| /* |
| * Copyright (c) 2013-2020, The Linux Foundation. All rights reserved. |
| * Not a contribution. |
| * |
| * Copyright (C) 2013 The Android Open Source Project |
| * |
| * Licensed under the Apache License, Version 2.0 (the "License"); |
| * you may not use this file except in compliance with the License. |
| * You may obtain a copy of the License at |
| * |
| * http://www.apache.org/licenses/LICENSE-2.0 |
| * |
| * Unless required by applicable law or agreed to in writing, software |
| * distributed under the License is distributed on an "AS IS" BASIS, |
| * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. |
| * See the License for the specific language governing permissions and |
| * limitations under the License. |
| */ |
| |
| #define LOG_TAG "voice" |
| /*#define LOG_NDEBUG 0*/ |
| #define LOG_NDDEBUG 0 |
| |
| #include <errno.h> |
| #include <stdlib.h> |
| #include <math.h> |
| #include <log/log.h> |
| #include <cutils/str_parms.h> |
| |
| #include "audio_hw.h" |
| #include "voice.h" |
| #include "voice_extn/voice_extn.h" |
| #include "platform.h" |
| #include "platform_api.h" |
| #include "audio_extn.h" |
| |
| #ifdef DYNAMIC_LOG_ENABLED |
| #include <log_xml_parser.h> |
| #define LOG_MASK HAL_MOD_FILE_VOICE |
| #include <log_utils.h> |
| #endif |
| |
| struct pcm_config pcm_config_voice_call = { |
| .channels = 1, |
| .rate = 8000, |
| .period_size = 160, |
| .period_count = 2, |
| .format = PCM_FORMAT_S16_LE, |
| }; |
| |
| #ifdef PLATFORM_AUTO |
| struct pcm *voice_loopback_tx = NULL; |
| struct pcm *voice_loopback_rx = NULL; |
| #endif |
| static struct voice_session *voice_get_session_from_use_case(struct audio_device *adev, |
| audio_usecase_t usecase_id) |
| { |
| struct voice_session *session = NULL; |
| int ret = 0; |
| |
| ret = voice_extn_get_session_from_use_case(adev, usecase_id, &session); |
| if (ret == -ENOSYS) { |
| session = &adev->voice.session[VOICE_SESS_IDX]; |
| } |
| |
| return session; |
| } |
| |
| static bool voice_is_sidetone_device(snd_device_t out_device, |
| char *mixer_path) |
| { |
| bool is_sidetone_dev; |
| |
| switch (out_device) { |
| case SND_DEVICE_OUT_VOICE_HANDSET: |
| is_sidetone_dev = true; |
| strlcpy(mixer_path, "sidetone-handset", MIXER_PATH_MAX_LENGTH); |
| break; |
| case SND_DEVICE_OUT_VOICE_HEADPHONES: |
| case SND_DEVICE_OUT_VOICE_HEADSET: |
| case SND_DEVICE_OUT_VOICE_ANC_HEADSET: |
| case SND_DEVICE_OUT_VOICE_ANC_FB_HEADSET: |
| is_sidetone_dev = true; |
| strlcpy(mixer_path, "sidetone-headphones", MIXER_PATH_MAX_LENGTH); |
| break; |
| case SND_DEVICE_OUT_VOICE_USB_HEADSET: |
| case SND_DEVICE_OUT_USB_HEADSET: |
| // USB does not use a QC mixer. |
| mixer_path[0] = '\0'; |
| is_sidetone_dev = true; |
| break; |
| default: |
| ALOGW("%s: %d is not a sidetone device", __func__, out_device); |
| is_sidetone_dev = false; |
| break; |
| } |
| |
| return is_sidetone_dev; |
| } |
| |
| void voice_set_sidetone(struct audio_device *adev, |
| snd_device_t out_snd_device, bool enable) |
| { |
| char mixer_path[MIXER_PATH_MAX_LENGTH]; |
| ALOGD("%s: %s, out_snd_device: %d\n", |
| __func__, (enable ? "enable" : "disable"), |
| out_snd_device); |
| if (voice_is_sidetone_device(out_snd_device, mixer_path)) |
