diff options
| -rw-r--r-- | audio/Android.mk | 33 | ||||
| -rwxr-xr-x | audio/audio_hw.c | 3125 | ||||
| -rw-r--r-- | audio/audio_hw.h | 228 | ||||
| -rwxr-xr-x | audio/ril_interface.c | 193 | ||||
| -rwxr-xr-x | audio/ril_interface.h | 83 |
5 files changed, 0 insertions, 3662 deletions
diff --git a/audio/Android.mk b/audio/Android.mk deleted file mode 100644 index 4655db0..0000000 --- a/audio/Android.mk +++ /dev/null @@ -1,33 +0,0 @@ -# Copyright (C) 2011 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. - -LOCAL_PATH := $(call my-dir) - -include $(CLEAR_VARS) - -LOCAL_MODULE := audio.primary.$(TARGET_BOOTLOADER_BOARD_NAME) -LOCAL_MODULE_PATH := $(TARGET_OUT_SHARED_LIBRARIES)/hw -LOCAL_MODULE_TAGS := optional - -LOCAL_SRC_FILES := audio_hw.c ril_interface.c - -LOCAL_C_INCLUDES += \ - external/tinyalsa/include \ - external/expat/lib \ - $(call include-path-for, audio-utils) \ - $(call include-path-for, audio-effects) - -LOCAL_SHARED_LIBRARIES := liblog libcutils libtinyalsa libaudioutils libdl libexpat - -include $(BUILD_SHARED_LIBRARY) diff --git a/audio/audio_hw.c b/audio/audio_hw.c deleted file mode 100755 index e06946a..0000000 --- a/audio/audio_hw.c +++ /dev/null @@ -1,3125 +0,0 @@ -/* - * Copyright (C) 2011 The Android Open Source Project - * Copyright (C) 2012 Wolfson Microelectronics plc - * Copyright (C) 2012 The CyanogenMod Project - * Daniel Hillenbrand <codeworkx@cyanogenmod.com> - * Guillaume "XpLoDWilD" Lesniak <xplodgui@gmail.com> - * - * Licensed under the Apache License, Version 2.0 (the "License"); - * you may not use this file except in compliance with the License. - * You may obtain a copy of the License at - * - * http://www.apache.org/licenses/LICENSE-2.0 - * - * Unless required by applicable law or agreed to in writing, software - * distributed under the License is distributed on an "AS IS" BASIS, - * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied. - * See the License for the specific language governing permissions and - * limitations under the License. - */ - -#define LOG_TAG "audio_hw_primary" -#define LOG_NDEBUG 0 - -#include <errno.h> -#include <pthread.h> -#include <stdint.h> -#include <sys/time.h> -#include <stdlib.h> -#include <expat.h> - -#include <cutils/log.h> -#include <cutils/str_parms.h> -#include <cutils/properties.h> - -#include <hardware/hardware.h> -#include <system/audio.h> -#include <hardware/audio.h> - -#include <tinyalsa/asoundlib.h> -#include <audio_utils/resampler.h> -#include <audio_utils/echo_reference.h> -#include <hardware/audio_effect.h> -#include <audio_effects/effect_aec.h> - -#include "audio_hw.h" -#include "ril_interface.h" - -struct pcm_config pcm_config_mm = { - .channels = 2, - .rate = MM_FULL_POWER_SAMPLING_RATE, - .period_size = DEEP_BUFFER_LONG_PERIOD_SIZE, - .period_count = PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT, - .format = PCM_FORMAT_S16_LE, -}; - -struct pcm_config pcm_config_tones = { - .channels = 2, - .rate = MM_FULL_POWER_SAMPLING_RATE, - .period_size = SHORT_PERIOD_SIZE, - .period_count = PLAYBACK_SHORT_PERIOD_COUNT, - .format = PCM_FORMAT_S16_LE, - .start_threshold = 0, - .avail_min = 0, -}; - -struct pcm_config pcm_config_capture = { - .channels = 2, - .rate = DEFAULT_IN_SAMPLING_RATE, - .period_size = CAPTURE_PERIOD_SIZE, - .period_count = CAPTURE_PERIOD_COUNT, - .format = PCM_FORMAT_S16_LE, -}; - -struct pcm_config pcm_config_vx = { - .channels = 2, - .rate = VX_NB_SAMPLING_RATE, - .period_size = 160, - .period_count = 2, - .format = PCM_FORMAT_S16_LE, -}; - -#define MIN(x, y) ((x) > (y) ? (y) : (x)) - -struct espresso_audio_device { - struct audio_hw_device hw_device; - - pthread_mutex_t lock; /* see note below on mutex acquisition order */ - struct espresso_dev_cfg *dev_cfgs; - int num_dev_cfgs; - struct mixer *mixer; - audio_mode_t mode; - int active_out_device; - int out_device; - int active_in_device; - int in_device; - struct pcm *pcm_modem_dl; - struct pcm *pcm_modem_ul; - struct pcm *pcm_bt_dl; - struct pcm *pcm_bt_ul; - int in_call; - float voice_volume; - struct espresso_stream_in *active_input; - struct espresso_stream_out *outputs[OUTPUT_TOTAL]; - bool mic_mute; - int tty_mode; - struct echo_reference_itfe *echo_reference; - bool bluetooth_nrec; - int wb_amr; - bool screen_off; - - /* RIL */ - struct ril_handle ril; -}; - -struct espresso_stream_out { - struct audio_stream_out stream; - - pthread_mutex_t lock; /* see note below on mutex acquisition order */ - struct pcm_config config[PCM_TOTAL]; - struct pcm *pcm[PCM_TOTAL]; - struct resampler_itfe *resampler; - char *buffer; - size_t buffer_frames; - int standby; - struct echo_reference_itfe *echo_reference; - int write_threshold; - bool use_long_periods; - audio_channel_mask_t channel_mask; - audio_channel_mask_t sup_channel_masks[3]; - - struct espresso_audio_device *dev; -}; - -#define MAX_PREPROCESSORS 3 /* maximum one AGC + one NS + one AEC per input stream */ - -struct effect_info_s { - effect_handle_t effect_itfe; - size_t num_channel_configs; - channel_config_t* channel_configs; -}; - -#define NUM_IN_AUX_CNL_CONFIGS 2 -channel_config_t in_aux_cnl_configs[NUM_IN_AUX_CNL_CONFIGS] = { - { AUDIO_CHANNEL_IN_FRONT , AUDIO_CHANNEL_IN_BACK}, - { AUDIO_CHANNEL_IN_STEREO , AUDIO_CHANNEL_IN_RIGHT} -}; - -struct espresso_stream_in { - struct audio_stream_in stream; - - pthread_mutex_t lock; /* see note below on mutex acquisition order */ - struct pcm_config config; - struct pcm *pcm; - int device; - struct resampler_itfe *resampler; - struct resampler_buffer_provider buf_provider; - unsigned int requested_rate; - int standby; - int source; - struct echo_reference_itfe *echo_reference; - bool need_echo_reference; - - int16_t *read_buf; - size_t read_buf_size; - size_t read_buf_frames; - - int16_t *proc_buf_in; - int16_t *proc_buf_out; - size_t proc_buf_size; - size_t proc_buf_frames; - - int16_t *ref_buf; - size_t ref_buf_size; - size_t ref_buf_frames; - - int read_status; - - int num_preprocessors; - struct effect_info_s preprocessors[MAX_PREPROCESSORS]; - - bool aux_channels_changed; - uint32_t main_channels; - uint32_t aux_channels; - struct espresso_audio_device *dev; -}; - -struct espresso_dev_cfg { - int mask; - - struct route_setting *on; - unsigned int on_len; - - struct route_setting *off; - unsigned int off_len; -}; - -/** - * NOTE: when multiple mutexes have to be acquired, always respect the following order: - * hw device > in stream > out stream - */ - -static void select_output_device(struct espresso_audio_device *adev); -static void select_input_device(struct espresso_audio_device *adev); -static int adev_set_voice_volume(struct audio_hw_device *dev, float volume); -static int do_input_standby(struct espresso_stream_in *in); -static int do_output_standby(struct espresso_stream_out *out); -static void in_update_aux_channels(struct espresso_stream_in *in, effect_handle_t effect); - -/* The enable flag when 0 makes the assumption that enums are disabled by - * "Off" and integers/booleans by 0 */ -static int set_bigroute_by_array(struct mixer *mixer, struct route_setting *route, - int enable) -{ - struct mixer_ctl *ctl; - unsigned int i, j, ret; - - /* Go through the route array and set each value */ - i = 0; - while (route[i].ctl_name) { - ctl = mixer_get_ctl_by_name(mixer, route[i].ctl_name); - if (!ctl) { - ALOGE("Unknown control '%s'\n", route[i].ctl_name); - return -EINVAL; - } - - if (route[i].strval) { - if (enable) { - ret = mixer_ctl_set_enum_by_string(ctl, route[i].strval); - if (ret != 0) { - ALOGE("Failed to set '%s' to '%s'\n", route[i].ctl_name, route[i].strval); - } else { - ALOGV("Set '%s' to '%s'\n", route[i].ctl_name, route[i].strval); - } - } else { - ret = mixer_ctl_set_enum_by_string(ctl, "Off"); - if (ret != 0) { - ALOGE("Failed to set '%s' to '%s'\n", route[i].ctl_name, route[i].strval); - } else { - ALOGV("Set '%s' to '%s'\n", route[i].ctl_name, "Off"); - } - } - } else { - /* This ensures multiple (i.e. stereo) values are set jointly */ - for (j = 0; j < mixer_ctl_get_num_values(ctl); j++) { - if (enable) { - ret = mixer_ctl_set_value(ctl, j, route[i].intval); - if (ret != 0) { - ALOGE("Failed to set '%s' to '%d'\n", route[i].ctl_name, route[i].intval); - } else { - ALOGV("Set '%s' to '%d'\n", route[i].ctl_name, route[i].intval); - } - } else { - ret = mixer_ctl_set_value(ctl, j, 0); - if (ret != 0) { - ALOGE("Failed to set '%s' to '%d'\n", route[i].ctl_name, route[i].intval); - } else { - ALOGV("Set '%s' to '%d'\n", route[i].ctl_name, 0); - } - } - } - } - i++; - } - - return 0; -} - -/* The enable flag when 0 makes the assumption that enums are disabled by - * "Off" and integers/booleans by 0 */ -static int set_route_by_array(struct mixer *mixer, struct route_setting *route, - unsigned int len) -{ - struct mixer_ctl *ctl; - unsigned int i, j, ret; - - /* Go through the route array and set each value */ - for (i = 0; i < len; i++) { - ctl = mixer_get_ctl_by_name(mixer, route[i].ctl_name); - if (!ctl) { - ALOGE("Unknown control '%s'\n", route[i].ctl_name); - return -EINVAL; - } - - if (route[i].strval) { - ret = mixer_ctl_set_enum_by_string(ctl, route[i].strval); - if (ret != 0) { - ALOGE("Failed to set '%s' to '%s'\n", - route[i].ctl_name, route[i].strval); - } else { - ALOGV("Set '%s' to '%s'\n", - route[i].ctl_name, route[i].strval); - } - - } else { - /* This ensures multiple (i.e. stereo) values are set jointly */ - for (j = 0; j < mixer_ctl_get_num_values(ctl); j++) { - ret = mixer_ctl_set_value(ctl, j, route[i].intval); - if (ret != 0) { - ALOGE("Failed to set '%s'.%d to %d\n", - route[i].ctl_name, j, route[i].intval); - } else { - ALOGV("Set '%s'.%d to %d\n", - route[i].ctl_name, j, route[i].intval); - } - } - } - } - - return 0; -} - -/* Must be called with lock */ -void select_devices(struct espresso_audio_device *adev) -{ - int i; - - if (adev->active_out_device == adev->out_device && adev->active_in_device == adev->in_device) - return; - - ALOGV("Changing output device %x => %x\n", adev->active_out_device, adev->out_device); - ALOGV("Changing input device %x => %x\n", adev->active_in_device, adev->in_device); - - /* Turn on new devices first so we don't glitch due to powerdown... */ - for (i = 0; i < adev->num_dev_cfgs; i++) - if ((adev->out_device & adev->dev_cfgs[i].mask) && - !(adev->active_out_device & adev->dev_cfgs[i].mask)) - set_route_by_array(adev->mixer, adev->dev_cfgs[i].on, - adev->dev_cfgs[i].on_len); - - for (i = 0; i < adev->num_dev_cfgs; i++) - if ((adev->in_device & adev->dev_cfgs[i].mask) && - !(adev->active_in_device & adev->dev_cfgs[i].mask)) - set_route_by_array(adev->mixer, adev->dev_cfgs[i].on, - adev->dev_cfgs[i].on_len); - - /* ...then disable old ones. */ - for (i = 0; i < adev->num_dev_cfgs; i++) - if (!(adev->out_device & adev->dev_cfgs[i].mask) && - (adev->active_out_device & adev->dev_cfgs[i].mask)) - set_route_by_array(adev->mixer, adev->dev_cfgs[i].off, - adev->dev_cfgs[i].off_len); - - for (i = 0; i < adev->num_dev_cfgs; i++) - if (!(adev->in_device & adev->dev_cfgs[i].mask) && - (adev->active_in_device & adev->dev_cfgs[i].mask)) - set_route_by_array(adev->mixer, adev->dev_cfgs[i].off, - adev->dev_cfgs[i].off_len); - - adev->active_out_device = adev->out_device; - adev->active_in_device = adev->in_device; -} - -static int start_call(struct espresso_audio_device *adev) -{ - ALOGV("Opening modem PCMs"); - int bt_on; - - bt_on = adev->out_device & AUDIO_DEVICE_OUT_ALL_SCO; - pcm_config_vx.rate = adev->wb_amr ? VX_WB_SAMPLING_RATE : VX_NB_SAMPLING_RATE; - - /* Open modem PCM channels */ - if (adev->pcm_modem_dl == NULL) { - ALOGD("Opening PCM modem DL stream"); - adev->pcm_modem_dl = pcm_open(CARD_DEFAULT, PORT_MODEM, PCM_OUT, &pcm_config_vx); - if (!pcm_is_ready(adev->pcm_modem_dl)) { - ALOGE("cannot open PCM modem DL stream: %s", pcm_get_error(adev->pcm_modem_dl)); - goto err_open_dl; - } - } - - if (adev->pcm_modem_ul == NULL) { - ALOGD("Opening PCM modem UL stream"); - adev->pcm_modem_ul = pcm_open(CARD_DEFAULT, PORT_MODEM, PCM_IN, &pcm_config_vx); - if (!pcm_is_ready(adev->pcm_modem_ul)) { - ALOGE("cannot open PCM modem UL stream: %s", pcm_get_error(adev->pcm_modem_ul)); - goto err_open_ul; - } - } - - ALOGD("Starting PCM modem streams"); - pcm_start(adev->pcm_modem_dl); - pcm_start(adev->pcm_modem_ul); - - /* Open bluetooth PCM channels */ - if (bt_on) { - ALOGV("Opening bluetooth PCMs"); - - if (adev->pcm_bt_dl == NULL) { - ALOGD("Opening PCM bluetooth DL stream"); - adev->pcm_bt_dl = pcm_open(CARD_DEFAULT, PORT_BT, PCM_OUT, &pcm_config_vx); - if (!pcm_is_ready(adev->pcm_bt_dl)) { - ALOGE("cannot open PCM bluetooth DL stream: %s", pcm_get_error(adev->pcm_bt_dl)); - goto err_open_dl; - } - } - - if (adev->pcm_bt_ul == NULL) { - ALOGD("Opening PCM bluetooth UL stream"); - adev->pcm_bt_ul = pcm_open(CARD_DEFAULT, PORT_BT, PCM_IN, &pcm_config_vx); - if (!pcm_is_ready(adev->pcm_bt_ul)) { - ALOGE("cannot open PCM bluetooth UL stream: %s", pcm_get_error(adev->pcm_bt_ul)); - goto err_open_ul; - } - } - ALOGD("Starting PCM bluetooth streams"); - pcm_start(adev->pcm_bt_dl); - pcm_start(adev->pcm_bt_ul); - } - - return 