audio: msm8909w caf release LW.BR.1.0-00410-8x09w.0
MSM8909w Audio HAL code copied from CAF release
LW.BR.1.0-00410-8x09w.0
dbcce50 hal: Port wcd9326 changes to 8909
410c530 hal: update error handling for pcm_prepare failures
ff79309 hal: fix compilation issues with audio FM extention
762d7eb policy_hal: add support for fm device loopback
7c418f9 audio_policy: modify few methods to appropriately override base
8b12163 audio: Add support to enable split A2DP
a0559fa Revert "Revert "policy_hal: Function prototype correction for custom policy"."
Fixed makefiles to be compatible with PDK without kernel source
Change-Id: I9c6f2139adee62426b877516deeb41d4ed8052b2
diff --git a/msm8909/hal/audio_hw.c b/msm8909/hal/audio_hw.c
new file mode 100644
index 0000000..1c177fc
--- /dev/null
+++ b/msm8909/hal/audio_hw.c
@@ -0,0 +1,3539 @@
+/*
+ * Copyright (c) 2013-2015, The Linux Foundation. All rights reserved.
+ * Not a Contribution.
+ *
+ * Copyright (C) 2013 The Android Open Source Project
+ *
+ * Licensed under the Apache License, Version 2.0 (the "License");
+ * you may not use this file except in compliance with the License.
+ * You may obtain a copy of the License at
+ *
+ * http://www.apache.org/licenses/LICENSE-2.0
+ *
+ * Unless required by applicable law or agreed to in writing, software
+ * distributed under the License is distributed on an "AS IS" BASIS,
+ * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
+ * See the License for the specific language governing permissions and
+ * limitations under the License.
+ */
+
+#define LOG_TAG "audio_hw_primary"
+/*#define LOG_NDEBUG 0*/
+/*#define VERY_VERY_VERBOSE_LOGGING*/
+#ifdef VERY_VERY_VERBOSE_LOGGING
+#define ALOGVV ALOGV
+#else
+#define ALOGVV(a...) do { } while(0)
+#endif
+
+#include <errno.h>
+#include <pthread.h>
+#include <stdint.h>
+#include <sys/time.h>
+#include <stdlib.h>
+#include <math.h>
+#include <dlfcn.h>
+#include <sys/resource.h>
+#include <sys/prctl.h>
+
+#include <cutils/log.h>
+#include <cutils/str_parms.h>
+#include <cutils/properties.h>
+#include <cutils/atomic.h>
+#include <cutils/sched_policy.h>
+
+#include <hardware/audio_effect.h>
+#include <system/thread_defs.h>
+#include <audio_effects/effect_aec.h>
+#include <audio_effects/effect_ns.h>
+#include "audio_hw.h"
+#include "platform_api.h"
+#include <platform.h>
+#include "audio_extn.h"
+#include "voice_extn.h"
+
+#include "sound/compress_params.h"
+#include "sound/asound.h"
+
+#define COMPRESS_OFFLOAD_NUM_FRAGMENTS 4
+/* ToDo: Check and update a proper value in msec */
+#define COMPRESS_OFFLOAD_PLAYBACK_LATENCY 96
+#define COMPRESS_PLAYBACK_VOLUME_MAX 0x2000
+
+#define PROXY_OPEN_RETRY_COUNT 100
+#define PROXY_OPEN_WAIT_TIME 20
+
+#define USECASE_AUDIO_PLAYBACK_PRIMARY USECASE_AUDIO_PLAYBACK_DEEP_BUFFER
+
+static unsigned int configured_low_latency_capture_period_size =
+ LOW_LATENCY_CAPTURE_PERIOD_SIZE;
+
+struct pcm_config pcm_config_deep_buffer = {
+ .channels = 2,
+ .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