| platform_set_sidetone(adev, out_snd_device, enable, mixer_path); |
| return; |
| } |
| |
| static bool voice_is_aanc_device(snd_device_t out_device, |
| char *mixer_path) |
| { |
| bool is_aanc_dev; |
| |
| switch (out_device) { |
| case SND_DEVICE_OUT_ANC_HANDSET: |
| is_aanc_dev = true; |
| strlcpy(mixer_path, "aanc-path", MIXER_PATH_MAX_LENGTH); |
| break; |
| default: |
| is_aanc_dev = false; |
| break; |
| } |
| |
| return is_aanc_dev; |
| } |
| |
| void voice_check_and_update_aanc_path(struct audio_device *adev, |
| snd_device_t out_snd_device, |
| bool enable) |
| { |
| char mixer_path[MIXER_PATH_MAX_LENGTH]; |
| |
| ALOGV("%s: %s, out_snd_device: %d\n", |
| __func__, (enable ? "enable" : "disable"), out_snd_device); |
| |
| if (voice_is_aanc_device(out_snd_device, mixer_path)) |
| platform_update_aanc_path(adev, out_snd_device, enable, mixer_path); |
| |
| return; |
| } |
| |
| int voice_stop_usecase(struct audio_device *adev, audio_usecase_t usecase_id) |
| { |
| int ret = 0; |
| struct audio_usecase *uc_info; |
| struct voice_session *session = NULL; |
| |
| ALOGD("%s: enter usecase:%s", __func__, use_case_table[usecase_id]); |
| |
| session = (struct voice_session *)voice_get_session_from_use_case(adev, usecase_id); |
| if (!session) { |
| ALOGE("stop_call: couldn't find voice session"); |
| return -EINVAL; |
| } |
| |
| uc_info = get_usecase_from_list(adev, usecase_id); |
| if (uc_info == NULL) { |
| ALOGE("%s: Could not find the usecase (%d) in the list", |
| __func__, usecase_id); |
| return -EINVAL; |
| } |
| |
| session->state.current = CALL_INACTIVE; |
| |
| /* Disable sidetone only when no calls are active */ |
| if (!voice_is_call_state_active_in_call(adev)) |
| voice_set_sidetone(adev, uc_info->out_snd_device, false); |
| |
| /* Disable aanc only when no calls are active */ |
| if (!voice_is_call_state_active_in_call(adev)) |
| voice_check_and_update_aanc_path(adev, uc_info->out_snd_device, false); |
| |
| ret = platform_stop_voice_call(adev->platform, session->vsid); |
| |
| /* 1. Close the PCM devices */ |
| if (session->pcm_rx) { |
| pcm_close(session->pcm_rx); |
| session->pcm_rx = NULL; |
| } |
| if (session->pcm_tx) { |
| pcm_close(session->pcm_tx); |
| session->pcm_tx = NULL; |
| } |
| |
| #ifdef PLATFORM_AUTO |
| if(voice_loopback_rx) { |
| pcm_close(voice_loopback_rx); |
| voice_loopback_rx = NULL; |
| } |
| if(voice_loopback_tx) { |
| pcm_close(voice_loopback_tx); |
| voice_loopback_tx = NULL; |
| } |
| #endif |
| /* 2. Get and set stream specific mixer controls */ |
| disable_audio_route(adev, uc_info); |
| |
| /* 3. Disable the rx and tx devices */ |
| disable_snd_device(adev, uc_info->out_snd_device); |
| disable_snd_device(adev, uc_info->in_snd_device); |
| |
| list_remove(&uc_info->list); |
| free(uc_info); |
| |
| ALOGD("%s: exit: status(%d)", __func__, ret); |
| return ret; |
| } |
| |
| int voice_start_usecase(struct audio_device *adev, audio_usecase_t usecase_id) |
| { |
| int ret = 0; |
| struct audio_usecase *uc_info; |
| int pcm_dev_rx_id, pcm_dev_tx_id; |
| #ifdef PLATFORM_AUTO |
| int pcm_dev_loopback_rx_id, pcm_dev_loopback_tx_id; |
| #endif |
| uint32_t sample_rate = 8000; |
| struct voice_session *session = NULL; |
| struct pcm_config voice_config = pcm_config_voice_call; |
