supported_formats[0]; if (config->offload_info.sample_rate == 0) config->offload_info.sample_rate = out->supported_sample_rates[0]; } if (!is_supported_format(config->offload_info.format) && !audio_extn_passthru_is_supported_format(config->offload_info.format)) { ALOGE("%s: Unsupported audio format %x " , __func__, config->offload_info.format); ret = -EINVAL; goto error_open; } /* TrueHD only supported for 48k multiples (48k, 96k, 192k) */ if ((config->offload_info.format == AUDIO_FORMAT_DOLBY_TRUEHD) && (audio_extn_passthru_is_passthrough_stream(out)) && !((config->sample_rate == 48000) || (config->sample_rate == 96000) || (config->sample_rate == 192000))) { ALOGE("%s: Unsupported sample rate %d for audio format %x", __func__, config->sample_rate, config->offload_info.format); ret = -EINVAL; goto error_open; } out->compr_config.codec = (struct snd_codec *) calloc(1, sizeof(struct snd_codec)); if (!out->compr_config.codec) { ret = -ENOMEM; goto error_open; } if (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) { out->stream.set_callback = out_set_callback; out->stream.pause = out_pause; out->stream.flush = out_flush; out->stream.resume = out_resume; out->usecase = get_offload_usecase(adev, true); ALOGV("DIRECT_PCM usecase ... usecase selected %d ", out->usecase); } else { 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->usecase = get_offload_usecase(adev, false); ALOGV("Compress Offload usecase .. usecase selected %d", out->usecase); } if (out->usecase == USECASE_INVALID) { if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL && config->format == 0 && config->sample_rate == 0 && config->channel_mask == 0) { ALOGI("%s dummy open to query sink capability",__func__); out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD; } else { ALOGE("%s, Max allowed OFFLOAD usecase reached ... ", __func__); ret = -EEXIST; goto error_open; } } if (config->offload_info.channel_mask) out->channel_mask = config->offload_info.channel_mask; else if (config->channel_mask) { out->channel_mask = config->channel_mask; config->offload_info.channel_mask = config->channel_mask; } else { ALOGE("out->channel_mask not set for OFFLOAD/DIRECT_PCM"); ret = -EINVAL; goto error_open; } format = out->format = config->offload_info.format; out->sample_rate = config->offload_info.sample_rate; out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH; out->compr_config.codec->id = get_snd_codec_id(config->offload_info.format); if (audio_extn_utils_is_dolby_format(config->offload_info.format)) { audio_extn_dolby_send_ddp_endp_params(adev); audio_extn_dolby_set_dmid(adev); } out->compr_config.codec->sample_rate = config->offload_info.sample_rate; out->compr_config.codec->bit_rate = config->offload_info.bit_rate; out->compr_config.codec->ch_in = audio_channel_count_from_out_mask(out->channel_mask); out->compr_config.codec->ch_out = out->compr_config.codec->ch_in; /* Update bit width only for non passthrough usecases. * For passthrough usecases, the output will always be opened @16 bit */ if (!audio_extn_passthru_is_passthrough_stream(out)) out->bit_width = AUDIO_OUTPUT_BIT_WIDTH; if (out->flags & AUDIO_OUTPUT_FLAG_TIMESTAMP) out->compr_config.codec->flags |= COMPRESSED_TIMESTAMP_FLAG; ALOGVV("%s : out->compr_config.codec->flags -> (%#x) ", __func__, out->compr_config.codec->flags); /*TODO: Do we need to change it for passthrough */ out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW; if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC) out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW; else if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_ADTS) out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS; else if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_AAC_LATM) out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4LATM; if ((config->offload_info.format & AUDIO_FORMAT_MAIN_MASK) == AUDIO_FORMAT_PCM) { /*Based on platform support, configure appropriate alsa format for corresponding *hal input format. */ out->compr_config.codec->format = hal_format_to_alsa( config->offload_info.format); out->hal_op_format = alsa_format_to_hal( out->compr_config.codec->format); out->hal_ip_format = out->format; /*for direct PCM playback populate bit_width based on selected alsa format as *hal input format and alsa format might differ based on platform support. */ out->bit_width = audio_bytes_per_sample( out->hal_op_format) << 3; out->compr_config.fragments = DIRECT_PCM_NUM_FRAGMENTS; /* Check if alsa session is configured with the same format as HAL input format, * if not then derive correct fragment size needed to accomodate the * conversion of HAL input format to alsa format. */ audio_extn_utils_update_direct_pcm_fragment_size(out); /*if hal input and output fragment size is different this indicates HAL input format is *not same as the alsa format */ if (out->hal_fragment_size != out->compr_config.fragment_size) { /*Allocate a buffer to convert input data to the alsa configured format. *size of convert buffer is equal to the size required to hold one fragment size *worth of pcm data, this is because flinger does not write more than fragment_size */ out->convert_buffer = calloc(1,out->compr_config.fragment_size); if (out->convert_buffer == NULL){ ALOGE("Allocation failed for convert buffer for size %d", out->compr_config.fragment_size); ret = -ENOMEM; goto error_open; } } } else if (audio_extn_passthru_is_passthrough_stream(out)) { out->compr_config.fragment_size = audio_extn_passthru_get_buffer_size(&config->offload_info); out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS; } else { out->compr_config.fragment_size = platform_get_compress_offload_buffer_size(&config->offload_info); out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS; } if (out->flags & AUDIO_OUTPUT_FLAG_TIMESTAMP) { out->compr_config.fragment_size += sizeof(struct snd_codec_metadata); } if (config->offload_info.format == AUDIO_FORMAT_FLAC) out->compr_config.codec->options.flac_dec.sample_size = AUDIO_OUTPUT_BIT_WIDTH; if (config->offload_info.format == AUDIO_FORMAT_APTX) { audio_extn_send_aptx_dec_bt_addr_to_dsp(out); } if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) out->non_blocking = 1; if ((flags & AUDIO_OUTPUT_FLAG_TIMESTAMP) && (flags & AUDIO_OUTPUT_FLAG_HW_AV_SYNC)) { out->render_mode = RENDER_MODE_AUDIO_STC_MASTER; } else if(flags & AUDIO_OUTPUT_FLAG_TIMESTAMP) { out->render_mode = RENDER_MODE_AUDIO_MASTER; } else { out->render_mode = RENDER_MODE_AUDIO_NO_TIMESTAMP; } memset(&out->channel_map_param, 0, sizeof(struct audio_out_channel_map_param)); out->send_new_metadata = 1; out->send_next_track_params = false; out->is_compr_metadata_avail = false; out->offload_state = OFFLOAD_STATE_IDLE; out->playback_started = 0; audio_extn_dts_create_state_notifier_node(out->usecase); ALOGV("%s: offloaded output offload_info version %04x bit rate %d", __func__, config->offload_info.version, config->offload_info.bit_rate); /* Check if DSD audio format is supported in codec * and there is no active native DSD use case */ if ((config->format == AUDIO_FORMAT_DSD) && (!platform_check_codec_dsd_support(adev->platform) || audio_is_dsd_native_stream_active(adev))) { ret = -EINVAL; goto error_open; } /* Disable gapless if any of the following is true * passthrough playback * AV playback * Direct PCM playback */ if (audio_extn_passthru_is_passthrough_stream(out) || (config->format == AUDIO_FORMAT_DSD) || (config->format == AUDIO_FORMAT_IEC61937) || config->offload_info.has_video || out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) { check_and_set_gapless_mode(adev, false); } else check_and_set_gapless_mode(adev, true); if (audio_extn_passthru_is_passthrough_stream(out)) { out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH; } if (config->format == AUDIO_FORMAT_DSD) { out->flags |= AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH; out->compr_config.codec->compr_passthr = PASSTHROUGH_DSD; } create_offload_callback_thread(out); } else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) { ret = voice_extn_check_and_set_incall_music_usecase(adev, out); if (ret != 0) { ALOGE("%s: Incall music delivery usecase cannot be set error:%d", __func__, ret); goto error_open; } } else if (out->devices == AUDIO_DEVICE_OUT_TELEPHONY_TX) { if (config->sample_rate == 0) config->sample_rate = AFE_PROXY_SAMPLING_RATE; if (config->sample_rate != 48000 && config->sample_rate != 16000 && config->sample_rate != 8000) { config->sample_rate = AFE_PROXY_SAMPLING_RATE; ret = -EINVAL; goto error_open; } out->sample_rate = config->sample_rate; out->config.rate = config->sample_rate; if (config->format == AUDIO_FORMAT_DEFAULT) config->format = AUDIO_FORMAT_PCM_16_BIT; if (config->format != AUDIO_FORMAT_PCM_16_BIT) { config->format = AUDIO_FORMAT_PCM_16_BIT; ret = -EINVAL; goto error_open; } out->format = config->format; out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY; out->config = pcm_config_afe_proxy_playback; adev->voice_tx_output = out; } else { unsigned int channels = 0; /*Update config params to default if not set by the caller*/ if (config->sample_rate == 0) config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE; if (config->channel_mask == AUDIO_CHANNEL_NONE) config->channel_mask = AUDIO_CHANNEL_OUT_STEREO; if (config->format == AUDIO_FORMAT_DEFAULT) config->format = AUDIO_FORMAT_PCM_16_BIT; channels = audio_channel_count_from_out_mask(out->channel_mask); if (out->flags & AUDIO_OUTPUT_FLAG_INTERACTIVE) { out->usecase = get_interactive_usecase(adev); out->config = pcm_config_low_latency; } else if (out->flags & AUDIO_OUTPUT_FLAG_RAW) { out->usecase = USECASE_AUDIO_PLAYBACK_ULL; out->realtime = may_use_noirq_mode(adev, USECASE_AUDIO_PLAYBACK_ULL, out->flags); out->config = out->realtime ? pcm_config_rt : pcm_config_low_latency; } else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) { out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY; out->config = pcm_config_low_latency; } else if (out->flags & AUDIO_OUTPUT_FLAG_DEEP_BUFFER) { out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER; out->config = pcm_config_deep_buffer; out->config.period_size = get_output_period_size(config->sample_rate, out->format, channels, DEEP_BUFFER_OUTPUT_PERIOD_DURATION); if (out->config.period_size <= 0) { ALOGE("Invalid configuration period size is not valid"); ret = -EINVAL; goto error_open; } } else { /* primary path is the default path selected if no other outputs are available/suitable */ out->usecase = USECASE_AUDIO_PLAYBACK_PRIMARY; out->config = PCM_CONFIG_AUDIO_PLAYBACK_PRIMARY; } out->hal_ip_format = format = out->format; out->config.format = hal_format_to_pcm(out->hal_ip_format); out->hal_op_format = pcm_format_to_hal(out->config.format); out->bit_width = format_to_bitwidth_table[out->hal_op_format] << 3; out->config.rate = config->sample_rate; out->sample_rate = out->config.rate; out->config.channels = channels; if (out->hal_ip_format != out->hal_op_format) { uint32_t buffer_size = out->config.period_size * format_to_bitwidth_table[out->hal_op_format] * out->config.channels; out->convert_buffer = calloc(1, buffer_size); if (out->convert_buffer == NULL){ ALOGE("Allocation failed for convert buffer for size %d", out->compr_config.fragment_size); ret = -ENOMEM; goto error_open; } ALOGD("Convert buffer allocated of size %d", buffer_size); } } ALOGV("%s devices:%d, format:%x, out->sample_rate:%d,out->bit_width:%d out->format:%d out->flags:%x, flags: %x usecase %d", __func__, devices, format, out->sample_rate, out->bit_width, out->format, out->flags, flags, out->usecase); /* TODO remove this hardcoding and check why width is zero*/ if (out->bit_width == 0) out->bit_width = 16; audio_extn_utils_update_stream_output_app_type_cfg(adev->platform, &adev->streams_output_cfg_list, devices, out->flags, format, out->sample_rate, out->bit_width, out->channel_mask, out->profile, &out->app_type_cfg); if ((out->usecase == USECASE_AUDIO_PLAYBACK_PRIMARY) || (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) { /* Ensure the default output is not selected twice */ if(adev->primary_output == NULL) adev->primary_output = out; else { ALOGE("%s: Primary output is already opened", __func__); ret = -EEXIST; goto error_open; } } /* Check if this usecase is already existing */ pthread_mutex_lock(&adev->lock); if ((get_usecase_from_list(adev, out->usecase) != NULL) && (out->usecase != USECASE_COMPRESS_VOIP_CALL)) { ALOGE("%s: Usecase (%d) is already present", __func__, out->usecase); pthread_mutex_unlock(&adev->lock); ret = -EEXIST; goto error_open; } pthread_mutex_unlock(&adev->lock); out->stream.common.get_sample_rate = out_get_sample_rate; out->stream.common.set_sample_rate = out_set_sample_rate; out->stream.common.get_buffer_size = out_get_buffer_size; out->stream.common.get_channels = out_get_channels; out->stream.common.get_format = out_get_format; out->stream.common.set_format = out_set_format; out->stream.common.standby = out_standby; out->stream.common.dump = out_dump; out->stream.common.set_parameters = out_set_parameters; out->stream.common.get_parameters = out_get_parameters; out->stream.common.add_audio_effect = out_add_audio_effect; out->stream.common.remove_audio_effect = out_remove_audio_effect; out->stream.get_latency = out_get_latency; out->stream.set_volume = out_set_volume; out->stream.write = out_write; out->stream.get_render_position = out_get_render_position; out->stream.get_next_write_timestamp = out_get_next_write_timestamp; out->stream.get_presentation_position = out_get_presentation_position; out->af_period_multiplier = out->realtime ? af_period_multiplier : 1; out->standby = 1; /* out->muted = false; by calloc() */ /* out->written = 0; by calloc() */ config->format = out->stream.common.get_format(&out->stream.common); config->channel_mask = out->stream.common.get_channels(&out->stream.common); config->sample_rate = out->stream.common.get_sample_rate(&out->stream.common); *stream_out = &out->stream; ALOGD("%s: Stream (%p) picks up usecase (%s)", __func__, &out->stream, use_case_table[out->usecase]); if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) audio_extn_dts_notify_playback_state(out->usecase, 0, out->sample_rate, popcount(out->channel_mask), out->playback_started); /* setup a channel for client <--> adsp communication for stream events */ is_direct_passthough = audio_extn_passthru_is_direct_passthrough(out); if ((out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) || (out->flags & AUDIO_OUTPUT_FLAG_DIRECT_PCM) || (audio_extn_ip_hdlr_intf_supported(config->format, is_direct_passthough, false))) { hdlr_stream_cfg.pcm_device_id = platform_get_pcm_device_id( out->usecase, PCM_PLAYBACK); hdlr_stream_cfg.flags = out->flags; hdlr_stream_cfg.type = PCM_PLAYBACK; ret = audio_extn_adsp_hdlr_stream_open(&out->adsp_hdlr_stream_handle, &hdlr_stream_cfg); if (ret) { ALOGE("%s: adsp_hdlr_stream_open failed %d",__func__, ret); out->adsp_hdlr_stream_handle = NULL; } } if (audio_extn_ip_hdlr_intf_supported(config->format, is_direct_passthough, false)) { ret = audio_extn_ip_hdlr_intf_init(&out->ip_hdlr_handle, NULL, NULL, adev, out->usecase); if (ret < 0) { ALOGE("%s: audio_extn_ip_hdlr_intf_init failed %d",__func__, ret); out->ip_hdlr_handle = NULL; } } ALOGV("%s: exit", __func__); return 0; error_open: if (out->convert_buffer) free(out->convert_buffer); free(out); *stream_out = NULL; ALOGD("%s: exit: ret %d", __func__, ret); return ret; } void adev_close_output_stream(struct audio_hw_device *dev __unused, struct audio_stream_out *stream) { struct stream_out *out = (struct stream_out *)stream; struct audio_device *adev = out->dev; int ret = 0; ALOGD("%s: enter:stream_handle(%p)",__func__, out); /* close adsp hdrl session before standby */ if (out->adsp_hdlr_stream_handle) { ret = audio_extn_adsp_hdlr_stream_close(out->adsp_hdlr_stream_handle); if (ret) ALOGE("%s: adsp_hdlr_stream_close failed %d",__func__, ret); out->adsp_hdlr_stream_handle = NULL; } if (out->ip_hdlr_handle) { audio_extn_ip_hdlr_intf_deinit(out->ip_hdlr_handle); out->ip_hdlr_handle = NULL; } if (out->usecase == USECASE_COMPRESS_VOIP_CALL) { pthread_mutex_lock(&adev->lock); ret = voice_extn_compress_voip_close_output_stream(&stream->common); out->started = 0; pthread_mutex_unlock(&adev->lock); if(ret != 0) ALOGE("%s: Compress voip output cannot be closed, error:%d", __func__, ret); } else out_standby(&stream->common); if (is_offload_usecase(out->usecase)) { audio_extn_dts_remove_state_notifier_node(out->usecase); destroy_offload_callback_thread(out); free_offload_usecase(adev, out->usecase); if (out->compr_config.codec != NULL) free(out->compr_config.codec); } if (is_interactive_usecase(out->usecase)) free_interactive_usecase(adev, out->usecase); if (out->convert_buffer != NULL) { free(out->convert_buffer); out->convert_buffer = NULL; } if (adev->voice_tx_output == out) adev->voice_tx_output = NULL; if (adev->primary_output == out) adev->primary_output = NULL; pthread_cond_destroy(&out->cond); pthread_mutex_destroy(&out->lock); free(stream); ALOGV("%s: exit", __func__); } static void close_compress_sessions(struct audio_device *adev) { struct stream_out *out; struct listnode *node, *tempnode; struct audio_usecase *usecase; pthread_mutex_lock(&adev->lock); list_for_each_safe(node, tempnode, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if (is_offload_usecase(usecase->id)) { if (usecase->stream.out) { ALOGI(" %s closing compress session %d on OFFLINE state", __func__, usecase->id); out = usecase->stream.out; pthread_mutex_unlock(&adev->lock); out_standby(&out->stream.common); pthread_mutex_lock(&adev->lock); tempnode = list_head(&adev->usecase_list); } } } pthread_mutex_unlock(&adev->lock); } static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) { struct audio_device *adev = (struct audio_device *)dev; struct str_parms *parms; char value[32]; int val; int ret; int status = 0; ALOGD("%s: enter: %s", __func__, kvpairs); parms = str_parms_create_str(kvpairs); if (!parms) goto error; ret = str_parms_get_str(parms, "SND_CARD_STATUS", value, sizeof(value)); if (ret >= 0) { char *snd_card_status = value+2; if (strstr(snd_card_status, "OFFLINE")) { ALOGD("Received sound card OFFLINE status"); set_snd_card_state(adev,SND_CARD_STATE_OFFLINE); //close compress sessions on OFFLINE status close_compress_sessions(adev); } else if (strstr(snd_card_status, "ONLINE")) { ALOGD("Received sound card ONLINE status"); set_snd_card_state(adev,SND_CARD_STATE_ONLINE); //send dts hpx license if enabled audio_extn_dts_eagle_send_lic(); } } ret = str_parms_get_str(parms, "BT_SCO", 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->bt_sco_on = true; else adev->bt_sco_on = false; } pthread_mutex_lock(&adev->lock); status = voice_set_parameters(adev, parms); if (status != 0) goto done; status = platform_set_parameters(adev->platform, parms); if (status != 0) goto done; ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value)); if (ret >= 0) { /* When set to false, HAL should disable EC and NS */ if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0) adev->bluetooth_nrec = true; else adev->bluetooth_nrec = false; } ret = str_parms_get_str(parms, "screen_state", value, sizeof(value)); if (ret >= 0) { if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0) adev->screen_off = false; else adev->screen_off = true; } ret = str_parms_get_int(parms, "rotation", &val); if (ret >= 0) { bool reverse_speakers = false; switch(val) { // FIXME: note that the code below assumes that the speakers are in the correct placement // relative to the user when the device is rotated 90deg from its default rotation. This // assumption is device-specific, not platform-specific like this code. case 270: reverse_speakers = true; break; case 0: case 90: case 180: break; default: ALOGE("%s: unexpected rotation of %d", __func__, val); status = -EINVAL; } if (status == 0) { if (adev->speaker_lr_swap != reverse_speakers) { adev->speaker_lr_swap = reverse_speakers; // only update the selected device if there is active pcm playback struct audio_usecase *usecase; struct listnode *node; list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if (usecase->type == PCM_PLAYBACK) { select_devices(adev, usecase->id); break; } } } } } ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_SCO_WB, value, sizeof(value)); if (ret >= 0) { if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0) adev->bt_wb_speech_enabled = true; else adev->bt_wb_speech_enabled = false; } ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_CONNECT, value, sizeof(value)); if (ret >= 0) { val = atoi(value); if (val & AUDIO_DEVICE_OUT_AUX_DIGITAL) { ALOGV("cache new ext disp type and edid"); ret = platform_get_ext_disp_type(adev->platform); if (ret < 0) { ALOGE("%s: Failed to query disp type, ret:%d", __func__, ret); status = ret; goto done; } platform_cache_edid(adev->platform); } else if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) || !(val ^ AUDIO_DEVICE_IN_USB_DEVICE)) { /* * Do not allow AFE proxy port usage by WFD source when USB headset is connected. * Per AudioPolicyManager, USB device is higher priority than WFD. * For Voice call over USB headset, voice call audio is routed to AFE proxy ports. * If WFD use case occupies AFE proxy, it may result unintended behavior while * starting voice call on USB */ ret = str_parms_get_str(parms, "card", value, sizeof(value)); if (ret >= 0) { audio_extn_usb_add_device(AUDIO_DEVICE_OUT_USB_DEVICE, atoi(value)); audio_extn_usb_add_device(AUDIO_DEVICE_IN_USB_DEVICE, atoi(value)); } ALOGV("detected USB connect .. disable proxy"); adev->allow_afe_proxy_usage = false; } } ret = str_parms_get_str(parms, AUDIO_PARAMETER_DEVICE_DISCONNECT, value, sizeof(value)); if (ret >= 0) { val = atoi(value); /* * The HDMI / Displayport disconnect handling has been moved to * audio extension to ensure that its parameters are not * invalidated prior to updating sysfs of the disconnect event * Invalidate will be handled by audio_extn_ext_disp_set_parameters() */ if ((val & AUDIO_DEVICE_OUT_USB_DEVICE) || !