+ .period_size = DEEP_BUFFER_OUTPUT_PERIOD_SIZE,
+ .period_count = DEEP_BUFFER_OUTPUT_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4,
+ .stop_threshold = INT_MAX,
+ .avail_min = DEEP_BUFFER_OUTPUT_PERIOD_SIZE / 4,
+};
+
+struct pcm_config pcm_config_low_latency = {
+ .channels = 2,
+ .rate = DEFAULT_OUTPUT_SAMPLING_RATE,
+ .period_size = LOW_LATENCY_OUTPUT_PERIOD_SIZE,
+ .period_count = LOW_LATENCY_OUTPUT_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
+ .stop_threshold = INT_MAX,
+ .avail_min = LOW_LATENCY_OUTPUT_PERIOD_SIZE / 4,
+};
+
+struct pcm_config pcm_config_hdmi_multi = {
+ .channels = HDMI_MULTI_DEFAULT_CHANNEL_COUNT, /* changed when the stream is opened */
+ .rate = DEFAULT_OUTPUT_SAMPLING_RATE, /* changed when the stream is opened */
+ .period_size = HDMI_MULTI_PERIOD_SIZE,
+ .period_count = HDMI_MULTI_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = 0,
+ .stop_threshold = INT_MAX,
+ .avail_min = 0,
+};
+
+struct pcm_config pcm_config_audio_capture = {
+ .channels = 2,
+ .period_count = AUDIO_CAPTURE_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+};
+
+#define AFE_PROXY_CHANNEL_COUNT 2
+#define AFE_PROXY_SAMPLING_RATE 48000
+
+#define AFE_PROXY_PLAYBACK_PERIOD_SIZE 768
+#define AFE_PROXY_PLAYBACK_PERIOD_COUNT 4
+
+struct pcm_config pcm_config_afe_proxy_playback = {
+ .channels = AFE_PROXY_CHANNEL_COUNT,
+ .rate = AFE_PROXY_SAMPLING_RATE,
+ .period_size = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
+ .period_count = AFE_PROXY_PLAYBACK_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
+ .stop_threshold = INT_MAX,
+ .avail_min = AFE_PROXY_PLAYBACK_PERIOD_SIZE,
+};
+
+#define AFE_PROXY_RECORD_PERIOD_SIZE 768
+#define AFE_PROXY_RECORD_PERIOD_COUNT 4
+
+struct pcm_config pcm_config_afe_proxy_record = {
+ .channels = AFE_PROXY_CHANNEL_COUNT,
+ .rate = AFE_PROXY_SAMPLING_RATE,
+ .period_size = AFE_PROXY_RECORD_PERIOD_SIZE,
+ .period_count = AFE_PROXY_RECORD_PERIOD_COUNT,
+ .format = PCM_FORMAT_S16_LE,
+ .start_threshold = AFE_PROXY_RECORD_PERIOD_SIZE,
+ .stop_threshold = INT_MAX,
+ .avail_min = AFE_PROXY_RECORD_PERIOD_SIZE,
+};
+
+const char * const use_case_table[AUDIO_USECASE_MAX] = {
+ [USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = "deep-buffer-playback",
+ [USECASE_AUDIO_PLAYBACK_LOW_LATENCY] = "low-latency-playback",
+ [USECASE_AUDIO_PLAYBACK_MULTI_CH] = "multi-channel-playback",
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD] = "compress-offload-playback",
+#ifdef MULTIPLE_OFFLOAD_ENABLED
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD2] = "compress-offload-playback2",
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD3] = "compress-offload-playback3",
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD4] = "compress-offload-playback4",
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD5] = "compress-offload-playback5",
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD6] = "compress-offload-playback6",