| bool is_in_call = (AUDIO_MODE_IN_CALL == adev->mode); |
| |
| ALOGD("%s: enter usecase:%s", __func__, use_case_table[usecase_id]); |
| |
| session = (struct voice_session *)voice_get_session_from_use_case(adev, usecase_id); |
| if (!session) { |
| ALOGE("start_call: couldn't find voice session"); |
| return -EINVAL; |
| } |
| |
| if (!adev->current_call_output) { |
| ALOGE("start_call: invalid current call output"); |
| return -EINVAL; |
| } |
| |
| uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase)); |
| if (!uc_info) { |
| ALOGE("start_call: couldn't allocate mem for audio_usecase"); |
| return -ENOMEM; |
| } |
| |
| uc_info->id = usecase_id; |
| uc_info->type = VOICE_CALL; |
| uc_info->stream.out = adev->current_call_output; |
| list_init(&uc_info->device_list); |
| assign_devices(&uc_info->device_list, &adev->current_call_output->device_list); |
| |
| if (is_in_call && list_length(&uc_info->device_list) == 2) { |
| ALOGE("%s: Invalid combo device(%#x) for voice call", __func__, |
| get_device_types(&uc_info->device_list)); |
| ret = -EIO; |
| goto error_start_voice; |
| } |
| |
| uc_info->in_snd_device = SND_DEVICE_NONE; |
| uc_info->out_snd_device = SND_DEVICE_NONE; |
| adev->voice.use_device_mute = false; |
| |
| if (is_sco_out_device_type(&uc_info->device_list) && !adev->bt_sco_on) { |
| ALOGE("start_call: couldn't find BT SCO, SCO is not ready"); |
| adev->voice.in_call = false; |
| ret = -EIO; |
| goto error_start_voice; |
| } |
| |
| list_add_tail(&adev->usecase_list, &uc_info->list); |
| |
| select_devices(adev, usecase_id); |
| |
| #ifdef PLATFORM_AUTO |
| pcm_dev_loopback_rx_id = HOST_LESS_RX_ID; |
| pcm_dev_loopback_tx_id = HOST_LESS_TX_ID; |
| #endif |
| pcm_dev_rx_id = platform_get_pcm_device_id(uc_info->id, PCM_PLAYBACK); |
| pcm_dev_tx_id = platform_get_pcm_device_id(uc_info->id, PCM_CAPTURE); |
| |
| if (pcm_dev_rx_id < 0 || pcm_dev_tx_id < 0) { |
| ALOGE("%s: Invalid PCM devices (rx: %d tx: %d) for the usecase(%d)", |
| __func__, pcm_dev_rx_id, pcm_dev_tx_id, uc_info->id); |
| ret = -EIO; |
| goto error_start_voice; |
| } |
| ret = platform_get_sample_rate(adev->platform, &sample_rate); |
| if (ret < 0) { |
| ALOGE("platform_get_sample_rate error %d\n", ret); |
| } else { |
| voice_config.rate = sample_rate; |
| } |
| ALOGD("voice_config.rate %d\n", voice_config.rate); |
| |
| voice_set_mic_mute(adev, adev->voice.mic_mute); |
| |
| ALOGV("%s: Opening PCM capture device card_id(%d) device_id(%d)", |
| __func__, adev->snd_card, pcm_dev_tx_id); |
| session->pcm_tx = pcm_open(adev->snd_card, |
| pcm_dev_tx_id, |
| PCM_IN, &voice_config); |
| if (session->pcm_tx && !pcm_is_ready(session->pcm_tx)) { |
| ALOGE("%s: %s", __func__, pcm_get_error(session->pcm_tx)); |
| ret = -EIO; |
| goto error_start_voice; |
| } |
| |
| ALOGV("%s: Opening PCM playback device card_id(%d) device_id(%d)", |
| __func__, adev->snd_card, pcm_dev_rx_id); |
| session->pcm_rx = pcm_open(adev->snd_card, |
| pcm_dev_rx_id, |
| PCM_OUT, &voice_config); |
| if (session->pcm_rx && !pcm_is_ready(session->pcm_rx)) { |
| ALOGE("%s: %s", __func__, pcm_get_error(session->pcm_rx)); |
| ret = -EIO; |
| goto error_start_voice; |
| } |
| |
| #ifdef PLATFORM_AUTO |