0; - -err_open_ul: - pcm_close(adev->pcm_modem_ul); - adev->pcm_modem_ul = NULL; - pcm_close(adev->pcm_bt_ul); - adev->pcm_bt_ul = NULL; -err_open_dl: - pcm_close(adev->pcm_modem_dl); - adev->pcm_modem_dl = NULL; - pcm_close(adev->pcm_bt_dl); - adev->pcm_bt_dl = NULL; - - return -ENOMEM; -} - -static void end_call(struct espresso_audio_device *adev) -{ - int bt_on; - bt_on = adev->out_device & AUDIO_DEVICE_OUT_ALL_SCO; - - if (adev->pcm_modem_dl != NULL) { - ALOGD("Stopping modem DL PCM"); - pcm_stop(adev->pcm_modem_dl); - ALOGV("Closing modem DL PCM"); - pcm_close(adev->pcm_modem_dl); - } - if (adev->pcm_modem_ul != NULL) { - ALOGD("Stopping modem UL PCM"); - pcm_stop(adev->pcm_modem_ul); - ALOGV("Closing modem UL PCM"); - pcm_close(adev->pcm_modem_ul); - } - adev->pcm_modem_dl = NULL; - adev->pcm_modem_ul = NULL; - - if (bt_on) { - if (adev->pcm_bt_dl != NULL) { - ALOGD("Stopping bluetooth DL PCM"); - pcm_stop(adev->pcm_bt_dl); - ALOGV("Closing bluetooth DL PCM"); - pcm_close(adev->pcm_bt_dl); - } - if (adev->pcm_bt_ul != NULL) { - ALOGD("Stopping bluetooth UL PCM"); - pcm_stop(adev->pcm_bt_ul); - ALOGV("Closing bluetooth UL PCM"); - pcm_close(adev->pcm_bt_ul); - } - } - adev->pcm_bt_dl = NULL; - adev->pcm_bt_ul = NULL; -} - -static void set_eq_filter(struct espresso_audio_device *adev) -{ -} - -void audio_set_wb_amr_callback(void *data, int enable) -{ - struct espresso_audio_device *adev = (struct espresso_audio_device *)data; - - pthread_mutex_lock(&adev->lock); - if (adev->wb_amr != enable) { - adev->wb_amr = enable; - - /* reopen the modem PCMs at the new rate */ - if (adev->in_call) { - end_call(adev); - select_output_device(adev); - start_call(adev); - } - } - pthread_mutex_unlock(&adev->lock); -} - -static void set_incall_device(struct espresso_audio_device *adev) -{ - int device_type; - - switch(adev->out_device) { - case AUDIO_DEVICE_OUT_EARPIECE: - device_type = SOUND_AUDIO_PATH_HANDSET; - break; - case AUDIO_DEVICE_OUT_SPEAKER: - case AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET: - case AUDIO_DEVICE_OUT_AUX_DIGITAL: - case AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET: - device_type = SOUND_AUDIO_PATH_SPEAKER; - break; - case AUDIO_DEVICE_OUT_WIRED_HEADSET: - device_type = SOUND_AUDIO_PATH_HEADSET; - break; - case AUDIO_DEVICE_OUT_WIRED_HEADPHONE: - device_type = SOUND_AUDIO_PATH_HEADPHONE; - break; - case AUDIO_DEVICE_OUT_BLUETOOTH_SCO: - case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_HEADSET: - case AUDIO_DEVICE_OUT_BLUETOOTH_SCO_CARKIT: - if (adev->bluetooth_nrec) { - device_type = SOUND_AUDIO_PATH_BLUETOOTH; - } else { - device_type = SOUND_AUDIO_PATH_BLUETOOTH_NO_NR; - } - break; - default: - device_type = SOUND_AUDIO_PATH_SPEAKER; - break; - } - - /* if output device isn't supported, open modem side to handset by default */ - ALOGE("%s: ril_set_call_audio_path(%d)", __func__, device_type); - ril_set_call_audio_path(&adev->ril, device_type); -} - -static void set_input_volumes(struct espresso_audio_device *adev, int main_mic_on, - int headset_mic_on, int sub_mic_on) -{ -} - -static void set_output_volumes(struct espresso_audio_device *adev, bool tty_volume) -{ -} - -static void force_all_standby(struct espresso_audio_device *adev) -{ - struct espresso_stream_in *in; - struct espresso_stream_out *out; - - /* only needed for low latency output streams as other streams are not used - * for voice use cases */ - if (adev->outputs[OUTPUT_LOW_LATENCY] != NULL && - !adev->outputs[OUTPUT_LOW_LATENCY]->standby) { - out = adev->outputs[OUTPUT_LOW_LATENCY]; - pthread_mutex_lock(&out->lock); - do_output_standby(out); - pthread_mutex_unlock(&out->lock); - } - - if (adev->active_input) { - in = adev->active_input; - pthread_mutex_lock(&in->lock); - do_input_standby(in); - pthread_mutex_unlock(&in->lock); - } -} - -static void select_mode(struct espresso_audio_device *adev) -{ - if (adev->mode == AUDIO_MODE_IN_CALL) { - ALOGE("Entering IN_CALL state, in_call=%d", adev->in_call); - if (!adev->in_call) { - force_all_standby(adev); - /* force earpiece route for in call state if speaker is the - only currently selected route. This prevents having to tear - down the modem PCMs to change route from speaker to earpiece - after the ringtone is played, but doesn't cause a route - change if a headset or bt device is already connected. If - speaker is not the only thing active, just remove it from - the route. We'll assume it'll never be used initally during - a call. This works because we're sure that the audio policy - manager will update the output device after the audio mode - change, even if the device selection did not change. */ - if (adev->out_device == AUDIO_DEVICE_OUT_SPEAKER) { - adev->out_device = AUDIO_DEVICE_OUT_EARPIECE; - adev->in_device = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN; - } else - adev->out_device &= ~AUDIO_DEVICE_OUT_SPEAKER; - select_output_device(adev); - start_call(adev); - ril_set_call_clock_sync(&adev->ril, SOUND_CLOCK_START); - adev_set_voice_volume(&adev->hw_device, adev->voice_volume); - adev->in_call = 1; - } - } else { - ALOGE("Leaving IN_CALL state, in_call=%d, mode=%d", - adev->in_call, adev->mode); - if (adev->in_call) { - adev->in_call = 0; - end_call(adev); - force_all_standby(adev); - select_output_device(adev); - select_input_device(adev); - } - } -} - -static void select_output_device(struct espresso_audio_device *adev) -{ - int headset_on; - int headphone_on; - int speaker_on; - int earpiece_on; - int bt_on; - bool tty_volume = false; - unsigned int channel; - - headset_on = adev->out_device & AUDIO_DEVICE_OUT_WIRED_HEADSET; - headphone_on = adev->out_device & AUDIO_DEVICE_OUT_WIRED_HEADPHONE; - speaker_on = adev->out_device & AUDIO_DEVICE_OUT_SPEAKER; - earpiece_on = adev->out_device & AUDIO_DEVICE_OUT_EARPIECE; - bt_on = adev->out_device & AUDIO_DEVICE_OUT_ALL_SCO; - - switch(adev->out_device) { - case AUDIO_DEVICE_OUT_SPEAKER: - ALOGD("%s: AUDIO_DEVICE_OUT_SPEAKER", __func__); - break; - case AUDIO_DEVICE_OUT_WIRED_HEADSET: - ALOGD("%s: AUDIO_DEVICE_OUT_WIRED_HEADSET", __func__); - break; - case AUDIO_DEVICE_OUT_WIRED_HEADPHONE: - ALOGD("%s: AUDIO_DEVICE_OUT_WIRED_HEADPHONE", __func__); - break; - case AUDIO_DEVICE_OUT_EARPIECE: - ALOGD("%s: AUDIO_DEVICE_OUT_EARPIECE", __func__); - break; - case AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET: - ALOGD("%s: AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET", __func__); - break; - case AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET: - ALOGD("%s: AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET", __func__); - break; - case AUDIO_DEVICE_OUT_AUX_DIGITAL: - ALOGD("%s: AUDIO_DEVICE_OUT_AUX_DIGITAL", __func__); - break; - case AUDIO_DEVICE_OUT_ALL_SCO: - ALOGD("%s: AUDIO_DEVICE_OUT_ALL_SCO", __func__); - break; - case AUDIO_DEVICE_OUT_USB_ACCESSORY: - ALOGD("%s: AUDIO_DEVICE_OUT_USB_ACCESSORY", __func__); - break; - case AUDIO_DEVICE_OUT_USB_DEVICE: - ALOGD("%s: AUDIO_DEVICE_OUT_USB_DEVICE", __func__); - break; - default: - ALOGD("%s: AUDIO_DEVICE_OUT_ALL", __func__); - break; - } - - select_devices(adev); - - set_eq_filter(adev); - - if (adev->mode == AUDIO_MODE_IN_CALL) { - if (!bt_on) { - /* force tx path according to TTY mode when in call */ - switch(adev->tty_mode) { - case TTY_MODE_FULL: - case TTY_MODE_HCO: - /* tx path from headset mic */ - headphone_on = 0; - headset_on = 1; - speaker_on = 0; - earpiece_on = 0; - break; - case TTY_MODE_VCO: - /* tx path from device sub mic */ - headphone_on = 0; - headset_on = 0; - speaker_on = 1; - earpiece_on = 0; - break; - case TTY_MODE_OFF: - default: - break; - } - } - - if (headset_on || headphone_on || speaker_on || earpiece_on) { - ALOGD("%s: set voicecall route: voicecall_default", __func__); - set_bigroute_by_array(adev->mixer, voicecall_default, 1); - } else { - ALOGD("%s: set voicecall route: voicecall_default_disable", __func__); - set_bigroute_by_array(adev->mixer, voicecall_default_disable, 1); - } - - if (speaker_on || earpiece_on || headphone_on) { - ALOGD("%s: set voicecall route: default_input", __func__); - set_bigroute_by_array(adev->mixer, default_input, 1); - } else { - ALOGD("%s: set voicecall route: default_input_disable", __func__); - set_bigroute_by_array(adev->mixer, default_input_disable, 1); - } - - if (headset_on) { - ALOGD("%s: set voicecall route: headset_input", __func__); - set_bigroute_by_array(adev->mixer, headset_input, 1); - } else { - ALOGD("%s: set voicecall route: headset_input_disable", __func__); - set_bigroute_by_array(adev->mixer, headset_input_disable, 1); - } - - if (bt_on) { - // bt uses a different port (PORT_BT) for playback, reopen the pcms - end_call(adev); - start_call(adev); - ALOGD("%s: set voicecall route: bt_input", __func__); - set_bigroute_by_array(adev->mixer, bt_input, 1); - ALOGD("%s: set voicecall route: bt_output", __func__); - set_bigroute_by_array(adev->mixer, bt_output, 1); - } else { - ALOGD("%s: set voicecall route: bt_input_disable", __func__); - set_bigroute_by_array(adev->mixer, bt_input_disable, 1); - ALOGD("%s: set voicecall route: bt_output_disable", __func__); - set_bigroute_by_array(adev->mixer, bt_output_disable, 1); - } - set_incall_device(adev); - } -} - -static void select_input_device(struct espresso_audio_device *adev) -{ - switch(adev->in_device) { - case AUDIO_DEVICE_IN_BUILTIN_MIC: - ALOGD("%s: AUDIO_DEVICE_IN_BUILTIN_MIC", __func__); - break; - case AUDIO_DEVICE_IN_WIRED_HEADSET: - ALOGD("%s: AUDIO_DEVICE_IN_WIRED_HEADSET", __func__); - break; - case AUDIO_DEVICE_IN_ALL_SCO: - ALOGD("%s: AUDIO_DEVICE_IN_ALL_SCO", __func__); - break; - default: - ALOGD("%s: AUDIO_DEVICE_IN_DEFAULT", __func__); - break; - } - - select_devices(adev); -} - -/* must be called with hw device and output stream mutexes locked */ -static int start_output_stream_low_latency(struct espresso_stream_out *out) -{ - struct espresso_audio_device *adev = out->dev; - unsigned int flags = PCM_OUT; - int i; - bool success = true; - - if (adev->mode != AUDIO_MODE_IN_CALL) { - select_output_device(adev); - } - - /* default to low power: will be corrected in out_write if necessary before first write to - * tinyalsa. - */ - - if (adev->out_device & ~(AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET | AUDIO_DEVICE_OUT_AUX_DIGITAL)) { - /* Something not a dock in use */ - out->config[PCM_NORMAL] = pcm_config_tones; - out->config[PCM_NORMAL].rate = MM_FULL_POWER_SAMPLING_RATE; - out->pcm[PCM_NORMAL] = pcm_open(CARD_DEFAULT, PORT_PLAYBACK, - flags, &out->config[PCM_NORMAL]); - } - - if (adev->out_device & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) { - /* SPDIF output in use */ - out->config[PCM_SPDIF] = pcm_config_tones; - out->config[PCM_SPDIF].rate = MM_FULL_POWER_SAMPLING_RATE; - out->pcm[PCM_SPDIF] = pcm_open(CARD_DEFAULT, PORT_PLAYBACK, - flags, &out->config[PCM_SPDIF]); - } - - /* Close any PCMs that could not be opened properly and return an error */ - for (i = 0; i < PCM_TOTAL; i++) { - if (out->pcm[i] && !pcm_is_ready(out->pcm[i])) { - ALOGE("%s: cannot open pcm_out driver %d: %s", __func__ , i, pcm_get_error(out->pcm[i])); - pcm_close(out->pcm[i]); - out->pcm[i] = NULL; - success = false; - } - } - - if (success) { - out->buffer_frames = pcm_config_tones.period_size * 2; - if (out->buffer == NULL) - out->buffer = malloc(out->buffer_frames * audio_stream_frame_size(&out->stream.common)); - - if (adev->echo_reference != NULL) - out->echo_reference = adev->echo_reference; - out->resampler->reset(out->resampler); - - return 0; - } - - return -ENOMEM; -} - -/* must be called with hw device and output stream mutexes locked */ -static int start_output_stream_deep_buffer(struct espresso_stream_out *out) -{ - struct espresso_audio_device *adev = out->dev; - - if (adev->mode != AUDIO_MODE_IN_CALL) { - select_output_device(adev); - } - - out->write_threshold = PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT * DEEP_BUFFER_LONG_PERIOD_SIZE; - out->use_long_periods = true; - - out->config[PCM_NORMAL] = pcm_config_mm; - out->config[PCM_NORMAL].rate = MM_FULL_POWER_SAMPLING_RATE; - out->pcm[PCM_NORMAL] = pcm_open(CARD_DEFAULT, PORT_PLAYBACK, - PCM_OUT | PCM_MMAP | PCM_NOIRQ, &out->config[PCM_NORMAL]); - if (out->pcm[PCM_NORMAL] && !pcm_is_ready(out->pcm[PCM_NORMAL])) { - ALOGE("%s: cannot open pcm_out driver: %s", __func__, pcm_get_error(out->pcm[PCM_NORMAL])); - pcm_close(out->pcm[PCM_NORMAL]); - out->pcm[PCM_NORMAL] = NULL; - return -ENOMEM; - } - out->buffer_frames = DEEP_BUFFER_SHORT_PERIOD_SIZE * 2; - if (out->buffer == NULL) - out->buffer = malloc(PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT * DEEP_BUFFER_LONG_PERIOD_SIZE); - - return 0; -} - -static int check_input_parameters(uint32_t sample_rate, audio_format_t format, int channel_count) -{ - if (format != AUDIO_FORMAT_PCM_16_BIT) - return -EINVAL; - - if ((channel_count < 1) || (channel_count > 2)) - return -EINVAL; - - switch(sample_rate) { - case 8000: - case 11025: - case 16000: - case 22050: - case 24000: - case 32000: - case 44100: - case 48000: - break; - default: - return -EINVAL; - } - - return 0; -} - -static size_t