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD7] = "compress-offload-playback7",
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD8] = "compress-offload-playback8",
+ [USECASE_AUDIO_PLAYBACK_OFFLOAD9] = "compress-offload-playback9",
+#endif
+ [USECASE_AUDIO_RECORD] = "audio-record",
+ [USECASE_AUDIO_RECORD_COMPRESS] = "audio-record-compress",
+ [USECASE_AUDIO_RECORD_LOW_LATENCY] = "low-latency-record",
+ [USECASE_AUDIO_RECORD_FM_VIRTUAL] = "fm-virtual-record",
+ [USECASE_AUDIO_PLAYBACK_FM] = "play-fm",
+ [USECASE_AUDIO_HFP_SCO] = "hfp-sco",
+ [USECASE_AUDIO_HFP_SCO_WB] = "hfp-sco-wb",
+ [USECASE_VOICE_CALL] = "voice-call",
+
+ [USECASE_VOICE2_CALL] = "voice2-call",
+ [USECASE_VOLTE_CALL] = "volte-call",
+ [USECASE_QCHAT_CALL] = "qchat-call",
+ [USECASE_VOWLAN_CALL] = "vowlan-call",
+ [USECASE_COMPRESS_VOIP_CALL] = "compress-voip-call",
+ [USECASE_INCALL_REC_UPLINK] = "incall-rec-uplink",
+ [USECASE_INCALL_REC_DOWNLINK] = "incall-rec-downlink",
+ [USECASE_INCALL_REC_UPLINK_AND_DOWNLINK] = "incall-rec-uplink-and-downlink",
+ [USECASE_INCALL_REC_UPLINK_COMPRESS] = "incall-rec-uplink-compress",
+ [USECASE_INCALL_REC_DOWNLINK_COMPRESS] = "incall-rec-downlink-compress",
+ [USECASE_INCALL_REC_UPLINK_AND_DOWNLINK_COMPRESS] = "incall-rec-uplink-and-downlink-compress",
+
+ [USECASE_INCALL_MUSIC_UPLINK] = "incall_music_uplink",
+ [USECASE_INCALL_MUSIC_UPLINK2] = "incall_music_uplink2",
+ [USECASE_AUDIO_SPKR_CALIB_RX] = "spkr-rx-calib",
+ [USECASE_AUDIO_SPKR_CALIB_TX] = "spkr-vi-record",
+
+ [USECASE_AUDIO_PLAYBACK_AFE_PROXY] = "afe-proxy-playback",
+ [USECASE_AUDIO_RECORD_AFE_PROXY] = "afe-proxy-record",
+};
+
+static const audio_usecase_t offload_usecases[] = {
+ USECASE_AUDIO_PLAYBACK_OFFLOAD,
+#ifdef MULTIPLE_OFFLOAD_ENABLED
+ USECASE_AUDIO_PLAYBACK_OFFLOAD2,
+ USECASE_AUDIO_PLAYBACK_OFFLOAD3,
+ USECASE_AUDIO_PLAYBACK_OFFLOAD4,
+ USECASE_AUDIO_PLAYBACK_OFFLOAD5,
+ USECASE_AUDIO_PLAYBACK_OFFLOAD6,
+ USECASE_AUDIO_PLAYBACK_OFFLOAD7,
+ USECASE_AUDIO_PLAYBACK_OFFLOAD8,
+ USECASE_AUDIO_PLAYBACK_OFFLOAD9,
+#endif
+};
+
+#define STRING_TO_ENUM(string) { #string, string }
+
+struct string_to_enum {
+ const char *name;
+ uint32_t value;
+};
+
+static const struct string_to_enum out_channels_name_to_enum_table[] = {
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_STEREO),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_5POINT1),
+ STRING_TO_ENUM(AUDIO_CHANNEL_OUT_7POINT1),
+};
+
+static const struct string_to_enum out_formats_name_to_enum_table[] = {
+ STRING_TO_ENUM(AUDIO_FORMAT_AC3),
+ STRING_TO_ENUM(AUDIO_FORMAT_E_AC3),
+ STRING_TO_ENUM(AUDIO_FORMAT_E_AC3_JOC),
+};
+
+static struct audio_device *adev = NULL;
+static pthread_mutex_t adev_init_lock;
+static unsigned int audio_device_ref_count;
+
+static int set_voice_volume_l(struct audio_device *adev, float volume);
+
+static int check_and_set_gapless_mode(struct audio_device *adev) {
+
+
+ char value[PROPERTY_VALUE_MAX] = {0};
+ bool gapless_enabled = false;