| voice_loopback_rx = pcm_open(adev->snd_card, |
| pcm_dev_loopback_rx_id, |
| PCM_OUT, &voice_config); |
| if (voice_loopback_rx < 0 || !pcm_is_ready(voice_loopback_rx)) { |
| ALOGE("%s: Either could not open pcm_dev_loopback_rx_id %d or %s", |
| __func__, pcm_dev_loopback_rx_id, pcm_get_error(voice_loopback_rx)); |
| ret = -EIO; |
| goto error_start_voice; |
| } |
| |
| voice_loopback_tx = pcm_open(adev->snd_card, |
| pcm_dev_loopback_tx_id, |
| PCM_IN, &voice_config); |
| if (voice_loopback_tx < 0 || !pcm_is_ready(voice_loopback_tx)) { |
| ALOGE("%s: Either could not open pcm_dev_loopback_tx_id %d or %s", |
| __func__, pcm_dev_loopback_tx_id, pcm_get_error(voice_loopback_tx)); |
| ret = -EIO; |
| goto error_start_voice; |
| } |
| #endif |
| |
| if (adev->mic_break_enabled) |
| platform_set_mic_break_det(adev->platform, true); |
| |
| ret = pcm_start(session->pcm_tx); |
| if (ret != 0) { |
| ALOGE("%s: %s", __func__, pcm_get_error(session->pcm_tx)); |
| goto error_start_voice; |
| } |
| |
| ret = pcm_start(session->pcm_rx); |
| if (ret != 0) { |
| ALOGE("%s: %s", __func__, pcm_get_error(session->pcm_rx)); |
| goto error_start_voice; |
| } |
| |
| #ifdef PLATFORM_AUTO |
| ret = pcm_start(voice_loopback_tx); |
| if (ret != 0) { |
| ALOGE("%s: %s", __func__, pcm_get_error(voice_loopback_tx)); |
| goto error_start_voice; |
| } |
| |
| ret = pcm_start(voice_loopback_rx); |
| if (ret != 0) { |
| ALOGE("%s: %s", __func__, pcm_get_error(voice_loopback_rx)); |
| goto error_start_voice; |
| } |
| #endif |
| |
| /* Enable aanc only when no calls are active */ |
| if (!voice_is_call_state_active_in_call(adev)) |
| voice_check_and_update_aanc_path(adev, uc_info->out_snd_device, true); |
| |
| /* Enable sidetone only when no calls are already active */ |
| if (!voice_is_call_state_active_in_call(adev)) |
| voice_set_sidetone(adev, uc_info->out_snd_device, true); |
| |
| voice_set_volume(adev, adev->voice.volume); |
| |
| ret = platform_start_voice_call(adev->platform, session->vsid); |
| if (ret < 0) { |
| ALOGE("%s: platform_start_voice_call error %d\n", __func__, ret); |
| goto error_start_voice; |
| } |
| |
| session->state.current = CALL_ACTIVE; |
| goto done; |
| |
| error_start_voice: |
| voice_stop_usecase(adev, usecase_id); |
| |
| done: |
| ALOGD("%s: exit: status(%d)", __func__, ret); |
| return ret; |
| } |
| |
| /* |
| * helper function to check whether call is active or not. |
| */ |
| static inline bool voice_is_active(struct audio_device *adev) { |
| bool call_state = false; |
| int ret = 0; |
| |
| ret = voice_extn_is_call_state_active(adev, &call_state); |
| if (ret == -ENOSYS) { |
| call_state = (adev->voice.session[VOICE_SESS_IDX].state.current == CALL_ACTIVE) ? true : false; |
| } |
| |
| return call_state; |
| } |
| |
| /* |
| * checks if call is active and in IN_CALL mode. |
| */ |
| bool voice_is_call_state_active_in_call(struct audio_device *adev) |
| { |
| bool call_state = voice_is_active(adev); |
| return call_state && adev->mode == AUDIO_MODE_IN_CALL; |
| } |
| |
| /* |
| * returns true if call is active no matter what mode is. |
| */ |
| bool voice_is_call_state_active(struct audio_device *adev) |
| { |
| return voice_is_active(adev); |
| } |
| |