(val ^ AUDIO_DEVICE_IN_USB_DEVICE)) { ret = str_parms_get_str(parms, "card", value, sizeof(value)); if (ret >= 0) { audio_extn_usb_remove_device(AUDIO_DEVICE_OUT_USB_DEVICE, atoi(value)); audio_extn_usb_remove_device(AUDIO_DEVICE_IN_USB_DEVICE, atoi(value)); } ALOGV("detected USB disconnect .. enable proxy"); adev->allow_afe_proxy_usage = true; } } ret = str_parms_get_str(parms,"reconfigA2dp", value, sizeof(value)); if (ret >= 0) { struct audio_usecase *usecase; struct listnode *node; list_for_each(node, &adev->usecase_list) { usecase = node_to_item(node, struct audio_usecase, list); if ((usecase->type == PCM_PLAYBACK) && (usecase->devices & AUDIO_DEVICE_OUT_ALL_A2DP)){ ALOGD("reconfigure a2dp... forcing device switch"); pthread_mutex_unlock(&adev->lock); lock_output_stream(usecase->stream.out); pthread_mutex_lock(&adev->lock); audio_extn_a2dp_set_handoff_mode(true); //force device switch to re configure encoder select_devices(adev, usecase->id); audio_extn_a2dp_set_handoff_mode(false); pthread_mutex_unlock(&usecase->stream.out->lock); break; } } } //handle vr audio setparam ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_VR_AUDIO_MODE, value, sizeof(value)); if (ret >= 0) { ALOGI("Setting vr mode to be %s", value); if (!strncmp(value, "true", 4)) { adev->vr_audio_mode_enabled = true; ALOGI("Setting vr mode to true"); } else if (!strncmp(value, "false", 5)) { adev->vr_audio_mode_enabled = false; ALOGI("Setting vr mode to false"); } else { ALOGI("wrong vr mode set"); } } audio_extn_set_parameters(adev, parms); done: str_parms_destroy(parms); pthread_mutex_unlock(&adev->lock); error: ALOGV("%s: exit with code(%d)", __func__, status); return status; } static char* adev_get_parameters(const struct audio_hw_device *dev, const char *keys) { struct audio_device *adev = (struct audio_device *)dev; struct str_parms *reply = str_parms_create(); struct str_parms *query = str_parms_create_str(keys); char *str; char value[256] = {0}; int ret = 0; if (!query || !reply) { if (reply) { str_parms_destroy(reply); } if (query) { str_parms_destroy(query); } ALOGE("adev_get_parameters: failed to create query or reply"); return NULL; } ret = str_parms_get_str(query, "SND_CARD_STATUS", value, sizeof(value)); if (ret >=0) { int val = 1; pthread_mutex_lock(&adev->snd_card_status.lock); if (SND_CARD_STATE_OFFLINE == adev->snd_card_status.state) val = 0; pthread_mutex_unlock(&adev->snd_card_status.lock); str_parms_add_int(reply, "SND_CARD_STATUS", val); goto exit; } //handle vr audio getparam ret = str_parms_get_str(query, AUDIO_PARAMETER_KEY_VR_AUDIO_MODE, value, sizeof(value)); if (ret >= 0) { bool vr_audio_enabled = false; pthread_mutex_lock(&adev->lock); vr_audio_enabled = adev->vr_audio_mode_enabled; pthread_mutex_unlock(&adev->lock); ALOGI("getting vr mode to %d", vr_audio_enabled); if (vr_audio_enabled) { str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VR_AUDIO_MODE, "true"); goto exit; } else { str_parms_add_str(reply, AUDIO_PARAMETER_KEY_VR_AUDIO_MODE, "false"); goto exit; } } pthread_mutex_lock(&adev->lock); audio_extn_get_parameters(adev, query, reply); voice_get_parameters(adev, query, reply); platform_get_parameters(adev->platform, query, reply); pthread_mutex_unlock(&adev->lock); exit: str = str_parms_to_str(reply); str_parms_destroy(query); str_parms_destroy(reply); ALOGV("%s: exit: returns - %s", __func__, str); return str; } static int adev_init_check(const struct audio_hw_device *dev __unused) { return 0; } static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) { int ret; struct audio_device *adev = (struct audio_device *)dev; pthread_mutex_lock(&adev->lock); /* cache volume */ ret = voice_set_volume(adev, volume); pthread_mutex_unlock(&adev->lock); return ret; } static int adev_set_master_volume(struct audio_hw_device *dev __unused, float volume __unused) { return -ENOSYS; } static int adev_get_master_volume(struct audio_hw_device *dev __unused, float *volume __unused) { return -ENOSYS; } static int adev_set_master_mute(struct audio_hw_device *dev __unused, bool muted __unused) { return -ENOSYS; } static int adev_get_master_mute(struct audio_hw_device *dev __unused, bool *muted __unused) { return -ENOSYS; } static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) { struct audio_device *adev = (struct audio_device *)dev; pthread_mutex_lock(&adev->lock); if (adev->mode != mode) { ALOGD("%s: mode %d\n", __func__, mode); adev->mode = mode; if ((mode == AUDIO_MODE_NORMAL) && voice_is_in_call(adev)) { voice_stop_call(adev); platform_set_gsm_mode(adev->platform, false); adev->current_call_output = NULL; } } pthread_mutex_unlock(&adev->lock); return 0; } static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) { int ret; pthread_mutex_lock(&adev->lock); ALOGD("%s state %d\n", __func__, state); ret = voice_set_mic_mute((struct audio_device *)dev, state); pthread_mutex_unlock(&adev->lock); return ret; } static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) { *state = voice_get_mic_mute((struct audio_device *)dev); return 0; } static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev __unused, const struct audio_config *config) { int channel_count = audio_channel_count_from_in_mask(config->channel_mask); return get_input_buffer_size(config->sample_rate, config->format, channel_count, false /* is_low_latency: since we don't know, be conservative */); } static int adev_open_input_stream(struct audio_hw_device *dev, audio_io_handle_t handle, audio_devices_t devices, struct audio_config *config, struct audio_stream_in **stream_in, audio_input_flags_t flags, const char *address __unused, audio_source_t source) { struct audio_device *adev = (struct audio_device *)dev; struct stream_in *in; int ret = 0, buffer_size, frame_size; int channel_count = audio_channel_count_from_in_mask(config->channel_mask); bool is_low_latency = false; bool channel_mask_updated = false; *stream_in = NULL; if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) { ALOGE("%s: invalid input parameters", __func__); return -EINVAL; } in = (struct stream_in *)calloc(1, sizeof(struct stream_in)); if (!in) { ALOGE("failed to allocate input stream"); return -ENOMEM; } ALOGD("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x)\ stream_handle(%p) io_handle(%d) source(%d) format %x",__func__, config->sample_rate, config->channel_mask, devices, &in->stream, handle, source, config->format); pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL); pthread_mutex_init(&in->pre_lock, (const pthread_mutexattr_t *) NULL); in->stream.common.get_sample_rate = in_get_sample_rate; in->stream.common.set_sample_rate = in_set_sample_rate; in->stream.common.get_buffer_size = in_get_buffer_size; in->stream.common.get_channels = in_get_channels; in->stream.common.get_format = in_get_format; in->stream.common.set_format = in_set_format; in->stream.common.standby = in_standby; in->stream.common.dump = in_dump; in->stream.common.set_parameters = in_set_parameters; in->stream.common.get_parameters = in_get_parameters; in->stream.common.add_audio_effect = in_add_audio_effect; in->stream.common.remove_audio_effect = in_remove_audio_effect; in->stream.set_gain = in_set_gain; in->stream.read = in_read; in->stream.get_input_frames_lost = in_get_input_frames_lost; in->device = devices; in->source = source; in->dev = adev; in->standby = 1; in->channel_mask = config->channel_mask; in->capture_handle = handle; in->flags = flags; in->usecase = USECASE_AUDIO_RECORD; if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE && (flags & AUDIO_INPUT_FLAG_FAST) != 0) { is_low_latency = true; #if LOW_LATENCY_CAPTURE_USE_CASE in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY; #endif in->realtime = may_use_noirq_mode(adev, in->usecase, in->flags); } in->format = config->format; if (in->realtime) { in->config = pcm_config_audio_capture_rt; in->sample_rate = in->config.rate; in->af_period_multiplier = af_period_multiplier; } else { in->config = pcm_config_audio_capture; in->config.rate = config->sample_rate; in->sample_rate = config->sample_rate; in->af_period_multiplier = 1; } in->bit_width = 16; /* restrict 24 bit capture for unprocessed source only * for other sources if 24 bit requested reject 24 and set 16 bit capture only */ if (config->format == AUDIO_FORMAT_DEFAULT) { config->format = AUDIO_FORMAT_PCM_16_BIT; } else if ((config->format == AUDIO_FORMAT_PCM_FLOAT) || (config->format == AUDIO_FORMAT_PCM_32_BIT) || (config->format == AUDIO_FORMAT_PCM_24_BIT_PACKED) || (config->format == AUDIO_FORMAT_PCM_8_24_BIT)) { bool ret_error = false; in->bit_width = 24; /* 24 bit is restricted to UNPROCESSED source only,also format supported from HAL is 24_packed and 8_24 *> In case of UNPROCESSED source, for 24 bit, if format requested is other than 24_packed return error indicating supported format is 24_packed *> In case of any other source requesting 24 bit or float return error indicating format supported is 16 bit only. on error flinger will retry with supported format passed */ if ((source != AUDIO_SOURCE_UNPROCESSED) && (source != AUDIO_SOURCE_CAMCORDER)) { config->format = AUDIO_FORMAT_PCM_16_BIT; if (config->sample_rate > 48000) config->sample_rate = 48000; ret_error = true; } else if (config->format == AUDIO_FORMAT_PCM_24_BIT_PACKED) { in->config.format = PCM_FORMAT_S24_3LE; } else if (config->format == AUDIO_FORMAT_PCM_8_24_BIT) { in->config.format = PCM_FORMAT_S24_LE; } else { config->format = AUDIO_FORMAT_PCM_24_BIT_PACKED; ret_error = true; } if (ret_error) { ret = -EINVAL; goto err_open; } } /* Update config params with the requested sample rate and channels */ if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) && (adev->mode != AUDIO_MODE_IN_CALL)) { ret = -EINVAL; goto err_open; } if ((in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) || (in->device == AUDIO_DEVICE_IN_PROXY)) { if (config->sample_rate == 0) config->sample_rate = AFE_PROXY_SAMPLING_RATE; if (config->sample_rate != 48000 && config->sample_rate != 16000 && config->sample_rate != 8000) { config->sample_rate = AFE_PROXY_SAMPLING_RATE; ret = -EINVAL; goto err_open; } if (config->format == AUDIO_FORMAT_DEFAULT) config->format = AUDIO_FORMAT_PCM_16_BIT; if (config->format != AUDIO_FORMAT_PCM_16_BIT) { config->format = AUDIO_FORMAT_PCM_16_BIT; ret = -EINVAL; goto err_open; } in->usecase = USECASE_AUDIO_RECORD_AFE_PROXY; in->config = pcm_config_afe_proxy_record; in->config.channels = channel_count; in->config.rate = config->sample_rate; in->sample_rate = config->sample_rate; } else if (!audio_extn_check_and_set_multichannel_usecase(adev, in, config, &channel_mask_updated)) { if (channel_mask_updated == true) { ALOGD("%s: return error to retry with updated channel mask (%#x)", __func__, config->channel_mask); ret = -EINVAL; goto err_open; } ALOGD("%s: created surround sound session succesfully",__func__); } else if (audio_extn_compr_cap_enabled() && audio_extn_compr_cap_format_supported(config->format) && (in->dev->mode != AUDIO_MODE_IN_COMMUNICATION)) { audio_extn_compr_cap_init(in); } else if (audio_extn_cin_applicable_stream(in)) { ret = audio_extn_cin_configure_input_stream(in); if (ret) goto err_open; } else { in->config.channels = channel_count; if (!in->realtime) { in->format = config->format; frame_size = audio_stream_in_frame_size(&in->stream); buffer_size = get_input_buffer_size(config->sample_rate, config->format, channel_count, is_low_latency); in->config.period_size = buffer_size / frame_size; } if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) && (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION || voice_extn_compress_voip_is_active(in->dev)) && (voice_extn_compress_voip_is_format_supported(in->format)) && (in->config.rate == 8000 || in->config.rate == 16000 || in->config.rate == 32000 || in->config.rate == 48000) && (audio_channel_count_from_in_mask(in->channel_mask) == 1)) { voice_extn_compress_voip_open_input_stream(in); } } audio_extn_utils_update_stream_input_app_type_cfg(adev->platform, &adev->streams_input_cfg_list, devices, flags, in->format, in->sample_rate, in->bit_width, in->profile, &in->app_type_cfg); /* This stream could be for sound trigger lab, get sound trigger pcm if present */ audio_extn_sound_trigger_check_and_get_session(in); *stream_in = &in->stream; ALOGV("%s: exit", __func__); return ret; err_open: free(in); *stream_in = NULL; return ret; } static void adev_close_input_stream(struct audio_hw_device *dev, struct audio_stream_in *stream) { int ret; struct stream_in *in = (struct stream_in *)stream; struct audio_device *adev = (struct audio_device *)dev; ALOGD("%s: enter:stream_handle(%p)",__func__, in); /* Disable echo reference while closing input stream */ platform_set_echo_reference(adev, false, AUDIO_DEVICE_NONE); if (in->usecase == USECASE_COMPRESS_VOIP_CALL) { pthread_mutex_lock(&adev->lock); ret = voice_extn_compress_voip_close_input_stream(&stream->common); pthread_mutex_unlock(&adev->lock); if (ret != 0) ALOGE("%s: Compress voip input cannot be closed, error:%d", __func__, ret); } else in_standby(&stream->common); if (audio_extn_ssr_get_stream() == in) { audio_extn_ssr_deinit(); } if (audio_extn_compr_cap_enabled() && audio_extn_compr_cap_format_supported(in->config.format)) audio_extn_compr_cap_deinit(); if (audio_extn_cin_attached_usecase(in->usecase)) audio_extn_cin_close_input_stream(in); if (in->is_st_session) { ALOGV("%s: sound trigger pcm stop lab", __func__); audio_extn_sound_trigger_stop_lab(in); } free(stream); return; } int adev_create_audio_patch(struct audio_hw_device *dev, unsigned int num_sources, const struct audio_port_config *sources, unsigned int num_sinks, const struct audio_port_config *sinks, audio_patch_handle_t *handle) { return audio_extn_hw_loopback_create_audio_patch(dev, num_sources, sources, num_sinks, sinks, handle); } int adev_release_audio_patch(struct audio_hw_device *dev, audio_patch_handle_t handle) { return audio_extn_hw_loopback_release_audio_patch(dev, handle); } int adev_get_audio_port(struct audio_hw_device *dev, struct audio_port *config) { return audio_extn_hw_loopback_get_audio_port(dev, config); } int adev_set_audio_port_config(struct audio_hw_device *dev, const struct audio_port_config *config) { return audio_extn_hw_loopback_set_audio_port_config(dev, config); } static int adev_dump(const audio_hw_device_t *device __unused, int fd __unused) { return 0; } static int adev_close(hw_device_t *device) { struct audio_device *adev = (struct audio_device *)device; if (!adev) return 0; pthread_mutex_lock(&adev_init_lock); if ((--audio_device_ref_count) == 0) { audio_extn_sound_trigger_deinit(adev); audio_extn_listen_deinit(adev); audio_extn_utils_release_streams_cfg_lists( &adev->streams_output_cfg_list, &adev->streams_input_cfg_list); if (audio_extn_qaf_is_enabled()) audio_extn_qaf_deinit(); audio_route_free(adev->audio_route); audio_extn_gef_deinit(); free(adev->snd_dev_ref_cnt); platform_deinit(adev->platform); if (adev->adm_deinit) adev->adm_deinit(adev->adm_data); qahwi_deinit(device); audio_extn_adsp_hdlr_deinit(); audio_extn_hw_loopback_deinit(adev); if (adev->device_cfg_params) { free(adev->device_cfg_params); adev->device_cfg_params = NULL; } free(device); adev = NULL; } pthread_mutex_unlock(&adev_init_lock); return 0; } /* This returns 1 if the input parameter looks at all plausible as a low latency period size, * or 0 otherwise. A return value of 1 doesn't mean the value is guaranteed to work, * just that it _might_ work. */ static int period_size_is_plausible_for_low_latency(int period_size) { switch (period_size) { case 160: case 192: case 240: case 320: case 480: return 1; default: return 0; } } static int adev_open(const hw_module_t *module, const char *name, hw_device_t **device) { int ret, i = 0; 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.create_audio_patch = adev_create_audio_patch; adev->device.release_audio_patch = adev_release_audio_patch; adev->device.get_audio_port = adev_get_audio_port; adev->device.set_audio_port_config = adev_set_audio_port_config; adev->device.dump = adev_dump; /* Set the default route before the PCM stream is opened */ adev->mode = AUDIO_MODE_NORMAL; adev->active_input = NULL; adev->primary_output = NULL; adev->out_device = AUDIO_DEVICE_NONE; adev->bluetooth_nrec = true; adev->acdb_settings = TTY_MODE_OFF; adev->allow_afe_proxy_usage = true; adev->bt_sco_on = false; /* adev->cur_hdmi_channels = 0; by calloc() */ adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int)); voice_init(adev); list_init(&adev->usecase_list); adev->cur_wfd_channels = 2; adev->offload_usecases_state = 0; adev->is_channel_status_set = false; adev->perf_lock_opts[0] = 0x101; adev->perf_lock_opts[1] = 0x20E; adev->perf_lock_opts_size = 2; pthread_mutex_init(&adev->snd_card_status.lock, (const pthread_mutexattr_t *) NULL); adev->snd_card_status.state = SND_CARD_STATE_OFFLINE; /* Loads platform specific libraries dynamically */ adev->platform = platform_init(adev); if (!adev->platform) { free(adev->snd_dev_ref_cnt); free(adev); ALOGE("%s: Failed to init platform data, aborting.", __func__); *device = NULL; pthread_mutex_unlock(&adev_init_lock); pthread_mutex_destroy(&adev->lock); pthread_mutex_destroy(&adev->snd_card_status.lock); return -EINVAL; } if (audio_extn_qaf_is_enabled()) { ret = audio_extn_qaf_init(adev); if (ret < 0) { free(adev); ALOGE("%s: Failed to init platform data, aborting.", __func__); *device = NULL; pthread_mutex_unlock(&adev_init_lock); pthread_mutex_destroy(&adev->lock); return ret; } adev->device.open_output_stream = audio_extn_qaf_open_output_stream; adev->device.close_output_stream = audio_extn_qaf_close_output_stream; } adev->snd_card_status.state = SND_CARD_STATE_ONLINE; if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) { adev->visualizer_lib = dlopen(VISUALIZER_LIBRARY_PATH, RTLD_NOW); if (adev->visualizer_lib == NULL) { ALOGE("%s: DLOPEN failed for %s", __func__, VISUALIZER_LIBRARY_PATH); } else { ALOGV("%s: DLOPEN successful for %s", __func__, VISUALIZER_LIBRARY_PATH); adev->visualizer_start_output = (int (*)(audio_io_handle_t, int))dlsym(adev->visualizer_lib, "visualizer_hal_start_output"); adev->visualizer_stop_output = (int (*)(audio_io_handle_t, int))dlsym(adev->visualizer_lib, "visualizer_hal_stop_output"); } } audio_extn_init(adev); audio_extn_listen_init(adev, adev->snd_card); audio_extn_sound_trigger_init(adev); audio_extn_gef_init(adev); audio_extn_hw_loopback_init(adev); if (access(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, R_OK) == 0) { adev->offload_effects_lib = dlopen(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, RTLD_NOW); if (adev->offload_effects_lib == NULL) { ALOGE("%s: DLOPEN failed for %s", __func__, OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH); } else { ALOGV("%s: DLOPEN successful for %s", __func__, OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH); adev->offload_effects_start_output = (int (*)(audio_io_handle_t, int, struct mixer *))dlsym(adev->offload_effects_lib, "offload_effects_bundle_hal_start_output"); adev->offload_effects_stop_output = (int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib, "offload_effects_bundle_hal_stop_output"); adev->offload_effects_set_hpx_state = (int (*)(bool))dlsym(adev->offload_effects_lib, "offload_effects_bundle_set_hpx_state"); adev->offload_effects_get_parameters = (void (*)(struct str_parms *, struct str_parms *)) dlsym(adev->offload_effects_lib, "offload_effects_bundle_get_parameters"); adev->offload_effects_set_parameters = (void (*)(struct str_parms *))dlsym(adev->offload_effects_lib, "offload_effects_bundle_set_parameters"); } } if (access(ADM_LIBRARY_PATH, R_OK) == 0) { adev->adm_lib = dlopen(ADM_LIBRARY_PATH, RTLD_NOW); if (adev->adm_lib == NULL) { ALOGE("%s: DLOPEN failed for %s", __func__, ADM_LIBRARY_PATH); } else { ALOGV("%s: DLOPEN successful for %s", __func__, ADM_LIBRARY_PATH); adev->adm_init = (adm_init_t) dlsym(adev->adm_lib, "adm_init"); adev->adm_deinit = (adm_deinit_t) dlsym(adev->adm_lib, "adm_deinit"); adev->adm_register_input_stream = (adm_register_input_stream_t) dlsym(adev->adm_lib, "adm_register_input_stream"); adev->adm_register_output_stream = (adm_register_output_stream_t) dlsym(adev->adm_lib, "adm_register_output_stream"); adev->adm_deregister_stream = (adm_deregister_stream_t) dlsym(adev->adm_lib, "adm_deregister_stream"); adev->adm_request_focus = (adm_request_focus_t) dlsym(adev->adm_lib, "adm_request_focus"); adev->adm_abandon_focus = (adm_abandon_focus_t) dlsym(adev->adm_lib, "adm_abandon_focus"); adev->adm_set_config = (adm_set_config_t) dlsym(adev->adm_lib, "adm_set_config"); adev->adm_request_focus_v2 = (adm_request_focus_v2_t) dlsym(adev->adm_lib, "adm_request_focus_v2"); adev->adm_is_noirq_avail = (adm_is_noirq_avail_t) dlsym(adev->adm_lib, "adm_is_noirq_avail"); adev->adm_on_routing_change = (adm_on_routing_change_t) dlsym(adev->adm_lib, "adm_on_routing_change"); } } adev->bt_wb_speech_enabled = false; //initialize this to false for now, //this will be set to true through set param adev->vr_audio_mode_enabled = false; audio_extn_ds2_enable(adev); *device = &adev->device.common; audio_extn_utils_update_streams_cfg_lists(adev->platform, adev->mixer, &adev->streams_output_cfg_list, &adev->streams_input_cfg_list); audio_device_ref_count++; char value[PROPERTY_VALUE_MAX]; int trial; if (property_get("audio_hal.period_size", value, NULL) > 0) { trial = atoi(value); if (period_size_is_plausible_for_low_latency(trial)) { pcm_config_low_latency.period_size = trial; pcm_config_low_latency.start_threshold = trial / 4; pcm_config_low_latency.avail_min = trial / 4; configured_low_latency_capture_period_size = trial; } } if (property_get("audio_hal.in_period_size", value, NULL) > 0) { trial = atoi(value); if (period_size_is_plausible_for_low_latency(trial)) { configured_low_latency_capture_period_size = trial; } } if (property_get("audio_hal.period_multiplier", value, NULL) > 0) { af_period_multiplier = atoi(value); if (af_period_multiplier < 0) af_period_multiplier = 2; else if (af_period_multiplier > 4) af_period_multiplier = 4; ALOGV("new period_multiplier = %d", af_period_multiplier); } adev->multi_offload_enable = property_get_bool("audio.offload.multiple.enabled", false); pthread_mutex_unlock(&adev_init_lock); if (adev->adm_init) adev->adm_data = adev->adm_init(); qahwi_init(*device); audio_extn_perf_lock_init(); audio_extn_adsp_hdlr_init(adev->mixer); /* Allocate memory for Device config params */ adev->device_cfg_params = (struct audio_device_config_param*) calloc(platform_get_max_codec_backend(), sizeof(struct audio_device_config_param)); if (adev->device_cfg_params == NULL) ALOGE("%s: Memory allocation failed for Device config params", __func__); 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, }, };