+ const char *mixer_ctl_name = "Compress Gapless Playback";
+ struct mixer_ctl *ctl;
+
+ ALOGV("%s:", __func__);
+ property_get("audio.offload.gapless.enabled", value, NULL);
+ gapless_enabled = atoi(value) || !strncmp("true", value, 4);
+
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+
+ if (mixer_ctl_set_value(ctl, 0, gapless_enabled) < 0) {
+ ALOGE("%s: Could not set gapless mode %d",
+ __func__, gapless_enabled);
+ return -EINVAL;
+ }
+ return 0;
+}
+
+static bool is_supported_format(audio_format_t format)
+{
+ if (format == AUDIO_FORMAT_MP3 ||
+ format == AUDIO_FORMAT_AAC_LC ||
+ format == AUDIO_FORMAT_AAC_HE_V1 ||
+ format == AUDIO_FORMAT_AAC_HE_V2 ||
+ format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD ||
+ format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD ||
+ format == AUDIO_FORMAT_FLAC ||
+ format == AUDIO_FORMAT_ALAC ||
+ format == AUDIO_FORMAT_APE ||
+ format == AUDIO_FORMAT_VORBIS ||
+ format == AUDIO_FORMAT_WMA ||
+ format == AUDIO_FORMAT_WMA_PRO)
+ return true;
+
+ return false;
+}
+
+static int get_snd_codec_id(audio_format_t format)
+{
+ int id = 0;
+
+ switch (format & AUDIO_FORMAT_MAIN_MASK) {
+ case AUDIO_FORMAT_MP3:
+ id = SND_AUDIOCODEC_MP3;
+ break;
+ case AUDIO_FORMAT_AAC:
+ id = SND_AUDIOCODEC_AAC;
+ break;
+ case AUDIO_FORMAT_PCM_OFFLOAD:
+ id = SND_AUDIOCODEC_PCM;
+ break;
+ case AUDIO_FORMAT_FLAC:
+ id = SND_AUDIOCODEC_FLAC;
+ break;
+ case AUDIO_FORMAT_ALAC:
+ id = SND_AUDIOCODEC_ALAC;
+ break;
+ case AUDIO_FORMAT_APE:
+ id = SND_AUDIOCODEC_APE;
+ break;
+ case AUDIO_FORMAT_VORBIS:
+ id = SND_AUDIOCODEC_VORBIS;
+ break;
+ case AUDIO_FORMAT_WMA:
+ id = SND_AUDIOCODEC_WMA;
+ break;
+ case AUDIO_FORMAT_WMA_PRO:
+ id = SND_AUDIOCODEC_WMA_PRO;
+ break;
+ default:
+ ALOGE("%s: Unsupported audio format :%x", __func__, format);
+ }
+
+ return id;
+}
+
+int get_snd_card_state(struct audio_device *adev)
+{
+ int snd_scard_state;
+
+ if (!adev)
+ return SND_CARD_STATE_OFFLINE;
+
+ pthread_mutex_lock(&adev->snd_card_status.lock);
+ snd_scard_state = adev->snd_card_status.state;
+ pthread_mutex_unlock(&adev->snd_card_status.lock);
+
+ return snd_scard_state;
+}
+
+static int set_snd_card_state(struct audio_device *adev, int snd_scard_state)
+{
+ if (!adev)
+ return -ENOSYS;
+
+ pthread_mutex_lock(&adev->snd_card_status.lock);
+ adev->snd_card_status.state = snd_scard_state;
+ pthread_mutex_unlock(&adev->snd_card_status.lock);
+
+ return 0;
+}
+
+static int enable_audio_route_for_voice_usecases(struct audio_device *adev,
+ struct audio_usecase *uc_info)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase;
+
+ if (uc_info == NULL)
+ return -EINVAL;
+
+ /* Re-route all voice usecases on the shared backend other than the
+ specified usecase to new snd devices */
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if ((usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) &&
+ (usecase != uc_info))
+ enable_audio_route(adev, usecase);
+ }
+ return 0;
+}
+
+int pcm_ioctl(struct pcm *pcm, int request, ...)