| bool voice_is_in_call(const struct audio_device *adev) |
| { |
| return adev->voice.in_call && adev->mode == AUDIO_MODE_IN_CALL; |
| } |
| |
| bool voice_is_in_call_or_call_screen(const struct audio_device *adev) |
| { |
| return adev->voice.in_call; |
| } |
| |
| bool voice_is_in_call_rec_stream(const struct stream_in *in) |
| { |
| bool in_call_rec = false; |
| |
| if (!in) { |
| ALOGE("%s: input stream is NULL", __func__); |
| return in_call_rec; |
| } |
| |
| if (in->source == AUDIO_SOURCE_VOICE_DOWNLINK || |
| in->source == AUDIO_SOURCE_VOICE_UPLINK || |
| in->source == AUDIO_SOURCE_VOICE_CALL) { |
| in_call_rec = true; |
| } |
| |
| return in_call_rec; |
| } |
| |
| uint32_t voice_get_active_session_id(struct audio_device *adev) |
| { |
| int ret = 0; |
| uint32_t session_id; |
| |
| ret = voice_extn_get_active_session_id(adev, &session_id); |
| if (ret == -ENOSYS) { |
| session_id = VOICE_VSID; |
| } |
| return session_id; |
| } |
| |
| bool voice_check_voicecall_usecases_active(struct audio_device *adev) |
| { |
| struct listnode *node; |
| struct audio_usecase *usecase = NULL; |
| |
| list_for_each(node, &adev->usecase_list) { |
| usecase = node_to_item(node, struct audio_usecase, list); |
| if (usecase->type == VOICE_CALL && adev->mode != AUDIO_MODE_CALL_SCREEN) { |
| ALOGV("%s: voice usecase:%s is active", __func__, |
| use_case_table[usecase->id]); |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| int voice_check_and_set_incall_rec_usecase(struct audio_device *adev, |
| struct stream_in *in) |
| { |
| int ret = 0; |
| uint32_t session_id; |
| int rec_mode = INCALL_REC_NONE; |
| |
| if (voice_is_call_state_active(adev)) { |
| switch (in->source) { |
| case AUDIO_SOURCE_VOICE_UPLINK: |
| if (audio_extn_compr_cap_enabled() && |
| audio_extn_compr_cap_format_supported(in->config.format)) { |
| in->usecase = USECASE_INCALL_REC_UPLINK_COMPRESS; |
| } else |
| in->usecase = USECASE_INCALL_REC_UPLINK; |
| rec_mode = INCALL_REC_UPLINK; |
| break; |
| case AUDIO_SOURCE_VOICE_DOWNLINK: |
| if (audio_extn_compr_cap_enabled() && |
| audio_extn_compr_cap_format_supported(in->config.format)) { |
| in->usecase = USECASE_INCALL_REC_DOWNLINK_COMPRESS; |
| } else |
| in->usecase = USECASE_INCALL_REC_DOWNLINK; |
| rec_mode = INCALL_REC_DOWNLINK; |
| break; |
| case AUDIO_SOURCE_VOICE_CALL: |
| if (audio_extn_compr_cap_enabled() && |
| audio_extn_compr_cap_format_supported(in->config.format)) { |
| in->usecase = USECASE_INCALL_REC_UPLINK_AND_DOWNLINK_COMPRESS; |
| } else |
| in->usecase = USECASE_INCALL_REC_UPLINK_AND_DOWNLINK; |
| rec_mode = INCALL_REC_UPLINK_AND_DOWNLINK; |
| break; |
| default: |
| ALOGV("%s: Source type %d doesnt match incall recording criteria", |
| __func__, in->source); |
| return ret; |
| } |
| |
| session_id = voice_get_active_session_id(adev); |
| ret = platform_set_incall_recording_session_id(adev->platform, |
| session_id, rec_mode); |
| platform_set_incall_recording_session_channels(adev->platform, |
| in->config.channels); |
| ALOGV("%s: Update usecase to %d",__func__, in->usecase); |
| } else { |
| /* |
| * Reject the recording instances, where the recording is started |
| * with In-call voice recording source types but voice call is not |
| * active by the time input is started |