get_input_buffer_size(uint32_t sample_rate, audio_format_t format, int channel_count) -{ - size_t size; - size_t device_rate; - - if (check_input_parameters(sample_rate, format, channel_count) != 0) - return 0; - - /* take resampling into account and return the closest majoring - multiple of 16 frames, as audioflinger expects audio buffers to - be a multiple of 16 frames */ - size = (pcm_config_capture.period_size * sample_rate) / pcm_config_capture.rate; - size = ((size + 15) / 16) * 16; - - return size * channel_count * sizeof(short); -} - -static void add_echo_reference(struct espresso_stream_out *out, - struct echo_reference_itfe *reference) -{ - pthread_mutex_lock(&out->lock); - out->echo_reference = reference; - pthread_mutex_unlock(&out->lock); -} - -static void remove_echo_reference(struct espresso_stream_out *out, - struct echo_reference_itfe *reference) -{ - pthread_mutex_lock(&out->lock); - if (out->echo_reference == reference) { - /* stop writing to echo reference */ - reference->write(reference, NULL); - out->echo_reference = NULL; - } - pthread_mutex_unlock(&out->lock); -} - -static void put_echo_reference(struct espresso_audio_device *adev, - struct echo_reference_itfe *reference) -{ - if (adev->echo_reference != NULL && - reference == adev->echo_reference) { - /* echo reference is taken from the low latency output stream used - * for voice use cases */ - if (adev->outputs[OUTPUT_LOW_LATENCY] != NULL && - !adev->outputs[OUTPUT_LOW_LATENCY]->standby) - remove_echo_reference(adev->outputs[OUTPUT_LOW_LATENCY], reference); - release_echo_reference(reference); - adev->echo_reference = NULL; - } -} - -static struct echo_reference_itfe *get_echo_reference(struct espresso_audio_device *adev, - audio_format_t format, - uint32_t channel_count, - uint32_t sampling_rate) -{ - put_echo_reference(adev, adev->echo_reference); - /* echo reference is taken from the low latency output stream used - * for voice use cases */ - if (adev->outputs[OUTPUT_LOW_LATENCY] != NULL && - !adev->outputs[OUTPUT_LOW_LATENCY]->standby) { - struct audio_stream *stream = - &adev->outputs[OUTPUT_LOW_LATENCY]->stream.common; - uint32_t wr_channel_count = popcount(stream->get_channels(stream)); - uint32_t wr_sampling_rate = stream->get_sample_rate(stream); - - int status = create_echo_reference(AUDIO_FORMAT_PCM_16_BIT, - channel_count, - sampling_rate, - AUDIO_FORMAT_PCM_16_BIT, - wr_channel_count, - wr_sampling_rate, - &adev->echo_reference); - if (status == 0) - add_echo_reference(adev->outputs[OUTPUT_LOW_LATENCY], - adev->echo_reference); - } - return adev->echo_reference; -} - -static int get_playback_delay(struct espresso_stream_out *out, - size_t frames, - struct echo_reference_buffer *buffer) -{ - size_t kernel_frames; - int status; - int primary_pcm = 0; - - /* Find the first active PCM to act as primary */ - while ((primary_pcm < PCM_TOTAL) && !out->pcm[primary_pcm]) - primary_pcm++; - - status = pcm_get_htimestamp(out->pcm[primary_pcm], &kernel_frames, &buffer->time_stamp); - if (status < 0) { - buffer->time_stamp.tv_sec = 0; - buffer->time_stamp.tv_nsec = 0; - buffer->delay_ns = 0; - ALOGV("%s: pcm_get_htimestamp error," - "setting playbackTimestamp to 0", __func__); - return status; - } - - kernel_frames = pcm_get_buffer_size(out->pcm[primary_pcm]) - kernel_frames; - - /* adjust render time stamp with delay added by current driver buffer. - * Add the duration of current frame as we want the render time of the last - * sample being written. */ - buffer->delay_ns = (long)(((int64_t)(kernel_frames + frames)* 1000000000)/ - MM_FULL_POWER_SAMPLING_RATE); - - return 0; -} - -static uint32_t out_get_sample_rate(const struct audio_stream *stream) -{ - return DEFAULT_OUT_SAMPLING_RATE; -} - -static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate) -{ - return 0; -} - -static size_t out_get_buffer_size_low_latency(const struct audio_stream *stream) -{ - struct espresso_stream_out *out = (struct espresso_stream_out *)stream; - - /* take resampling into account and return the closest majoring - multiple of 16 frames, as audioflinger expects audio buffers to - be a multiple of 16 frames. Note: we use the default rate here - from pcm_config_tones.rate. */ - size_t size = (SHORT_PERIOD_SIZE * DEFAULT_OUT_SAMPLING_RATE) / pcm_config_tones.rate; - size = ((size + 15) / 16) * 16; - return size * audio_stream_frame_size((struct audio_stream *)stream); -} - -static size_t out_get_buffer_size_deep_buffer(const struct audio_stream *stream) -{ - struct espresso_stream_out *out = (struct espresso_stream_out *)stream; - - /* take resampling into account and return the closest majoring - multiple of 16 frames, as audioflinger expects audio buffers to - be a multiple of 16 frames. Note: we use the default rate here - from pcm_config_mm.rate. */ - size_t size = (DEEP_BUFFER_SHORT_PERIOD_SIZE * DEFAULT_OUT_SAMPLING_RATE) / - pcm_config_mm.rate; - size = ((size + 15) / 16) * 16; - return size * audio_stream_frame_size((struct audio_stream *)stream); -} - -static audio_channel_mask_t out_get_channels(const struct audio_stream *stream) -{ - struct espresso_stream_out *out = (struct espresso_stream_out *)stream; - - return out->channel_mask; -} - -static audio_format_t out_get_format(const struct audio_stream *stream) -{ - return AUDIO_FORMAT_PCM_16_BIT; -} - -static int out_set_format(struct audio_stream *stream, audio_format_t format) -{ - return 0; -} - -/* must be called with hw device and output stream mutexes locked */ -static int do_output_standby(struct espresso_stream_out *out) -{ - struct espresso_audio_device *adev = out->dev; - int i; - bool all_outputs_in_standby = true; - - if (!out->standby) { - out->standby = 1; - - for (i = 0; i < PCM_TOTAL; i++) { - if (out->pcm[i]) { - pcm_close(out->pcm[i]); - out->pcm[i] = NULL; - } - } - - for (i = 0; i < OUTPUT_TOTAL; i++) { - if (adev->outputs[i] != NULL && !adev->outputs[i]->standby) { - all_outputs_in_standby = false; - break; - } - } - - /* force standby on low latency output stream so that it can reuse HDMI driver if - * necessary when restarted */ - if (out == adev->outputs[OUTPUT_HDMI]) { - if (adev->outputs[OUTPUT_LOW_LATENCY] != NULL && - !adev->outputs[OUTPUT_LOW_LATENCY]->standby) { - struct espresso_stream_out *ll_out = adev->outputs[OUTPUT_LOW_LATENCY]; - pthread_mutex_lock(&ll_out->lock); - do_output_standby(ll_out); - pthread_mutex_unlock(&ll_out->lock); - } - } - - /* stop writing to echo reference */ - if (out->echo_reference != NULL) { - out->echo_reference->write(out->echo_reference, NULL); - out->echo_reference = NULL; - } - } - return 0; -} - -static int out_standby(struct audio_stream *stream) -{ - struct espresso_stream_out *out = (struct espresso_stream_out *)stream; - int status; - - pthread_mutex_lock(&out->dev->lock); - pthread_mutex_lock(&out->lock); - status = do_output_standby(out); - pthread_mutex_unlock(&out->lock); - pthread_mutex_unlock(&out->dev->lock); - return status; -} - -static int out_dump(const struct audio_stream *stream, int fd) -{ - return 0; -} - -static int out_set_parameters(struct audio_stream *stream, const char *kvpairs) -{ - struct espresso_stream_out *out = (struct espresso_stream_out *)stream; - struct espresso_audio_device *adev = out->dev; - struct espresso_stream_in *in; - struct str_parms *parms; - char *str; - char value[32]; - int ret, val = 0; - bool force_input_standby = false; - - parms = str_parms_create_str(kvpairs); - - ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); - if (ret >= 0) { - val = atoi(value); - pthread_mutex_lock(&adev->lock); - pthread_mutex_lock(&out->lock); - if (((adev->out_device) != val) && (val != 0)) { - /* this is needed only when changing device on low latency output - * as other output streams are not used for voice use cases nor - * handle duplication to HDMI or SPDIF */ - if (out == adev->outputs[OUTPUT_LOW_LATENCY] && !out->standby) { - /* a change in output device may change the microphone selection */ - if (adev->active_input && - adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) { - force_input_standby = true; - } - /* force standby if moving to/from HDMI/SPDIF or if the output - * device changes when in HDMI/SPDIF mode */ - /* FIXME also force standby when in call as some audio path switches do not work - * while in call and an output stream is active (e.g BT SCO => earpiece) */ - - /* FIXME workaround for audio being dropped when switching path without forcing standby - * (several hundred ms of audio can be lost: e.g beginning of a ringtone. We must understand - * the root cause in audio HAL, driver or ABE. - if (((val & AUDIO_DEVICE_OUT_AUX_DIGITAL) ^ - (adev->out_device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) || - ((val & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) ^ - (adev->out_device & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) || - (adev->out_device & (AUDIO_DEVICE_OUT_AUX_DIGITAL | - AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET))) - */ - if (((val & AUDIO_DEVICE_OUT_AUX_DIGITAL) ^ - (adev->out_device & AUDIO_DEVICE_OUT_AUX_DIGITAL)) || - ((val & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET) ^ - (adev->out_device & AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) || - (adev->out_device & (AUDIO_DEVICE_OUT_AUX_DIGITAL | - AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET)) || - ((val & AUDIO_DEVICE_OUT_SPEAKER) ^ - (adev->out_device & AUDIO_DEVICE_OUT_SPEAKER)) || - (adev->mode == AUDIO_MODE_IN_CALL)) - do_output_standby(out); - } - if (out != adev->outputs[OUTPUT_HDMI]) { - adev->out_device = val; - select_output_device(adev); - } - } - pthread_mutex_unlock(&out->lock); - if (force_input_standby) { - in = adev->active_input; - pthread_mutex_lock(&in->lock); - do_input_standby(in); - pthread_mutex_unlock(&in->lock); - } - pthread_mutex_unlock(&adev->lock); - } - - str_parms_destroy(parms); - return ret; -} - -static char * out_get_parameters(const struct audio_stream *stream, const char *keys) -{ - struct espresso_stream_out *out = (struct espresso_stream_out *)stream; - - struct str_parms *query = str_parms_create_str(keys); - char *str; - char value[256]; - struct str_parms *reply = str_parms_create(); - size_t i, j; - int ret; - bool first = true; - - ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value)); - if (ret >= 0) { - value[0] = '\0'; - i = 0; - while (out->sup_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->sup_channel_masks[i]) { - if (!first) { - strcat(value, "|"); - } - strcat(value, out_channels_name_to_enum_table[j].name); - first = false; - break; - } - } - i++; - } - str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value); - str = strdup(str_parms_to_str(reply)); - } else { - str = strdup(keys); - } - str_parms_destroy(query); - str_parms_destroy(reply); - return str; -} - -static uint32_t out_get_latency_low_latency(const struct audio_stream_out *stream) -{ - struct espresso_stream_out *out = (struct espresso_stream_out *)stream; - - /* Note: we use the default rate here from pcm_config_mm.rate */ - return (SHORT_PERIOD_SIZE * PLAYBACK_SHORT_PERIOD_COUNT * 1000) / pcm_config_tones.rate; -} - -static uint32_t out_get_latency_deep_buffer(const struct audio_stream_out *stream) -{ - struct espresso_stream_out *out = (struct espresso_stream_out *)stream; - - /* Note: we use the default rate here from pcm_config_mm.rate */ - return (DEEP_BUFFER_LONG_PERIOD_SIZE * PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT * 1000) / - pcm_config_mm.rate; -} - -static int out_set_volume(struct audio_stream_out *stream, float left, - float right) -{ - return -ENOSYS; -} - -static ssize_t out_write_low_latency(struct audio_stream_out *stream, const void* buffer, - size_t bytes) -{ - int ret; - struct espresso_stream_out *out = (struct espresso_stream_out *)stream; - struct espresso_audio_device *adev = out->dev; - size_t frame_size = audio_stream_frame_size(&out->stream.common); - size_t in_frames = bytes / frame_size; - size_t out_frames = in_frames; - bool force_input_standby = false; - struct espresso_stream_in *in; - int i; - - /* acquiring hw device mutex systematically is useful if a low priority thread is waiting - * on the output stream mutex - e.g. executing select_mode() while holding the hw device - * mutex - */ - pthread_mutex_lock(&adev->lock); - pthread_mutex_lock(&out->lock); - if (out->standby) { - ret = start_output_stream_low_latency(out); - if (ret != 0) { - pthread_mutex_unlock(&adev->lock); - goto exit; - } - out->standby = 0; - /* a change in output device may change the microphone selection */ - if (adev->active_input && - adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION) - force_input_standby = true; - } - pthread_mutex_unlock(&adev->lock); - - for (i = 0; i < PCM_TOTAL; i++) { - /* only use resampler if required */ - if (out->pcm[i] && (out->config[i].rate != DEFAULT_OUT_SAMPLING_RATE)) { - out_frames = out->buffer_frames; - out->resampler->resample_from_input(out->resampler, - (int16_t *)buffer, - &in_frames, - (int16_t *)out->buffer, - &out_frames); - break; - } - } - - if (out->echo_reference != NULL) { - struct echo_reference_buffer b; - b.raw = (void *)buffer; - b.frame_count = in_frames; - - get_playback_delay(out, out_frames, &b); - out->echo_reference->write(out->echo_reference, &b); - } - - /* Write to all active PCMs */ - for (i = 0; i < PCM_TOTAL; i++) { - if (out->pcm[i]) { - if (out->config[i].rate == DEFAULT_OUT_SAMPLING_RATE) { - /* PCM uses native sample rate */ - ret = PCM_WRITE(out->pcm[i], (void *)buffer, bytes); - } else { - /* PCM needs resampler */ - ret = PCM_WRITE(out->pcm[i], (void *)out->buffer, out_frames * frame_size); - } - if (ret) - break; - } - } - -exit: - pthread_mutex_unlock(&out->lock); - - if (ret != 0) { - usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) / - out_get_sample_rate(&stream->common)); - } - - if (force_input_standby) { - pthread_mutex_lock(&adev->lock); - if (adev->active_input) { - in = adev->active_input; - pthread_mutex_lock(&in->lock); - do_input_standby(in); - pthread_mutex_unlock(&in->lock); - } - pthread_mutex_unlock(&adev->lock); - } - - return bytes; -} - -static ssize_t out_write_deep_buffer(struct audio_stream_out *stream, const void* buffer, - size_t bytes) -{ - int ret; - struct espresso_stream_out *out = (struct espresso_stream_out *)stream; - struct espresso_audio_device *adev = out->dev; - size_t frame_size = audio_stream_frame_size(&out->stream.common); - size_t in_frames = bytes / frame_size; - size_t out_frames; - bool use_long_periods; - int kernel_frames; - void *buf; - - /* acquiring hw device mutex systematically is useful if a low priority thread is waiting - * on the output stream mutex - e.g. executing select_mode() while holding the hw device - * mutex - */ - pthread_mutex_lock(&adev->lock); - pthread_mutex_lock(&out->lock); - if (out->standby) { - ret = start_output_stream_deep_buffer(out); - if (ret != 0) { - pthread_mutex_unlock(&adev->lock); - goto exit; - } - out->standby = 0; - } - use_long_periods = adev->screen_off && !adev->active_input; - pthread_mutex_unlock(&adev->lock); - - if (use_long_periods != out->use_long_periods) { - size_t period_size; - size_t period_count; - - if (use_long_periods) { - period_size = DEEP_BUFFER_LONG_PERIOD_SIZE; - period_count = PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT; - } else { - period_size = DEEP_BUFFER_SHORT_PERIOD_SIZE; - period_count = PLAYBACK_DEEP_BUFFER_SHORT_PERIOD_COUNT; - } - out->write_threshold = period_size * period_count; - pcm_set_avail_min(out->pcm[PCM_NORMAL], period_size); - out->use_long_periods = use_long_periods; - } - - /* only use resampler if required */ - if (out->config[PCM_NORMAL].rate != DEFAULT_OUT_SAMPLING_RATE) { - out_frames = out->buffer_frames; - out->resampler->resample_from_input(out->resampler, - (int16_t *)buffer, - &in_frames, - (int16_t *)out->buffer, - &out_frames); - buf = (void *)out->buffer; - } else { - out_frames = in_frames; - buf = (void *)buffer; - } - - /* do not allow more than out->write_threshold frames in kernel pcm driver buffer */ - do { - struct timespec time_stamp; - - if (pcm_get_htimestamp(out->pcm[PCM_NORMAL], - (unsigned int *)&kernel_frames, &time_stamp) < 0) - break; - kernel_frames = pcm_get_buffer_size(out->pcm[PCM_NORMAL]) - kernel_frames; - - if (kernel_frames > out->write_threshold) { - unsigned long time = (unsigned long) - (((int64_t)(kernel_frames - out->write_threshold) * 1000000) / - MM_FULL_POWER_SAMPLING_RATE); - if (time < MIN_WRITE_SLEEP_US) - time = MIN_WRITE_SLEEP_US; - usleep(time); - } - } while (kernel_frames > out->write_threshold); - - ret = pcm_mmap_write(out->pcm[PCM_NORMAL], buf, out_frames * frame_size); - -exit: - pthread_mutex_unlock(&out->lock); - - if (ret != 0) { - usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) / - out_get_sample_rate(&stream->common)); - } - - return bytes; -} - -static int out_get_render_position(const struct audio_stream_out *stream, - uint32_t *dsp_frames) -{ - return -EINVAL; -} - -static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) -{ - return 0; -} - -static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) -{ - return 0; -} - -/** audio_stream_in implementation **/ - -/* must be called with hw device and input stream mutexes locked */ -static int start_input_stream(struct espresso_stream_in *in) -{ - int ret = 0; - struct espresso_audio_device *adev = in->dev; - - adev->active_input = in; - - if (adev->mode != AUDIO_MODE_IN_CALL) { - adev->in_device = in->device; - select_input_device(adev); - } - - if (in->aux_channels_changed) - { - in->aux_channels_changed = false; - in->config.channels = popcount(in->main_channels | in->aux_channels); - - if (in->resampler) { - /* release and recreate the resampler with the new number of channel of the input */ - release_resampler(in->resampler); - in->resampler = NULL; - ret = create_resampler(in->config.rate, - in->requested_rate, - in->config.channels, - RESAMPLER_QUALITY_DEFAULT, - &in->buf_provider, - &in->resampler); - } - ALOGV("%s: New channel configuration, " - "main_channels = [%04x], aux_channels = [%04x], config.channels = %d", - __func__, in->main_channels, in->aux_channels, in->config.channels); - } - - if (in->need_echo_reference && in->echo_reference == NULL) - in->echo_reference = get_echo_reference(adev, - AUDIO_FORMAT_PCM_16_BIT, - popcount(in->main_channels), - in->requested_rate); - - /* this assumes routing is done previously */ - in->pcm = pcm_open(CARD_DEFAULT, PORT_CAPTURE, PCM_IN, &in->config); - if (!pcm_is_ready(in->pcm)) { - ALOGE("cannot open pcm_in driver: %s", pcm_get_error(in->pcm)); - pcm_close(in->pcm); - adev->active_input = NULL; - return -ENOMEM; - } - - /* force read and proc buf reallocation case of frame size or channel count change */ - in->read_buf_frames = 0; - in->read_buf_size = 0; - in->proc_buf_frames = 0; - in->proc_buf_size = 0; - /* if no supported sample rate is available, use the resampler */ - if (in->resampler) { - in->resampler->reset(in->resampler); - } - return 0; -} - -static uint32_t in_get_sample_rate(const struct audio_stream *stream) -{ - struct espresso_stream_in *in = (struct espresso_stream_in *)stream; - - return in->requested_rate; -} - -static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) -{ - return 0; -} - -static size_t in_get_buffer_size(const struct audio_stream *stream) -{ - struct espresso_stream_in *in = (struct espresso_stream_in *)stream; - - return get_input_buffer_size(in->requested_rate, - AUDIO_FORMAT_PCM_16_BIT, - popcount(in->main_channels)); -} - -static audio_channel_mask_t in_get_channels(const struct audio_stream *stream) -{ - struct espresso_stream_in *in = (struct espresso_stream_in *)stream; - - return AUDIO_CHANNEL_IN_STEREO; -} - -static audio_format_t in_get_format(const struct audio_stream *stream) -{ - return AUDIO_FORMAT_PCM_16_BIT; -} - -static int in_set_format(struct audio_stream *stream, audio_format_t format) -{ - return 0; -} - -/* must be called with hw device and input stream mutexes locked */ -static int do_input_standby(struct espresso_stream_in *in) -{ - struct espresso_audio_device *adev = in->dev; - - if (!in->standby) { - pcm_close(in->pcm); - in->pcm = NULL; - - adev->active_input = 0; - if (adev->mode != AUDIO_MODE_IN_CALL) { - adev->in_device = AUDIO_DEVICE_NONE; - select_input_device(adev); - } - - if (in->echo_reference != NULL) { - /* stop reading from echo reference */ - in->echo_reference->read(in->echo_reference, NULL); - put_echo_reference(adev, in->echo_reference); - in->echo_reference = NULL; - } - - in->standby = 1; - } - return 0; -} - -static int in_standby(struct audio_stream *stream) -{ - struct espresso_stream_in *in = (struct espresso_stream_in *)stream; - int status; - - pthread_mutex_lock(&in->dev->lock); - pthread_mutex_lock(&in->lock); - status = do_input_standby(in); - pthread_mutex_unlock(&in->lock); - pthread_mutex_unlock(&in->dev->lock); - return status; -} - -static int in_dump(const struct audio_stream *stream, int fd) -{ - return 0; -} - -static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) -{ - struct espresso_stream_in *in = (struct espresso_stream_in *)stream; - struct espresso_audio_device *adev = in->dev; - struct str_parms *parms; - char *str; - char value[32]; - int ret, val = 0; - bool do_standby = false; - - parms = str_parms_create_str(kvpairs); - - ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value)); - - pthread_mutex_lock(&adev->lock); - pthread_mutex_lock(&in->lock); - if (ret >= 0) { - val = atoi(value); - /* no audio source uses val == 0 */ - if ((in->source != val) && (val != 0)) { - in->source = val; - do_standby = true; - } - } - - ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); - if (ret >= 0) { - val = atoi(value) & ~AUDIO_DEVICE_BIT_IN; - if ((in->device != val) && (val != 0)) { - in->device = val; - do_standby = true; - /* make sure new device selection is incompatible with multi-mic pre processing - * configuration */ - in_update_aux_channels(in, NULL); - } - } - - if (do_standby) - do_input_standby(in); - pthread_mutex_unlock(&in->lock); - pthread_mutex_unlock(&adev->lock); - - str_parms_destroy(parms); - return ret; -} - -static char * in_get_parameters(const struct audio_stream *stream, - const char *keys) -{ - return strdup(""); -} - -static int in_set_gain(struct audio_stream_in *stream, float gain) -{ - return 0; -} - -static void get_capture_delay(struct espresso_stream_in *in, - size_t frames, - struct echo_reference_buffer *buffer) -{ - - /* read frames available in kernel driver buffer */ - size_t kernel_frames; - struct timespec tstamp; - long buf_delay; - long rsmp_delay; - long kernel_delay; - long delay_ns; - - if (pcm_get_htimestamp(in->pcm, &kernel_frames, &tstamp) < 0) { - buffer->time_stamp.tv_sec = 0; - buffer->time_stamp.tv_nsec = 0; - buffer->delay_ns = 0; - ALOGW("%s: pcm_htimestamp error", __func__); - return; - } - - /* read frames available in audio HAL input buffer - * add number of frames being read as we want the capture time of first sample - * in current buffer */ - /* frames in in->buffer are at driver sampling rate while frames in in->proc_buf are - * at requested sampling rate */ - buf_delay = (long)(((int64_t)(in->read_buf_frames) * 1000000000) / in->config.rate + - ((int64_t)(in->proc_buf_frames) * 1000000000) / - in->requested_rate); - - /* add delay introduced by resampler */ - rsmp_delay = 0; - if (in->resampler) { - rsmp_delay = in->resampler->delay_ns(in->resampler); - } - - kernel_delay = (long)(((int64_t)kernel_frames * 1000000000) / in->config.rate); - - delay_ns = kernel_delay + buf_delay + rsmp_delay; - - buffer->time_stamp = tstamp; - buffer->delay_ns = delay_ns; - ALOGV("%s: time_stamp = [%ld].[%ld], delay_ns: [%d]," - " kernel_delay:[%ld], buf_delay:[%ld], rsmp_delay:[%ld], kernel_frames:[%d], " - "in->read_buf_frames:[%d], in->proc_buf_frames:[%d], frames:[%d]", - __func__, buffer->time_stamp.tv_sec , buffer->time_stamp.tv_nsec, buffer->delay_ns, - kernel_delay, buf_delay, rsmp_delay, kernel_frames, - in->read_buf_frames, in->proc_buf_frames, frames); - -} - -static int32_t update_echo_reference(struct espresso_stream_in *in, size_t frames) -{ - struct echo_reference_buffer b; - b.delay_ns = 0; - - ALOGV("%s: frames = [%d], in->ref_frames_in = [%d], " - "b.frame_count = [%d]", - __func__, frames, in->ref_buf_frames, frames - in->ref_buf_frames); - if (in->ref_buf_frames < frames) { - if (in->ref_buf_size < frames) { - in->ref_buf_size = frames; - in->ref_buf = (int16_t *)realloc(in->ref_buf, pcm_frames_to_bytes(in->pcm, frames)); - ALOG_ASSERT((in->ref_buf != NULL), - "%s failed to reallocate ref_buf", __func__); - ALOGV("%s: ref_buf %p extended to %d bytes", - __func__, in->ref_buf, pcm_frames_to_bytes(in->pcm, frames)); - } - b.frame_count = frames - in->ref_buf_frames; - b.raw = (void *)(in->ref_buf + in->ref_buf_frames * in->config.channels); - - get_capture_delay(in, frames, &b); - - if (in->echo_reference->read(in->echo_reference, &b) == 0) - { - in->ref_buf_frames += b.frame_count; - ALOGD("%s: in->ref_buf_frames:[%d], " - "in->ref_buf_size:[%d], frames:[%d], b.frame_count:[%d]", - __func__, in->ref_buf_frames, in->ref_buf_size, frames, b.frame_count); - } - } else - ALOGW("%s: NOT enough frames to read ref buffer", __func__); - return b.delay_ns; -} - -static int set_preprocessor_param(effect_handle_t handle, - effect_param_t *param) -{ - uint32_t size = sizeof(int); - uint32_t psize = ((param->psize - 1) / sizeof(int) + 1) * sizeof(int) + - param->vsize; - - int status = (*handle)->command(handle, - EFFECT_CMD_SET_PARAM, - sizeof (effect_param_t) + psize, - param, - &size, - ¶m->status); - if (status == 0) - status = param->status; - - return status; -} - -static int set_preprocessor_echo_delay(effect_handle_t handle, - int32_t delay_us) -{ - uint32_t buf[sizeof(effect_param_t) / sizeof(uint32_t) + 2]; - effect_param_t *param = (effect_param_t *)buf; - - param->psize = sizeof(uint32_t); - param->vsize = sizeof(uint32_t); - *(uint32_t *)param->data = AEC_PARAM_ECHO_DELAY; - *((int32_t *)param->data + 1) = delay_us; - - return set_preprocessor_param(handle, param); -} - -static void