| */ |
| if ((in->source == AUDIO_SOURCE_VOICE_UPLINK) || |
| (in->source == AUDIO_SOURCE_VOICE_DOWNLINK) || |
| (in->source == AUDIO_SOURCE_VOICE_CALL)) { |
| ret = -EINVAL; |
| ALOGE("%s: As voice call is not active, Incall rec usecase can't be \ |
| selected for requested source:%d",__func__, in->source); |
| } |
| ALOGV("%s: voice call not active", __func__); |
| } |
| |
| return ret; |
| } |
| |
| int voice_check_and_stop_incall_rec_usecase(struct audio_device *adev, |
| struct stream_in *in) |
| { |
| int ret = 0; |
| |
| if (in->source == AUDIO_SOURCE_VOICE_UPLINK || |
| in->source == AUDIO_SOURCE_VOICE_DOWNLINK || |
| in->source == AUDIO_SOURCE_VOICE_CALL) { |
| ret = platform_stop_incall_recording_usecase(adev->platform); |
| ALOGV("%s: Stop In-call recording", __func__); |
| } |
| |
| return ret; |
| } |
| |
| snd_device_t voice_get_incall_rec_backend_device(struct stream_in *in) |
| { |
| snd_device_t incall_record_device = {0}; |
| |
| if (!in) { |
| ALOGE("%s: input stream is NULL", __func__); |
| return 0; |
| } |
| |
| switch(in->source) { |
| case AUDIO_SOURCE_VOICE_UPLINK: |
| incall_record_device = SND_DEVICE_IN_INCALL_REC_TX; |
| break; |
| case AUDIO_SOURCE_VOICE_DOWNLINK: |
| incall_record_device = SND_DEVICE_IN_INCALL_REC_RX; |
| break; |
| case AUDIO_SOURCE_VOICE_CALL: |
| incall_record_device = SND_DEVICE_IN_INCALL_REC_RX_TX; |
| break; |
| default: |
| ALOGI("Invalid source %d", in->source); |
| } |
| |
| return incall_record_device; |
| } |
| |
| snd_device_t voice_get_incall_rec_snd_device(snd_device_t in_snd_device) |
| { |
| snd_device_t incall_record_device = in_snd_device; |
| |
| /* |
| * For incall recording stream, AUDIO_COPP topology will be picked up |
| * from the calibration data of the input sound device which is nothing |
| * but the voice call's input device. But there are requirements to use |
| * AUDIO_COPP_MONO topology even if the voice call's input device is |
| * different. Hence override the input device with the one which uses |
| * the AUDIO_COPP_MONO topology. |
| */ |
| switch(in_snd_device) { |
| case SND_DEVICE_IN_HANDSET_MIC: |
| case SND_DEVICE_IN_VOICE_DMIC: |
| case SND_DEVICE_IN_AANC_HANDSET_MIC: |
| incall_record_device = SND_DEVICE_IN_HANDSET_MIC; |
| break; |
| case SND_DEVICE_IN_VOICE_SPEAKER_MIC: |
| case SND_DEVICE_IN_VOICE_SPEAKER_DMIC: |
| case SND_DEVICE_IN_VOICE_SPEAKER_DMIC_BROADSIDE: |
| case SND_DEVICE_IN_VOICE_SPEAKER_QMIC: |
| incall_record_device = SND_DEVICE_IN_VOICE_SPEAKER_MIC; |
| break; |
| default: |
| incall_record_device = in_snd_device; |
| } |
| |
| ALOGD("%s: in_snd_device(%d: %s) incall_record_device(%d: %s)", __func__, |
| in_snd_device, platform_get_snd_device_name(in_snd_device), |
| incall_record_device, platform_get_snd_device_name(incall_record_device)); |
| |
| return incall_record_device; |
| } |
| |
| int voice_set_mic_mute(struct audio_device *adev, bool state) |
| { |
| int err = 0; |
| |
| adev->voice.mic_mute = state; |
| |
| if (audio_extn_hfp_is_active(adev)) { |
| err = audio_extn_hfp_set_mic_mute2(adev, state); |
| } else if (adev->mode == AUDIO_MODE_IN_CALL) { |
| /* Use device mute if incall music delivery usecase is in progress */ |
| if (adev->voice.use_device_mute) |
| err = platform_set_device_mute(adev->platform, state, "tx"); |
| else |
| err = platform_set_mic_mute(adev->platform, state); |
| ALOGV("%s: voice mute status=%d, use_device_mute flag=%d", |
| __func__, state, adev->voice.use_device_mute); |
| } else if (adev->mode == AUDIO_MODE_IN_COMMUNICATION) { |
| err = voice_extn_compress_voip_set_mic_mute(adev, state); |
| } |
| |
| return err; |
| } |
| |
| bool voice_get_mic_mute(struct audio_device *adev) |
| { |
| return adev->voice.mic_mute; |
| } |
| |
| /* |
| * Following function is called when incall music uplink usecase is |
| * created or destroyed while mic is muted. If incall music uplink |
| * usecase is active, apply voice device mute to mute only voice Tx |
| * path and not the mixed voice Tx + inncall-music path. Revert to |
| * voice stream mute once incall music uplink usecase is inactive |
| */ |
| void voice_set_device_mute_flag(struct audio_device *adev, bool state) |
| { |
| if (adev->voice.mic_mute) { |
| if (state) { |
| platform_set_device_mute(adev->platform, true, "tx"); |
| platform_set_mic_mute(adev->platform, false); |
| } else { |
| platform_set_mic_mute(adev->platform, true); |
| platform_set_device_mute(adev->platform, false, "tx"); |
| } |
| } |
| adev->voice.use_device_mute = state; |
| } |
| |
| int voice_set_volume(struct audio_device *adev, float volume) |
| { |
| int vol, err = 0; |
| |
| adev->voice.volume = volume; |
| if (adev->mode == AUDIO_MODE_IN_CALL) { |
| if (volume < 0.0) { |
| volume = 0.0; |
| } else if (volume > 1.0) { |
| volume = 1.0; |
| } |
| |
| vol = lrint(volume * 100.0); |
| |
| // Voice volume levels from android are mapped to driver volume levels as follows. |
| // 0 -> 5, 20 -> 4, 40 ->3, 60 -> 2, 80 -> 1, 100 -> 0 |
| // So adjust the volume to get the correct volume index in driver |
| vol = 100 - vol; |
| |
| err = platform_set_voice_volume(adev->platform, vol); |
| } |
| if (adev->mode == AUDIO_MODE_IN_COMMUNICATION) |
| err = voice_extn_compress_voip_set_volume(adev, volume); |
| |
| |
| return err; |
| } |
| |
| int voice_start_call(struct audio_device *adev) |
| { |
| int ret = 0; |
| |
| adev->voice.in_call = true; |
| |
| voice_set_mic_mute(adev, adev->voice.mic_mute); |
| |
| ret = voice_extn_start_call(adev); |
| if (ret == -ENOSYS) { |
| ret = voice_start_usecase(adev, USECASE_VOICE_CALL); |
| } |
| |
| return ret; |
| } |
| |
| int voice_stop_call(struct audio_device *adev) |
| { |
| int ret = 0; |
| |
| adev->voice.in_call = false; |
| ret = voice_extn_stop_call(adev); |
| if (ret == -ENOSYS) { |
| ret = voice_stop_usecase(adev, USECASE_VOICE_CALL); |
| } |
| |
| return ret; |
| } |
| |
| void voice_get_parameters(struct audio_device *adev, |
| struct str_parms *query, |
| struct str_parms *reply) |
| { |
| voice_extn_get_parameters(adev, query, reply); |
| } |
| |
| int voice_set_parameters(struct audio_device *adev, struct str_parms *parms) |
| { |
| char value[32]; |
| int ret = 0, err; |
| char *kv_pairs = str_parms_to_str(parms); |
| |
| ALOGV_IF(kv_pairs != NULL, "%s: enter: %s", __func__, kv_pairs); |
| |
| ret = voice_extn_set_parameters(adev, parms); |
| if (ret != 0) { |
| if (ret == -ENOSYS) |
| ret = 0; |
| else |
| goto done; |
| } |
| |