push_echo_reference(struct espresso_stream_in *in, size_t frames) -{ - /* read frames from echo reference buffer and update echo delay - * in->ref_buf_frames is updated with frames available in in->ref_buf */ - int32_t delay_us = update_echo_reference(in, frames)/1000; - int i; - audio_buffer_t buf; - - if (in->ref_buf_frames < frames) - frames = in->ref_buf_frames; - - buf.frameCount = frames; - buf.raw = in->ref_buf; - - for (i = 0; i < in->num_preprocessors; i++) { - if ((*in->preprocessors[i].effect_itfe)->process_reverse == NULL) - continue; - - (*in->preprocessors[i].effect_itfe)->process_reverse(in->preprocessors[i].effect_itfe, - &buf, - NULL); - set_preprocessor_echo_delay(in->preprocessors[i].effect_itfe, delay_us); - } - - in->ref_buf_frames -= buf.frameCount; - if (in->ref_buf_frames) { - memcpy(in->ref_buf, - in->ref_buf + buf.frameCount * in->config.channels, - in->ref_buf_frames * in->config.channels * sizeof(int16_t)); - } -} - -static int get_next_buffer(struct resampler_buffer_provider *buffer_provider, - struct resampler_buffer* buffer) -{ - struct espresso_stream_in *in; - - if (buffer_provider == NULL || buffer == NULL) - return -EINVAL; - - in = (struct espresso_stream_in *)((char *)buffer_provider - - offsetof(struct espresso_stream_in, buf_provider)); - - if (in->pcm == NULL) { - buffer->raw = NULL; - buffer->frame_count = 0; - in->read_status = -ENODEV; - return -ENODEV; - } - - if (in->read_buf_frames == 0) { - size_t size_in_bytes = pcm_frames_to_bytes(in->pcm, in->config.period_size); - if (in->read_buf_size < in->config.period_size) { - in->read_buf_size = in->config.period_size; - in->read_buf = (int16_t *) realloc(in->read_buf, size_in_bytes); - ALOG_ASSERT((in->read_buf != NULL), - "%s failed to reallocate read_buf", __func__); - ALOGV("%s: read_buf %p extended to %d bytes", - __func__, in->read_buf, size_in_bytes); - } - - in->read_status = pcm_read(in->pcm, (void*)in->read_buf, size_in_bytes); - - if (in->read_status != 0) { - ALOGE("%s: pcm_read error %d", __func__, in->read_status); - buffer->raw = NULL; - buffer->frame_count = 0; - return in->read_status; - } - in->read_buf_frames = in->config.period_size; - } - - buffer->frame_count = (buffer->frame_count > in->read_buf_frames) ? - in->read_buf_frames : buffer->frame_count; - buffer->i16 = in->read_buf + (in->config.period_size - in->read_buf_frames) * - in->config.channels; - - return in->read_status; - -} - -static void release_buffer(struct resampler_buffer_provider *buffer_provider, - struct resampler_buffer* buffer) -{ - struct espresso_stream_in *in; - - if (buffer_provider == NULL || buffer == NULL) - return; - - in = (struct espresso_stream_in *)((char *)buffer_provider - - offsetof(struct espresso_stream_in, buf_provider)); - - in->read_buf_frames -= buffer->frame_count; -} - -/* read_frames() reads frames from kernel driver, down samples to capture rate - * if necessary and output the number of frames requested to the buffer specified */ -static ssize_t read_frames(struct espresso_stream_in *in, void *buffer, ssize_t frames) -{ - ssize_t frames_wr = 0; - - while (frames_wr < frames) { - size_t frames_rd = frames - frames_wr; - if (in->resampler != NULL) { - in->resampler->resample_from_provider(in->resampler, - (int16_t *)((char *)buffer + - pcm_frames_to_bytes(in->pcm ,frames_wr)), - &frames_rd); - - } else { - struct resampler_buffer buf = { - { raw : NULL, }, - frame_count : frames_rd, - }; - get_next_buffer(&in->buf_provider, &buf); - if (buf.raw != NULL) { - memcpy((char *)buffer + - pcm_frames_to_bytes(in->pcm, frames_wr), - buf.raw, - pcm_frames_to_bytes(in->pcm, buf.frame_count)); - frames_rd = buf.frame_count; - } - release_buffer(&in->buf_provider, &buf); - } - /* in->read_status is updated by getNextBuffer() also called by - * in->resampler->resample_from_provider() */ - if (in->read_status != 0) - return in->read_status; - - frames_wr += frames_rd; - } - return frames_wr; -} - -/* process_frames() reads frames from kernel driver (via read_frames()), - * calls the active audio pre processings and output the number of frames requested - * to the buffer specified */ -static ssize_t process_frames(struct espresso_stream_in *in, void* buffer, ssize_t frames) -{ - ssize_t frames_wr = 0; - audio_buffer_t in_buf; - audio_buffer_t out_buf; - int i; - bool has_aux_channels = (~in->main_channels & in->aux_channels); - void *proc_buf_out; - - if (has_aux_channels) - proc_buf_out = in->proc_buf_out; - else - proc_buf_out = buffer; - - /* since all the processing below is done in frames and using the config.channels - * as the number of channels, no changes is required in case aux_channels are present */ - while (frames_wr < frames) { - /* first reload enough frames at the end of process input buffer */ - if (in->proc_buf_frames < (size_t)frames) { - ssize_t frames_rd; - - if (in->proc_buf_size < (size_t)frames) { - size_t size_in_bytes = pcm_frames_to_bytes(in->pcm, frames); - - in->proc_buf_size = (size_t)frames; - in->proc_buf_in = (int16_t *)realloc(in->proc_buf_in, size_in_bytes); - ALOG_ASSERT((in->proc_buf_in != NULL), - "%s failed to reallocate proc_buf_in", __func__); - if (has_aux_channels) { - in->proc_buf_out = (int16_t *)realloc(in->proc_buf_out, size_in_bytes); - ALOG_ASSERT((in->proc_buf_out != NULL), - "%s failed to reallocate proc_buf_out", __func__); - proc_buf_out = in->proc_buf_out; - } - ALOGV("process_frames(): proc_buf_in %p extended to %d bytes", - in->proc_buf_in, size_in_bytes); - } - frames_rd = read_frames(in, - in->proc_buf_in + - in->proc_buf_frames * in->config.channels, - frames - in->proc_buf_frames); - if (frames_rd < 0) { - frames_wr = frames_rd; - break; - } - in->proc_buf_frames += frames_rd; - } - - if (in->echo_reference != NULL) - push_echo_reference(in, in->proc_buf_frames); - - /* in_buf.frameCount and out_buf.frameCount indicate respectively - * the maximum number of frames to be consumed and produced by process() */ - in_buf.frameCount = in->proc_buf_frames; - in_buf.s16 = in->proc_buf_in; - out_buf.frameCount = frames - frames_wr; - out_buf.s16 = (int16_t *)proc_buf_out + frames_wr * in->config.channels; - - /* FIXME: this works because of current pre processing library implementation that - * does the actual process only when the last enabled effect process is called. - * The generic solution is to have an output buffer for each effect and pass it as - * input to the next. - */ - for (i = 0; i < in->num_preprocessors; i++) { - (*in->preprocessors[i].effect_itfe)->process(in->preprocessors[i].effect_itfe, - &in_buf, - &out_buf); - } - - /* process() has updated the number of frames consumed and produced in - * in_buf.frameCount and out_buf.frameCount respectively - * move remaining frames to the beginning of in->proc_buf_in */ - in->proc_buf_frames -= in_buf.frameCount; - - if (in->proc_buf_frames) { - memcpy(in->proc_buf_in, - in->proc_buf_in + in_buf.frameCount * in->config.channels, - in->proc_buf_frames * in->config.channels * sizeof(int16_t)); - } - - /* if not enough frames were passed to process(), read more and retry. */ - if (out_buf.frameCount == 0) { - ALOGW("No frames produced by preproc"); - continue; - } - - if ((frames_wr + (ssize_t)out_buf.frameCount) <= frames) { - frames_wr += out_buf.frameCount; - } else { - /* The effect does not comply to the API. In theory, we should never end up here! */ - ALOGE("%s: preprocessing produced too many frames: %d + %d > %d !", __func__, - (unsigned int)frames_wr, out_buf.frameCount, (unsigned int)frames); - frames_wr = frames; - } - } - - /* Remove aux_channels that have been added on top of main_channels - * Assumption is made that the channels are interleaved and that the main - * channels are first. */ - if (has_aux_channels) - { - size_t src_channels = in->config.channels; - size_t dst_channels = popcount(in->main_channels); - int16_t* src_buffer = (int16_t *)proc_buf_out; - int16_t* dst_buffer = (int16_t *)buffer; - - if (dst_channels == 1) { - for (i = frames_wr; i > 0; i--) - { - *dst_buffer++ = *src_buffer; - src_buffer += src_channels; - } - } else { - for (i = frames_wr; i > 0; i--) - { - memcpy(dst_buffer, src_buffer, dst_channels*sizeof(int16_t)); - dst_buffer += dst_channels; - src_buffer += src_channels; - } - } - } - - return frames_wr; -} - -static ssize_t in_read(struct audio_stream_in *stream, void* buffer, - size_t bytes) -{ - int ret = 0; - struct espresso_stream_in *in = (struct espresso_stream_in *)stream; - struct espresso_audio_device *adev = in->dev; - size_t frames_rq = bytes / audio_stream_frame_size(&stream->common); - - /* acquiring hw device mutex systematically is useful if a low priority thread is waiting - * on the input stream mutex - e.g. executing select_mode() while holding the hw device - * mutex - */ - pthread_mutex_lock(&adev->lock); - pthread_mutex_lock(&in->lock); - if (in->standby) { - ret = start_input_stream(in); - if (ret == 0) - in->standby = 0; - } - pthread_mutex_unlock(&adev->lock); - - if (ret < 0) - goto exit; - - if (in->num_preprocessors != 0) - ret = process_frames(in, buffer, frames_rq); - else if (in->resampler != NULL) - ret = read_frames(in, buffer, frames_rq); - else - ret = pcm_read(in->pcm, buffer, bytes); - - if (ret > 0) - ret = 0; - - if (ret == 0 && adev->mic_mute) - memset(buffer, 0, bytes); - -exit: - if (ret < 0) - usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) / - in_get_sample_rate(&stream->common)); - - pthread_mutex_unlock(&in->lock); - return bytes; -} - -static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) -{ - return 0; -} - -#define GET_COMMAND_STATUS(status, fct_status, cmd_status) \ - do { \ - if (fct_status != 0) \ - status = fct_status; \ - else if (cmd_status != 0) \ - status = cmd_status; \ - } while(0) - -static int in_configure_reverse(struct espresso_stream_in *in) -{ - int32_t cmd_status; - uint32_t size = sizeof(int); - effect_config_t config; - int32_t status = 0; - int32_t fct_status = 0; - int i; - - if (in->num_preprocessors > 0) { - config.inputCfg.channels = in->main_channels; - config.outputCfg.channels = in->main_channels; - config.inputCfg.format = AUDIO_FORMAT_PCM_16_BIT; - config.outputCfg.format = AUDIO_FORMAT_PCM_16_BIT; - config.inputCfg.samplingRate = in->requested_rate; - config.outputCfg.samplingRate = in->requested_rate; - config.inputCfg.mask = - ( EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT ); - config.outputCfg.mask = - ( EFFECT_CONFIG_SMP_RATE | EFFECT_CONFIG_CHANNELS | EFFECT_CONFIG_FORMAT ); - - for (i = 0; i < in->num_preprocessors; i++) - { - if ((*in->preprocessors[i].effect_itfe)->process_reverse == NULL) - continue; - fct_status = (*(in->preprocessors[i].effect_itfe))->command( - in->preprocessors[i].effect_itfe, - EFFECT_CMD_SET_CONFIG_REVERSE, - sizeof(effect_config_t), - &config, - &size, - &cmd_status); - GET_COMMAND_STATUS(status, fct_status, cmd_status); - } - } - return status; -} - -#define MAX_NUM_CHANNEL_CONFIGS 10 - -static void in_read_audio_effect_channel_configs(struct espresso_stream_in *in, - struct effect_info_s *effect_info) -{ - /* size and format of the cmd are defined in hardware/audio_effect.h */ - effect_handle_t effect = effect_info->effect_itfe; - uint32_t cmd_size = 2 * sizeof(uint32_t); - uint32_t cmd[] = { EFFECT_FEATURE_AUX_CHANNELS, MAX_NUM_CHANNEL_CONFIGS }; - /* reply = status + number of configs (n) + n x channel_config_t */ - uint32_t reply_size = - 2 * sizeof(uint32_t) + (MAX_NUM_CHANNEL_CONFIGS * sizeof(channel_config_t)); - int32_t reply[reply_size]; - int32_t cmd_status; - - ALOG_ASSERT((effect_info->num_channel_configs == 0), - "in_read_audio_effect_channel_configs() num_channel_configs not cleared"); - ALOG_ASSERT((effect_info->channel_configs == NULL), - "in_read_audio_effect_channel_configs() channel_configs not cleared"); - - /* if this command is not supported, then the effect is supposed to return -EINVAL. - * This error will be interpreted as if the effect supports the main_channels but does not - * support any aux_channels */ - cmd_status = (*effect)->command(effect, - EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS, - cmd_size, - (void*)&cmd, - &reply_size, - (void*)&reply); - - if (cmd_status != 0) { - ALOGI("%s: fx->command returned %d", __func__, cmd_status); - return; - } - - if (reply[0] != 0) { - ALOGW("%s: " - "command EFFECT_CMD_GET_FEATURE_SUPPORTED_CONFIGS error %d num configs %d", - __func__, reply[0], (reply[0] == -ENOMEM) ? reply[1] : MAX_NUM_CHANNEL_CONFIGS); - return; - } - - /* the feature is not supported */ - ALOGI("in_read_audio_effect_channel_configs()(): " - "Feature supported and adding %d channel configs to the list", reply[1]); - effect_info->num_channel_configs = reply[1]; - effect_info->channel_configs = - (channel_config_t *) malloc(sizeof(channel_config_t) * reply[1]); /* n x configs */ - memcpy(effect_info->channel_configs, (reply + 2), sizeof(channel_config_t) * reply[1]); -} - - -static uint32_t in_get_aux_channels(struct espresso_stream_in *in) -{ - int i; - channel_config_t new_chcfg = {0, 0}; - - if (in->num_preprocessors == 0) - return 0; - - /* do not enable dual mic configurations when capturing from other microphones than - * main or sub */ - if (!