| ret = voice_extn_compress_voip_set_parameters(adev, parms); |
| if (ret != 0) { |
| if (ret == -ENOSYS) |
| ret = 0; |
| else |
| goto done; |
| } |
| |
| err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_TTY_MODE, value, sizeof(value)); |
| if (err >= 0) { |
| int tty_mode; |
| str_parms_del(parms, AUDIO_PARAMETER_KEY_TTY_MODE); |
| if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_OFF) == 0) |
| tty_mode = TTY_MODE_OFF; |
| else if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_VCO) == 0) |
| tty_mode = TTY_MODE_VCO; |
| else if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_HCO) == 0) |
| tty_mode = TTY_MODE_HCO; |
| else if (strcmp(value, AUDIO_PARAMETER_VALUE_TTY_FULL) == 0) |
| tty_mode = TTY_MODE_FULL; |
| else { |
| ret = -EINVAL; |
| goto done; |
| } |
| |
| if (tty_mode != adev->voice.tty_mode) { |
| adev->voice.tty_mode = tty_mode; |
| adev->acdb_settings = (adev->acdb_settings & TTY_MODE_CLEAR) | tty_mode; |
| if (voice_is_call_state_active_in_call(adev)) |
| voice_update_devices_for_all_voice_usecases(adev); |
| } |
| } |
| |
| err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_HAC, |
| value, sizeof(value)); |
| if (err >= 0) { |
| bool hac = false; |
| str_parms_del(parms, AUDIO_PARAMETER_KEY_HAC); |
| if (strcmp(value, AUDIO_PARAMETER_VALUE_HAC_ON) == 0) |
| hac = true; |
| |
| if (hac != adev->voice.hac) { |
| adev->voice.hac = hac; |
| if (voice_is_in_call(adev)) |
| voice_update_devices_for_all_voice_usecases(adev); |
| } |
| } |
| |
| err = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_INCALLMUSIC, |
| value, sizeof(value)); |
| if (err >= 0) { |
| str_parms_del(parms, AUDIO_PARAMETER_KEY_INCALLMUSIC); |
| if (strcmp(value, AUDIO_PARAMETER_VALUE_TRUE) == 0) |
| platform_start_incall_music_usecase(adev->platform); |
| else |
| platform_stop_incall_music_usecase(adev->platform); |
| } |
| |
| done: |
| ALOGV("%s: exit with code(%d)", __func__, ret); |
| free(kv_pairs); |
| return ret; |
| } |
| |
| void voice_init(struct audio_device *adev) |
| { |
| int i = 0; |
| int max_voice_sessions = MAX_VOICE_SESSIONS; |
| |
| if (!voice_extn_is_multi_session_supported()) |
| max_voice_sessions = 1; |
| |
| memset(&adev->voice, 0, sizeof(adev->voice)); |
| adev->voice.tty_mode = TTY_MODE_OFF; |
| adev->voice.hac = false; |
| adev->voice.volume = 1.0f; |
| adev->voice.mic_mute = false; |
| adev->voice.in_call = false; |
| for (i = 0; i < max_voice_sessions; i++) { |
| adev->voice.session[i].pcm_rx = NULL; |
| adev->voice.session[i].pcm_tx = NULL; |
| adev->voice.session[i].state.current = CALL_INACTIVE; |
| adev->voice.session[i].state.new = CALL_INACTIVE; |
| adev->voice.session[i].vsid = VOICE_VSID; |
| } |
| |
| voice_extn_init(adev); |
| } |
| |
| void voice_update_devices_for_all_voice_usecases(struct audio_device *adev) |
| { |
| struct listnode *node; |
| struct audio_usecase *usecase; |
| |
| list_for_each(node, &adev->usecase_list) { |
| usecase = node_to_item(node, struct audio_usecase, list); |
| if (usecase->type == VOICE_CALL) { |
| ALOGV("%s: updating device for usecase:%s", __func__, |
| use_case_table[usecase->id]); |
| usecase->stream.out = adev->current_call_output; |
| select_devices(adev, usecase->id); |
| } |
| } |
| } |
| |
| |