(in->device & (AUDIO_DEVICE_IN_BUILTIN_MIC | AUDIO_DEVICE_IN_BACK_MIC))) - return 0; - - /* retain most complex aux channels configuration compatible with requested main channels and - * supported by audio driver and all pre processors */ - for (i = 0; i < NUM_IN_AUX_CNL_CONFIGS; i++) { - channel_config_t *cur_chcfg = &in_aux_cnl_configs[i]; - if (cur_chcfg->main_channels == in->main_channels) { - size_t match_cnt; - size_t idx_preproc; - for (idx_preproc = 0, match_cnt = 0; - /* no need to continue if at least one preprocessor doesn't match */ - idx_preproc < (size_t)in->num_preprocessors && match_cnt == idx_preproc; - idx_preproc++) { - struct effect_info_s *effect_info = &in->preprocessors[idx_preproc]; - size_t idx_chcfg; - - for (idx_chcfg = 0; idx_chcfg < effect_info->num_channel_configs; idx_chcfg++) { - if (memcmp(effect_info->channel_configs + idx_chcfg, - cur_chcfg, - sizeof(channel_config_t)) == 0) { - match_cnt++; - break; - } - } - } - /* if all preprocessors match, we have a candidate */ - if (match_cnt == (size_t)in->num_preprocessors) { - /* retain most complex aux channels configuration */ - if (popcount(cur_chcfg->aux_channels) > popcount(new_chcfg.aux_channels)) { - new_chcfg = *cur_chcfg; - } - } - } - } - - ALOGI("in_get_aux_channels(): return %04x", new_chcfg.aux_channels); - - return new_chcfg.aux_channels; -} - -static int in_configure_effect_channels(effect_handle_t effect, - channel_config_t *channel_config) -{ - int status = 0; - int fct_status; - int32_t cmd_status; - uint32_t reply_size; - effect_config_t config; - uint32_t cmd[(sizeof(uint32_t) + sizeof(channel_config_t) - 1) / sizeof(uint32_t) + 1]; - - ALOGI("in_configure_effect_channels(): configure effect with channels: [%04x][%04x]", - channel_config->main_channels, - channel_config->aux_channels); - - config.inputCfg.mask = EFFECT_CONFIG_CHANNELS; - config.outputCfg.mask = EFFECT_CONFIG_CHANNELS; - reply_size = sizeof(effect_config_t); - fct_status = (*effect)->command(effect, - EFFECT_CMD_GET_CONFIG, - 0, - NULL, - &reply_size, - &config); - if (fct_status != 0) { - ALOGE("in_configure_effect_channels(): EFFECT_CMD_GET_CONFIG failed"); - return fct_status; - } - - config.inputCfg.channels = channel_config->main_channels | channel_config->aux_channels; - config.outputCfg.channels = config.inputCfg.channels; - reply_size = sizeof(uint32_t); - fct_status = (*effect)->command(effect, - EFFECT_CMD_SET_CONFIG, - sizeof(effect_config_t), - &config, - &reply_size, - &cmd_status); - GET_COMMAND_STATUS(status, fct_status, cmd_status); - - cmd[0] = EFFECT_FEATURE_AUX_CHANNELS; - memcpy(cmd + 1, channel_config, sizeof(channel_config_t)); - reply_size = sizeof(uint32_t); - fct_status = (*effect)->command(effect, - EFFECT_CMD_SET_FEATURE_CONFIG, - sizeof(cmd), //sizeof(uint32_t) + sizeof(channel_config_t), - cmd, - &reply_size, - &cmd_status); - GET_COMMAND_STATUS(status, fct_status, cmd_status); - - /* some implementations need to be re-enabled after a config change */ - reply_size = sizeof(uint32_t); - fct_status = (*effect)->command(effect, - EFFECT_CMD_ENABLE, - 0, - NULL, - &reply_size, - &cmd_status); - GET_COMMAND_STATUS(status, fct_status, cmd_status); - - return status; -} - -static int in_reconfigure_channels(struct espresso_stream_in *in, - effect_handle_t effect, - channel_config_t *channel_config, - bool config_changed) { - - int status = 0; - - ALOGI("%s: config_changed %d effect %p", - __func__, config_changed, effect); - - /* if config changed, reconfigure all previously added effects */ - if (config_changed) { - int i; - for (i = 0; i < in->num_preprocessors; i++) - { - int cur_status = in_configure_effect_channels(in->preprocessors[i].effect_itfe, - channel_config); - if (cur_status != 0) { - ALOGI("%s: error %d configuring effect " - "%d with channels: [%04x][%04x]", - __func__, - cur_status, - i, - channel_config->main_channels, - channel_config->aux_channels); - status = cur_status; - } - } - } else if (effect != NULL && channel_config->aux_channels) { - /* if aux channels config did not change but aux channels are present, - * we still need to configure the effect being added */ - status = in_configure_effect_channels(effect, channel_config); - } - return status; -} - -static void in_update_aux_channels(struct espresso_stream_in *in, - effect_handle_t effect) -{ - uint32_t aux_channels; - channel_config_t channel_config; - int status; - - aux_channels = in_get_aux_channels(in); - - channel_config.main_channels = in->main_channels; - channel_config.aux_channels = aux_channels; - status = in_reconfigure_channels(in, - effect, - &channel_config, - (aux_channels != in->aux_channels)); - - if (status != 0) { - ALOGI("%s: in_reconfigure_channels error %d", __func__, status); - /* resetting aux channels configuration */ - aux_channels = 0; - channel_config.aux_channels = 0; - in_reconfigure_channels(in, effect, &channel_config, true); - } - if (in->aux_channels != aux_channels) { - in->aux_channels_changed = true; - in->aux_channels = aux_channels; - do_input_standby(in); - } -} - -static int in_add_audio_effect(const struct audio_stream *stream, - effect_handle_t effect) -{ - struct espresso_stream_in *in = (struct espresso_stream_in *)stream; - int status; - effect_descriptor_t desc; - - pthread_mutex_lock(&in->dev->lock); - pthread_mutex_lock(&in->lock); - if (in->num_preprocessors >= MAX_PREPROCESSORS) { - status = -ENOSYS; - goto exit; - } - - status = (*effect)->get_descriptor(effect, &desc); - if (status != 0) - goto exit; - - in->preprocessors[in->num_preprocessors].effect_itfe = effect; - /* add the supported channel of the effect in the channel_configs */ - in_read_audio_effect_channel_configs(in, &in->preprocessors[in->num_preprocessors]); - - in->num_preprocessors++; - - /* check compatibility between main channel supported and possible auxiliary channels */ - in_update_aux_channels(in, effect); - - ALOGV("%s: effect type: %08x", __func__, desc.type.timeLow); - - if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) { - in->need_echo_reference = true; - do_input_standby(in); - in_configure_reverse(in); - } - -exit: - - ALOGW_IF(status != 0, "%s: error %d", __func__, status); - pthread_mutex_unlock(&in->lock); - pthread_mutex_unlock(&in->dev->lock); - return status; -} - -static int in_remove_audio_effect(const struct audio_stream *stream, - effect_handle_t effect) -{ - struct espresso_stream_in *in = (struct espresso_stream_in *)stream; - int i; - int status = -EINVAL; - effect_descriptor_t desc; - - pthread_mutex_lock(&in->dev->lock); - pthread_mutex_lock(&in->lock); - if (in->num_preprocessors <= 0) { - status = -ENOSYS; - goto exit; - } - - for (i = 0; i < in->num_preprocessors; i++) { - if (status == 0) { /* status == 0 means an effect was removed from a previous slot */ - in->preprocessors[i - 1].effect_itfe = in->preprocessors[i].effect_itfe; - in->preprocessors[i - 1].channel_configs = in->preprocessors[i].channel_configs; - in->preprocessors[i - 1].num_channel_configs = in->preprocessors[i].num_channel_configs; - ALOGI("in_remove_audio_effect moving fx from %d to %d", i, i - 1); - continue; - } - if (in->preprocessors[i].effect_itfe == effect) { - ALOGI("in_remove_audio_effect found fx at index %d", i); - free(in->preprocessors[i].channel_configs); - status = 0; - } - } - - if (status != 0) - goto exit; - - in->num_preprocessors--; - /* if we remove one effect, at least the last preproc should be reset */ - in->preprocessors[in->num_preprocessors].num_channel_configs = 0; - in->preprocessors[in->num_preprocessors].effect_itfe = NULL; - in->preprocessors[in->num_preprocessors].channel_configs = NULL; - - - /* check compatibility between main channel supported and possible auxiliary channels */ - in_update_aux_channels(in, NULL); - - status = (*effect)->get_descriptor(effect, &desc); - if (status != 0) - goto exit; - - ALOGV("%s: effect type: %08x", __func__, desc.type.timeLow); - - if (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0) { - in->need_echo_reference = false; - do_input_standby(in); - } - -exit: - - ALOGW_IF(status != 0, "%s: error %d", __func__, status); - pthread_mutex_unlock(&in->lock); - pthread_mutex_unlock(&in->dev->lock); - return status; -} - -static int adev_open_output_stream(struct audio_hw_device *dev, - audio_io_handle_t handle, - audio_devices_t devices, - audio_output_flags_t flags, - struct audio_config *config, - struct audio_stream_out **stream_out) -{ - struct espresso_audio_device *ladev = (struct espresso_audio_device *)dev; - struct espresso_stream_out *out; - int ret; - int output_type; - - *stream_out = NULL; - - out = (struct espresso_stream_out *)calloc(1, sizeof(struct espresso_stream_out)); - if (!out) { - ALOGE("%s: out of memory!", __func__); - return -ENOMEM; - } - - out->sup_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO; - out->channel_mask = AUDIO_CHANNEL_OUT_STEREO; - - if (ladev->outputs[OUTPUT_DEEP_BUF] != NULL) { - ret = -ENOSYS; - ALOGW("%s: output not available!", __func__); - goto err_open; - } - output_type = OUTPUT_DEEP_BUF; - out->channel_mask = AUDIO_CHANNEL_OUT_STEREO; - out->stream.common.get_buffer_size = out_get_buffer_size_deep_buffer; - out->stream.common.get_sample_rate = out_get_sample_rate; - out->stream.get_latency = out_get_latency_deep_buffer; - out->stream.write = out_write_deep_buffer; - - ret = create_resampler(DEFAULT_OUT_SAMPLING_RATE, - MM_FULL_POWER_SAMPLING_RATE, - 2, - RESAMPLER_QUALITY_DEFAULT, - NULL, - &out->resampler); - if (ret != 0) { - ALOGE("%s: error on resampler create!", __func__); - goto err_open; - } - - out->stream.common.set_sample_rate = out_set_sample_rate; - 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.set_volume = out_set_volume; - out->stream.get_render_position = out_get_render_position; - - out->dev = ladev; - out->standby = 1; - - /* FIXME: when we support multiple output devices, we will want to - * do the following: - * adev->out_device = out->device; - * select_output_device(adev); - * This is because out_set_parameters() with a route is not - * guaranteed to be called after an output stream is opened. */ - - 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; - ladev->outputs[output_type] = out; - - return 0; - -err_open: - ALOGE("%s: error opening output stream", __func__); - free(out); - return ret; -} - -static void adev_close_output_stream(struct audio_hw_device *dev, - struct audio_stream_out *stream) -{ - struct espresso_audio_device *ladev = (struct espresso_audio_device *)dev; - struct espresso_stream_out *out = (struct espresso_stream_out *)stream; - int i; - - out_standby(&stream->common); - for (i = 0; i < OUTPUT_TOTAL; i++) { - if (ladev->outputs[i] == out) { - ladev->outputs[i] = NULL; - break; - } - } - - if (out->buffer) - free(out->buffer); - if (out->resampler) - release_resampler(out->resampler); - free(stream); -} - -static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs) -{ - struct espresso_audio_device *adev = (struct espresso_audio_device *)dev; - struct str_parms *parms; - char *str; - char value[32]; - int ret; - - parms = str_parms_create_str(kvpairs); - ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_TTY_MODE, value, sizeof(value)); - if (ret >= 0) { - int 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 - return -EINVAL; - - pthread_mutex_lock(&adev->lock); - if (tty_mode != adev->tty_mode) { - adev->tty_mode = tty_mode; - if (adev->mode == AUDIO_MODE_IN_CALL) - select_output_device(adev); - } - pthread_mutex_unlock(&adev->lock); - } - - ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value)); - if (ret >= 0) { - if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0) - adev->bluetooth_nrec = true; - else - adev->bluetooth_nrec = false; - } - - ret = str_parms_get_str(parms, "screen_off", 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_str(parms, "noise_suppression", value, sizeof(value)); - if (ret >= 0) { - if (strcmp(value, "on") == 0) { - ALOGE("%s: enabling two mic control", __func__); - ril_set_two_mic_control(&adev->ril, AUDIENCE, TWO_MIC_SOLUTION_ON); - /* sub mic */ - set_bigroute_by_array(adev->mixer, noise_suppression, 1); - } else { - ALOGE("%s: disabling two mic control", __func__); - ril_set_two_mic_control(&adev->ril, AUDIENCE, TWO_MIC_SOLUTION_OFF); - /* sub mic */ - set_bigroute_by_array(adev->mixer, noise_suppression_disable, 1); - } - } - - str_parms_destroy(parms); - return ret; -} - -static char * adev_get_parameters(const struct audio_hw_device *dev, - const char *keys) -{ - return strdup(""); -} - -static int adev_init_check(const struct audio_hw_device *dev) -{ - return 0; -} - -static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) -{ - struct espresso_audio_device *adev = (struct espresso_audio_device *)dev; - - adev->voice_volume = volume; - - if (adev->mode == AUDIO_MODE_IN_CALL) - ril_set_call_volume(&adev->ril, SOUND_TYPE_VOICE, volume); - - return 0; -} - -static int adev_set_master_volume(struct audio_hw_device *dev, float volume) -{ - return -ENOSYS; -} - -static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode) -{ - struct espresso_audio_device *adev = (struct espresso_audio_device *)dev; - - pthread_mutex_lock(&adev->lock); - if (adev->mode != mode) { - adev->mode = mode; - select_mode(adev); - } - pthread_mutex_unlock(&adev->lock); - - return 0; -} - -static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) -{ - struct espresso_audio_device *adev = (struct espresso_audio_device *)dev; - - adev->mic_mute = state; - - return 0; -} - -static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) -{ - struct espresso_audio_device *adev = (struct espresso_audio_device *)dev; - - *state = adev->mic_mute; - - return 0; -} - -static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, - const struct audio_config *config) -{ - size_t size; - int channel_count = popcount(config->channel_mask); - if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) - return 0; - - return get_input_buffer_size(config->sample_rate, config->format, channel_count); -} - -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) -{ - struct espresso_audio_device *ladev = (struct espresso_audio_device *)dev; - struct espresso_stream_in *in; - int ret; - - /* Respond with a request for stereo if a different format is given. */ - if (config->channel_mask != AUDIO_CHANNEL_IN_STEREO) { - config->channel_mask = AUDIO_CHANNEL_IN_STEREO; - return -EINVAL; - } - - int channel_count = popcount(config->channel_mask); - - *stream_in = NULL; - - if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0) - return -EINVAL; - - in = (struct espresso_stream_in *)calloc(1, sizeof(struct espresso_stream_in)); - if (!in) - return -ENOMEM; - - 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->requested_rate = config->sample_rate; - - memcpy(&in->config, &pcm_config_capture, sizeof(pcm_config_capture)); - in->config.channels = channel_count; - - in->main_channels = config->channel_mask; - - /* initialisation of preprocessor structure array is implicit with the calloc. - * same for in->aux_channels and in->aux_channels_changed */ - - if (in->requested_rate != in->config.rate) { - in->buf_provider.get_next_buffer = get_next_buffer; - in->buf_provider.release_buffer = release_buffer; - - ret = create_resampler(in->config.rate, - in->requested_rate, - in->config.channels, - RESAMPLER_QUALITY_DEFAULT, - &in->buf_provider, - &in->resampler); - if (ret != 0) { - ret = -EINVAL; - goto err; - } - } - - in->dev = ladev; - in->standby = 1; - in->device = devices & ~AUDIO_DEVICE_BIT_IN; - - *stream_in = &in->stream; - return 0; - -err: - if (in->resampler) - release_resampler(in->resampler); - - free(in); - return ret; -} - -static void adev_close_input_stream(struct audio_hw_device *dev, - struct audio_stream_in *stream) -{ - struct espresso_stream_in *in = (struct espresso_stream_in *)stream; - int i; - - in_standby(&stream->common); - - for (i = 0; i < in->num_preprocessors; i++) { - free(in->preprocessors[i].channel_configs); - } - - free(in->read_buf); - if (in->resampler) { - release_resampler(in->resampler); - } - if (in->proc_buf_in) - free(in->proc_buf_in); - if (in->proc_buf_out) - free(in->proc_buf_out); - if (in->ref_buf) - free(in->ref_buf); - - free(stream); - return; -} - -static int adev_dump(const audio_hw_device_t *device, int fd) -{ - return 0; -} - -static int adev_close(hw_device_t *device) -{ - struct espresso_audio_device *adev = (struct espresso_audio_device *)device; - - /* RIL */ - ril_close(&adev->ril); - - mixer_close(adev->mixer); - free(device); - return 0; -} - -struct config_parse_state { - struct espresso_audio_device *adev; - struct espresso_dev_cfg *dev; - bool on; - - struct route_setting *path; - unsigned int path_len; -}; - -static const struct { - int mask; - const char *name; -} dev_names[] = { - { AUDIO_DEVICE_OUT_SPEAKER, "speaker" }, - { AUDIO_DEVICE_OUT_WIRED_HEADSET | AUDIO_DEVICE_OUT_WIRED_HEADPHONE, "headphone" }, - { AUDIO_DEVICE_OUT_EARPIECE, "earpiece" }, - { AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET, "dock" }, - { AUDIO_DEVICE_OUT_DGTL_DOCK_HEADSET, "dock" }, - { AUDIO_DEVICE_OUT_ALL_SCO, "sco-out" }, - - { AUDIO_DEVICE_IN_BUILTIN_MIC, "builtin-mic" }, - { AUDIO_DEVICE_IN_BACK_MIC, "back-mic" }, - { AUDIO_DEVICE_IN_WIRED_HEADSET, "headset-in" }, - { AUDIO_DEVICE_IN_ALL_SCO, "sco-in" }, -}; - -static void adev_config_start(void *data, const XML_Char *elem, - const XML_Char **attr) -{ - struct config_parse_state *s = data; - struct espresso_dev_cfg *dev_cfg; - const XML_Char *name = NULL; - const XML_Char *val = NULL; - unsigned int i, j; - - for (i = 0; attr[i]; i += 2) { - if (strcmp(attr[i], "name") == 0) - name = attr[i + 1]; - - if (strcmp(attr[i], "val") == 0) - val = attr[i + 1]; - } - - if (strcmp(elem, "device") == 0) { - if (!name) { - ALOGE("Unnamed device\n"); - return; - } - - for (i = 0; i < sizeof(dev_names) / sizeof(dev_names[0]); i++) { - if (strcmp(dev_names[i].name, name) == 0) { - ALOGI("Allocating device %s\n", name); - dev_cfg = realloc(s->adev->dev_cfgs, - (s->adev->num_dev_cfgs + 1) - * sizeof(*dev_cfg)); - if (!dev_cfg) { - ALOGE("Unable to allocate dev_cfg\n"); - return; - } - - s->dev = &dev_cfg[s->adev->num_dev_cfgs]; - memset(s->dev, 0, sizeof(*s->dev)); - s->dev->mask = dev_names[i].mask; - - s->adev->dev_cfgs = dev_cfg; - s->adev->num_dev_cfgs++; - } - } - - } else if (strcmp(elem, "path") == 0) { - if (s->path_len) - ALOGW("Nested paths\n"); - - /* If this a path for a device it must have a role */ - if (s->dev) { - /* Need to refactor a bit... */ - if (strcmp(name, "on") == 0) { - s->on = true; - } else if (strcmp(name, "off") == 0) { - s->on = false; - } else { - ALOGW("Unknown path name %s\n", name); - } - } - - } else if (strcmp(elem, "ctl") == 0) { - struct route_setting *r; - - if (!name) { - ALOGE("Unnamed control\n"); - return; - } - - if (!val) { - ALOGE("No value specified for %s\n", name); - return; - } - - ALOGV("Parsing control %s => %s\n", name, val); - - r = realloc(s->path, sizeof(*r) * (s->path_len + 1)); - if (!r) { - ALOGE("Out of memory handling %s => %s\n", name, val); - return; - } - - r[s->path_len].ctl_name = strdup(name); - r[s->path_len].strval = NULL; - - /* This can be fooled but it'll do */ - r[s->path_len].intval = atoi(val); - if (!r[s->path_len].intval && strcmp(val, "0") != 0) - r[s->path_len].strval = strdup(val); - - s->path = r; - s->path_len++; - } -} - -static void adev_config_end(void *data, const XML_Char *name) -{ - struct config_parse_state *s = data; - unsigned int i; - - if (strcmp(name, "path") == 0) { - if (!s->path_len) - ALOGW("Empty path\n"); - - if (!s->dev) { - ALOGV("Applying %d element default route\n", s->path_len); - - set_route_by_array(s->adev->mixer, s->path, s->path_len); - - for (i = 0; i < s->path_len; i++) { - free(s->path[i].ctl_name); - free(s->path[i].strval); - } - - free(s->path); - - /* Refactor! */ - } else if (s->on) { - ALOGV("%d element on sequence\n", s->path_len); - s->dev->on = s->path; - s->dev->on_len = s->path_len; - - } else { - ALOGV("%d element off sequence\n", s->path_len); - - /* Apply it, we'll reenable anything that's wanted later */ - set_route_by_array(s->adev->mixer, s->path, s->path_len); - - s->dev->off = s->path; - s->dev->off_len = s->path_len; - } - - s->path_len = 0; - s->path = NULL; - - } else if (strcmp(name, "device") == 0) { - s->dev = NULL; - } -} - -static int adev_config_parse(struct espresso_audio_device *adev) -{ - struct config_parse_state s; - FILE *f; - XML_Parser p; - char property[PROPERTY_VALUE_MAX]; - char file[80]; - int ret = 0; - bool eof = false; - int len; - - property_get("ro.product.device", property, "tiny_hw"); - snprintf(file, sizeof(file), "/system/etc/sound/%s", property); - - ALOGV("Reading configuration from %s\n", file); - f = fopen(file, "r"); - if (!f) { - ALOGE("Failed to open %s\n", file); - return -ENODEV; - } - - p = XML_ParserCreate(NULL); - if (!p) { - ALOGE("Failed to create XML parser\n"); - ret = -ENOMEM; - goto out; - } - - memset(&s, 0, sizeof(s)); - s.adev = adev; - XML_SetUserData(p, &s); - - XML_SetElementHandler(p, adev_config_start, adev_config_end); - - while (!eof) { - len = fread(file, 1, sizeof(file), f); - if (ferror(f)) { - ALOGE("I/O error reading config\n"); - ret = -EIO; - goto out_parser; - } - eof = feof(f); - - if (XML_Parse(p, file, len, eof) == XML_STATUS_ERROR) { - ALOGE("Parse error at line %u:\n%s\n", - (unsigned int)XML_GetCurrentLineNumber(p), - XML_ErrorString(XML_GetErrorCode(p))); - ret = -EINVAL; - goto out_parser; - } - } - - out_parser: - XML_ParserFree(p); - out: - fclose(f); - - return ret; -} - -static int adev_open(const hw_module_t* module, const char* name, - hw_device_t** device) -{ - struct espresso_audio_device *adev; - int i, ret; - - if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) - return -EINVAL; - - adev = calloc(1, sizeof(struct espresso_audio_device)); - if (!adev) - return -ENOMEM; - - adev->hw_device.common.tag = HARDWARE_DEVICE_TAG; - adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0; - adev->hw_device.common.module = (struct hw_module_t *) module; - adev->hw_device.common.close = adev_close; - - adev->hw_device.init_check = adev_init_check; - adev->hw_device.set_voice_volume = adev_set_voice_volume; - adev->hw_device.set_master_volume = adev_set_master_volume; - adev->hw_device.set_mode = adev_set_mode; - adev->hw_device.set_mic_mute = adev_set_mic_mute; - adev->hw_device.get_mic_mute = adev_get_mic_mute; - adev->hw_device.set_parameters = adev_set_parameters; - adev->hw_device.get_parameters = adev_get_parameters; - adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size; - adev->hw_device.open_output_stream = adev_open_output_stream; - adev->hw_device.close_output_stream = adev_close_output_stream; - adev->hw_device.open_input_stream = adev_open_input_stream; - adev->hw_device.close_input_stream = adev_close_input_stream; - adev->hw_device.dump = adev_dump; - - adev->mixer = mixer_open(CARD_DEFAULT); - if (!adev->mixer) { - free(adev); - ALOGE("Unable to open the mixer, aborting."); - return -EINVAL; - } - - ret = adev_config_parse(adev); - if (ret != 0) - goto err_mixer; - - /* Set the default route before the PCM stream is opened */ - pthread_mutex_init(&adev->lock, NULL); - adev->mode = AUDIO_MODE_NORMAL; - adev->out_device = AUDIO_DEVICE_OUT_SPEAKER; - adev->in_device = AUDIO_DEVICE_IN_BUILTIN_MIC & ~AUDIO_DEVICE_BIT_IN; - select_devices(adev); - - for (i = 0; i < OUTPUT_TOTAL; i++) { - adev->outputs[i] = NULL; - } - - adev->pcm_modem_dl = NULL; - adev->pcm_modem_ul = NULL; - adev->pcm_bt_dl = NULL; - adev->pcm_bt_ul = NULL; - adev->voice_volume = 1.0f; - adev->tty_mode = TTY_MODE_OFF; - adev->bluetooth_nrec = true; - adev->wb_amr = 0; - - /* RIL */ - ril_open(&adev->ril); - pthread_mutex_unlock(&adev->lock); - /* register callback for wideband AMR setting */ - ril_register_set_wb_amr_callback(audio_set_wb_amr_callback, (void *)adev); - - *device = &adev->hw_device.common; - - return 0; - -err_mixer: - mixer_close(adev->mixer); -err: - return -EINVAL; -} - -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 = "Piranha audio HW HAL", - .author = "The CyanogenMod Project", - .methods = &hal_module_methods, - }, -}; diff --git a/audio/audio_hw.h b/audio/audio_hw.h deleted file mode 100644 index d2b2e0e..0000000 --- a/audio/audio_hw.h +++ /dev/null @@ -1,228 +0,0 @@ -/* - * Copyright (C) 2011 The Android Open Source Project - * Copyright (C) 2012 Wolfson Microelectronics plc - * Copyright (C) 2012 The CyanogenMod Project - * Daniel Hillenbrand <codeworkx@cyanogenmod.com> - * Guillaume "XpLoDWilD" Lesniak <xplodgui@gmail.com> - * - * 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. - */ - -/* ALSA cards for WM1811 */ -#define CARD_DEFAULT 0 - -#define PORT_PLAYBACK 0 -#define PORT_CAPTURE 0 -#define PORT_MODEM 1 -#define PORT_BT 2 - -#define PCM_WRITE pcm_write - -#define PLAYBACK_PERIOD_SIZE 880 -#define PLAYBACK_PERIOD_COUNT 8 -#define PLAYBACK_SHORT_PERIOD_COUNT 2 - -#define CAPTURE_PERIOD_SIZE 1056 -#define CAPTURE_PERIOD_COUNT 2 - -#define SHORT_PERIOD_SIZE 192 - -// -// deep buffer -// -/* screen on */ -#define DEEP_BUFFER_SHORT_PERIOD_SIZE 1056 -#define PLAYBACK_DEEP_BUFFER_SHORT_PERIOD_COUNT 4 -/* screen off */ -#define DEEP_BUFFER_LONG_PERIOD_SIZE 880 -#define PLAYBACK_DEEP_BUFFER_LONG_PERIOD_COUNT 8 - - -/* minimum sleep time in out_write() when write threshold is not reached */ -#define MIN_WRITE_SLEEP_US 5000 - -#define RESAMPLER_BUFFER_FRAMES (PLAYBACK_PERIOD_SIZE * 2) -#define RESAMPLER_BUFFER_SIZE (4 * RESAMPLER_BUFFER_FRAMES) - -#define DEFAULT_OUT_SAMPLING_RATE 44100 -#define MM_LOW_POWER_SAMPLING_RATE 44100 -#define MM_FULL_POWER_SAMPLING_RATE 44100 -#define DEFAULT_IN_SAMPLING_RATE 44100 - -/* sampling rate when using VX port for narrow band */ -#define VX_NB_SAMPLING_RATE 8000 -/* sampling rate when using VX port for wide band */ -#define VX_WB_SAMPLING_RATE 16000 - -/* product-specific defines */ -#define PRODUCT_DEVICE_PROPERTY "ro.product.device" -#define PRODUCT_NAME_PROPERTY "ro.product.name" - -#define ARRAY_SIZE(a) (sizeof(a) / sizeof((a)[0])) - -#define STRING_TO_ENUM(string) { #string, string } - -struct string_to_enum { - const char *name; - uint32_t value; -}; - -const struct string_to_enum out_channels_name_to_enum_table[] = { - STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO), - STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1), - STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1), -}; - -enum pcm_type { - PCM_NORMAL = 0, - PCM_SPDIF, - PCM_HDMI, - PCM_TOTAL, -}; - -enum output_type { - OUTPUT_DEEP_BUF, // deep PCM buffers output stream - OUTPUT_LOW_LATENCY, // low latency output stream - OUTPUT_HDMI, - OUTPUT_TOTAL -}; - -enum tty_modes { - TTY_MODE_OFF, - TTY_MODE_VCO, - TTY_MODE_HCO, - TTY_MODE_FULL -}; - -struct route_setting -{ - char *ctl_name; - int intval; - char *strval; -}; - -struct route_setting voicecall_default[] = { - { .ctl_name = "HP Output Mode", .intval = 0, }, - { .ctl_name = "AIF2 Mode", .intval = 0, }, - { .ctl_name = "AIF2DACL Source", .intval = 0, }, - { .ctl_name = "AIF2DACR Source", .intval = 0, }, - { .ctl_name = "DAC1L Mixer AIF1.1 Switch", .intval = 1, }, - { .ctl_name = "DAC1R Mixer AIF1.1 Switch", .intval = 1, }, - { .ctl_name = "DAC1L Mixer AIF2 Switch", .intval = 1, }, - { .ctl_name = "DAC1R Mixer AIF2 Switch", .intval = 1, }, - { .ctl_name = "AIF2DAC Mux", .strval = "AIF2DACDAT", }, - { .ctl_name = NULL, }, -}; - -struct route_setting voicecall_default_disable[] = { - { .ctl_name = "AIF2 Mode", .intval = 0, }, - { .ctl_name = "AIF2DACL Source", .intval = 0, }, - { .ctl_name = "AIF2DACR Source", .intval = 1, }, - { .ctl_name = "DAC1L Mixer AIF2 Switch", .intval = 0, }, - { .ctl_name = "DAC1R Mixer AIF2 Switch", .intval = 0, }, - { .ctl_name = "AIF2DAC Mux", .strval = "AIF2DACDAT", }, - { .ctl_name = NULL, }, -}; - -struct route_setting default_input[] = { - { .ctl_name = "Main Mic Switch", .intval = 1, }, - { .ctl_name = "IN1L Volume", .intval = 30, }, - { .ctl_name = "MIXINL IN1L Switch", .intval = 1, }, - { .ctl_name = "MIXINL IN1L Volume", .intval = 0, }, - { .ctl_name = "AIF1ADC1 HPF Mode", .intval = 1, }, - { .ctl_name = "AIF1ADC1 HPF Switch", .intval = 1, }, - { .ctl_name = "AIF2DAC2L Mixer Left Sidetone Switch", .intval = 1, }, - { .ctl_name = "AIF2DAC2L Mixer Right Sidetone Switch", .intval = 0, }, - { .ctl_name = "AIF2DAC2R Mixer Left Sidetone Switch", .intval = 1, }, - { .ctl_name = "AIF2DAC2R Mixer Right Sidetone Switch", .intval = 0, }, - { .ctl_name = NULL, }, -}; - -struct route_setting default_input_disable[] = { - { .ctl_name = "Main Mic Switch", .intval = 0, }, - { .ctl_name = "MIXINL IN1L Switch", .intval = 0, }, - { .ctl_name = "AIF1ADC1 HPF Mode", .intval = 0, }, - { .ctl_name = "AIF1ADC1 HPF Switch", .intval = 0, }, - { .ctl_name = NULL, }, -}; - -struct route_setting noise_suppression[] = { - { .ctl_name = "Sub Mic Switch", .intval = 1, }, - { .ctl_name = "AIF1ADCR Source", .intval = 1, }, - { .ctl_name = "MIXINR IN2R Switch", .intval = 1, }, - { .ctl_name = "IN2R Volume", .intval = 25, }, - { .ctl_name = NULL, }, -}; - -struct route_setting noise_suppression_disable[] = { - { .ctl_name = "Sub Mic Switch", .intval = 0, }, - { .ctl_name = "AIF1ADCR Source", .intval = 0, }, - { .ctl_name = "MIXINR IN2R Switch", .intval = 0, }, - { .ctl_name = "IN2R Volume", .intval = 11, }, - { .ctl_name = NULL, }, -}; - -struct route_setting headset_input[] = { - { .ctl_name = "Headset Mic Switch", .intval = 1, }, - { .ctl_name = "MIXINR IN1R Switch", .intval = 1, }, - { .ctl_name = "AIF2DAC2L Mixer Left Sidetone Switch", .intval = 0, }, - { .ctl_name = "AIF2DAC2L Mixer Right Sidetone Switch", .intval = 1, }, - { .ctl_name = "AIF2DAC2R Mixer Left Sidetone Switch", .intval = 0, }, - { .ctl_name = "AIF2DAC2R Mixer Right Sidetone Switch", .intval = 1, }, - { .ctl_name = NULL, }, -}; - -struct route_setting headset_input_disable[] = { - { .ctl_name = "Headset Mic Switch", .intval = 0, }, - { .ctl_name = "MIXINR IN1R Switch", .intval = 0, }, - { .ctl_name = NULL, }, -}; - -struct route_setting bt_output[] = { - { .ctl_name = "AIF2DAC2L Mixer AIF2 Switch", .intval = 1, }, - { .ctl_name = "AIF2DAC2R Mixer AIF2 Switch", .intval = 1, }, - { .ctl_name = "AIF2DAC Volume", .intval = 96, }, - { .ctl_name = "DAC2 Volume", .intval = 96, }, - { .ctl_name = "AIF2ADC Volume", .intval = 96, }, - { .ctl_name = NULL, }, -}; - -struct route_setting bt_output_disable[] = { - { .ctl_name = "AIF2DAC2L Mixer AIF2 Switch", .intval = 0, }, - { .ctl_name = "AIF2DAC2R Mixer AIF2 Switch", .intval = 0, }, - { .ctl_name = NULL, }, -}; - -struct route_setting bt_input[] = { - { .ctl_name = "AIF2ADC Mux", .intval = 1, }, - { .ctl_name = "AIF2DACR Source", .intval = 1, }, - { .ctl_name = "HP Switch", .intval = 0, }, - { .ctl_name = "DAC1L Mixer AIF2 Switch", .intval = 1, }, - { .ctl_name = "DAC1R Mixer AIF2 Switch", .intval = 1, }, - { .ctl_name = "AIF1ADC1R Mixer AIF2 Switch", .intval = 0, }, - { .ctl_name = "AIF1ADC1L Mixer AIF2 Switch", .intval = 0, }, - { .ctl_name = "AIF2DAC2L Mixer Left Sidetone Switch", .intval = 1, }, - { .ctl_name = "AIF2DAC2L Mixer Right Sidetone Switch", .intval = 0, }, - { .ctl_name = "AIF2DAC2R Mixer Left Sidetone Switch", .intval = 0, }, - { .ctl_name = "AIF2DAC2R Mixer Right Sidetone Switch", .intval = 1, }, - { .ctl_name = "AIF1ADC1R Mixer AIF2 Switch", .intval = 0, }, - { .ctl_name = "AIF1ADC1L Mixer AIF2 Switch", .intval = 0, }, - { .ctl_name = "MIXINL IN1L Switch", .intval = 0, }, - { .ctl_name = NULL, }, -}; - -struct route_setting bt_input_disable[] = { - { .ctl_name = "AIF2ADC Mux", .intval = 0, }, - { .ctl_name = "HP Switch", .intval = 1, }, - { .ctl_name = NULL, }, -}; diff --git a/audio/ril_interface.c b/audio/ril_interface.c deleted file mode 100755 index a77efc0..0000000 --- a/audio/ril_interface.c +++ /dev/null @@ -1,193 +0,0 @@ -/* - * Copyright (C) 2011 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 ALOG_TAG "audio_hw_primary" -/*#define ALOG_NDEBUG 0*/ - -#include <dlfcn.h> -#include <stdlib.h> - -#include <utils/Log.h> -#include <cutils/properties.h> - -#include "ril_interface.h" - -#define VOLUME_STEPS_DEFAULT "5" -#define VOLUME_STEPS_PROPERTY "ro.config.vc_call_vol_steps" - -/* Function pointers */ -void *(*_ril_open_client)(void); -int (*_ril_close_client)(void *); -int (*_ril_connect)(void *); -int (*_ril_is_connected)(void *); -int (*_ril_disconnect)(void *); -int (*_ril_set_call_volume)(void *, enum ril_sound_type, int); -int (*_ril_set_call_audio_path)(void *, enum ril_audio_path); -int (*_ril_set_call_clock_sync)(void *, enum ril_clock_state); -int (*_ril_set_two_mic_control)(void *, enum ril_two_mic_device, enum ril_two_mic_state); -int (*_ril_register_unsolicited_handler)(void *, int, void *); -int (*_ril_get_wb_amr)(void *, void *); - -/* Audio WB AMR callback */ -void (*_audio_set_wb_amr_callback)(void *, int); -void *callback_data = NULL; - -void ril_register_set_wb_amr_callback(void *function, void *data) -{ - _audio_set_wb_amr_callback = function; - callback_data = data; -} - -/* This is the callback function that the RIL uses to -set the wideband AMR state */ -static int ril_set_wb_amr_callback(void *ril_client, - const void *data, - size_t datalen) -{ - int enable = ((int *)data)[0]; - - if (!callback_data || !_audio_set_wb_amr_callback) - return -1; - - _audio_set_wb_amr_callback(callback_data, enable); - - return 0; -} - -static int ril_connect_if_required(struct ril_handle *ril) -{ - if (_ril_is_connected(ril->client)) - return 0; - - if (_ril_connect(ril->client) != RIL_CLIENT_ERR_SUCCESS) { - ALOGE("ril_connect() failed"); - return -1; - } - - /* get wb amr status to set pcm samplerate depending on - wb amr status when ril is connected. */ - if(_ril_get_wb_amr) - _ril_get_wb_amr(ril->client, ril_set_wb_amr_callback); - - return 0; -} - -int ril_open(struct ril_handle *ril) -{ - char property[PROPERTY_VALUE_MAX]; - - if (!ril) - return -1; - - ril->handle = dlopen(RIL_CLIENT_LIBPATH, RTLD_NOW); - - if (!ril->handle) { - ALOGE("Cannot open '%s'", RIL_CLIENT_LIBPATH); - return -1; - } - - _ril_open_client = dlsym(ril->handle, "OpenClient_RILD"); - _ril_close_client = dlsym(ril->handle, "CloseClient_RILD"); - _ril_connect = dlsym(ril->handle, "Connect_RILD"); - _ril_is_connected = dlsym(ril->handle, "isConnected_RILD"); - _ril_disconnect = dlsym(ril->handle, "Disconnect_RILD"); - _ril_set_call_volume = dlsym(ril->handle, "SetCallVolume"); - _ril_set_call_audio_path = dlsym(ril->handle, "SetCallAudioPath"); - _ril_set_call_clock_sync = dlsym(ril->handle, "SetCallClockSync"); - _ril_set_two_mic_control = dlsym(ril->handle, "SetTwoMicControl"); - _ril_register_unsolicited_handler = dlsym(ril->handle, - "RegisterUnsolicitedHandler"); - /* since this function is not supported in all RILs, don't require it */ - _ril_get_wb_amr = dlsym(ril->handle, "GetWB_AMR"); - - if (!_ril_open_client || !_ril_close_client || !_ril_connect || - !_ril_is_connected || !_ril_disconnect || !_ril_set_call_volume || - !_ril_set_call_audio_path || !_ril_set_two_mic_control || !_ril_set_call_clock_sync || - !_ril_register_unsolicited_handler) { - ALOGE("Cannot get symbols from '%s'", RIL_CLIENT_LIBPATH); - dlclose(ril->handle); - return -1; - } - - ril->client = _ril_open_client(); - if (!ril->client) { - ALOGE("ril_open_client() failed"); - dlclose(ril->handle); - return -1; - } - - /* register the wideband AMR callback */ - _ril_register_unsolicited_handler(ril->client, RIL_UNSOL_WB_AMR_STATE, - ril_set_wb_amr_callback); - - property_get(VOLUME_STEPS_PROPERTY, property, VOLUME_STEPS_DEFAULT); - ril->volume_steps_max = atoi(property); - /* this catches the case where VOLUME_STEPS_PROPERTY does not contain - an integer */ - if (ril->volume_steps_max == 0) - ril->volume_steps_max = atoi(VOLUME_STEPS_DEFAULT); - - return 0; -} - -int ril_close(struct ril_handle *ril) -{ - if (!ril || !ril->handle || !ril->client) - return -1; - - if ((_ril_disconnect(ril->client) != RIL_CLIENT_ERR_SUCCESS) || - (_ril_close_client(ril->client) != RIL_CLIENT_ERR_SUCCESS)) { - ALOGE("ril_disconnect() or ril_close_client() failed"); - return -1; - } - - dlclose(ril->handle); - return 0; -} - -int ril_set_call_volume(struct ril_handle *ril, enum ril_sound_type sound_type, - float volume) -{ - if (ril_connect_if_required(ril)) - return 0; - - return _ril_set_call_volume(ril->client, sound_type, - (int)(volume * ril->volume_steps_max)); -} - -int ril_set_call_audio_path(struct ril_handle *ril, enum ril_audio_path path) -{ - if (ril_connect_if_required(ril)) - return 0; - - return _ril_set_call_audio_path(ril->client, path); -} - -int ril_set_call_clock_sync(struct ril_handle *ril, enum ril_clock_state state) -{ - if (ril_connect_if_required(ril)) - return 0; - - return _ril_set_call_clock_sync(ril->client, state); -} - -int ril_set_two_mic_control(struct ril_handle *ril, enum ril_two_mic_device device, enum ril_two_mic_state state) -{ - if (ril_connect_if_required(ril)) - return 0; - - return _ril_set_two_mic_control(ril->client, device, state); -} diff --git a/audio/ril_interface.h b/audio/ril_interface.h deleted file mode 100755 index 4b401aa..0000000 --- a/audio/ril_interface.h +++ /dev/null @@ -1,83 +0,0 @@ -/* - * Copyright (C) 2011 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. - */ - -#ifndef RIL_INTERFACE_H -#define RIL_INTERFACE_H - -#define RIL_CLIENT_LIBPATH "libsecril-client.so" - -#define RIL_CLIENT_ERR_SUCCESS 0 -#define RIL_CLIENT_ERR_AGAIN 1 -#define RIL_CLIENT_ERR_INIT 2 // Client is not initialized -#define RIL_CLIENT_ERR_INVAL 3 // Invalid value -#define RIL_CLIENT_ERR_CONNECT 4 // Connection error -#define RIL_CLIENT_ERR_IO 5 // IO error -#define RIL_CLIENT_ERR_RESOURCE 6 // Resource not available -#define RIL_CLIENT_ERR_UNKNOWN 7 - -#define RIL_OEM_UNSOL_RESPONSE_BASE 11000 // RIL response base index -#define RIL_UNSOL_WB_AMR_STATE \ - (RIL_OEM_UNSOL_RESPONSE_BASE + 17) // RIL AMR state index - -struct ril_handle -{ - void *handle; - void *client; - int volume_steps_max; -}; - -enum ril_sound_type { - SOUND_TYPE_VOICE, - SOUND_TYPE_SPEAKER, - SOUND_TYPE_HEADSET, - SOUND_TYPE_BTVOICE -}; - -enum ril_audio_path { - SOUND_AUDIO_PATH_HANDSET, - SOUND_AUDIO_PATH_HEADSET, - SOUND_AUDIO_PATH_SPEAKER, - SOUND_AUDIO_PATH_BLUETOOTH, - SOUND_AUDIO_PATH_BLUETOOTH_NO_NR, - SOUND_AUDIO_PATH_HEADPHONE -}; - -enum ril_clock_state { - SOUND_CLOCK_STOP, - SOUND_CLOCK_START -}; - -enum ril_two_mic_device { - AUDIENCE, - FORTEMEDIA -}; - -enum ril_two_mic_state { - TWO_MIC_SOLUTION_OFF, - TWO_MIC_SOLUTION_ON -}; - -/* Function prototypes */ -int ril_open(struct ril_handle *ril); -int ril_close(struct ril_handle *ril); -int ril_set_call_volume(struct ril_handle *ril, enum ril_sound_type sound_type, - float volume); -int ril_set_call_audio_path(struct ril_handle *ril, enum ril_audio_path path); -int ril_set_call_clock_sync(struct ril_handle *ril, enum ril_clock_state state); -void ril_register_set_wb_amr_callback(void *function, void *data); -int ril_set_two_mic_control(struct ril_handle *ril, enum ril_two_mic_device device, enum ril_two_mic_state state); -#endif - |