+{
+ va_list ap;
+ void * arg;
+ int pcm_fd = *(int*)pcm;
+
+ va_start(ap, request);
+ arg = va_arg(ap, void *);
+ va_end(ap);
+
+ return ioctl(pcm_fd, request, arg);
+}
+
+int enable_audio_route(struct audio_device *adev,
+ struct audio_usecase *usecase)
+{
+ snd_device_t snd_device;
+ char mixer_path[MIXER_PATH_MAX_LENGTH];
+
+ if (usecase == NULL)
+ return -EINVAL;
+
+ ALOGV("%s: enter: usecase(%d)", __func__, usecase->id);
+
+ if (usecase->type == PCM_CAPTURE)
+ snd_device = usecase->in_snd_device;
+ else
+ snd_device = usecase->out_snd_device;
+
+#ifdef DS1_DOLBY_DAP_ENABLED
+ audio_extn_dolby_set_dmid(adev);
+ audio_extn_dolby_set_endpoint(adev);
+#endif
+ audio_extn_dolby_ds2_set_endpoint(adev);
+ audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_BUSY);
+ audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_BUSY);
+ audio_extn_utils_send_audio_calibration(adev, usecase);
+ audio_extn_utils_send_app_type_cfg(usecase);
+ strcpy(mixer_path, use_case_table[usecase->id]);
+ platform_add_backend_name(mixer_path, snd_device);
+ ALOGD("%s: apply mixer and update path: %s", __func__, mixer_path);
+ audio_route_apply_and_update_path(adev->audio_route, mixer_path);
+ ALOGV("%s: exit", __func__);
+ return 0;
+}
+
+int disable_audio_route(struct audio_device *adev,
+ struct audio_usecase *usecase)
+{
+ snd_device_t snd_device;
+ char mixer_path[MIXER_PATH_MAX_LENGTH];
+
+ if (usecase == NULL || usecase->id == USECASE_INVALID)
+ return -EINVAL;
+
+ ALOGV("%s: enter: usecase(%d)", __func__, usecase->id);
+ if (usecase->type == PCM_CAPTURE)
+ snd_device = usecase->in_snd_device;
+ else
+ snd_device = usecase->out_snd_device;
+ strcpy(mixer_path, use_case_table[usecase->id]);
+ platform_add_backend_name(mixer_path, snd_device);
+ ALOGD("%s: reset and update mixer path: %s", __func__, mixer_path);
+ audio_route_reset_and_update_path(adev->audio_route, mixer_path);
+ audio_extn_sound_trigger_update_stream_status(usecase, ST_EVENT_STREAM_FREE);
+ audio_extn_listen_update_stream_status(usecase, LISTEN_EVENT_STREAM_FREE);
+ ALOGV("%s: exit", __func__);
+ return 0;
+}
+
+int enable_snd_device(struct audio_device *adev,
+ snd_device_t snd_device)
+{
+ char device_name[DEVICE_NAME_MAX_SIZE] = {0};
+
+ if (snd_device < SND_DEVICE_MIN ||
+ snd_device >= SND_DEVICE_MAX) {
+ ALOGE("%s: Invalid sound device %d", __func__, snd_device);
+ return -EINVAL;
+ }
+
+ adev->snd_dev_ref_cnt[snd_device]++;
+
+ if(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0 ) {
+ ALOGE("%s: Invalid sound device returned", __func__);
+ return -EINVAL;
+ }
+ if (adev->snd_dev_ref_cnt[snd_device] > 1) {
+ ALOGV("%s: snd_device(%d: %s) is already active",
+ __func__, snd_device, device_name);
+ return 0;
+ }
+
+ if (audio_extn_spkr_prot_is_enabled())
+ audio_extn_spkr_prot_calib_cancel(adev);
+ /* start usb playback thread */
+ if(SND_DEVICE_OUT_USB_HEADSET == snd_device ||
+ SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET == snd_device)
+ audio_extn_usb_start_playback(adev);
+
+ /* start usb capture thread */
+ if(SND_DEVICE_IN_USB_HEADSET_MIC == snd_device)
+ audio_extn_usb_start_capture(adev);
+
+ if (SND_DEVICE_OUT_BT_A2DP == snd_device ||
+ (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP) == snd_device)
+ audio_extn_a2dp_start_playback();
+
+ if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
+ audio_extn_spkr_prot_is_enabled()) {
+ if (audio_extn_spkr_prot_get_acdb_id(snd_device) < 0) {
+ adev->snd_dev_ref_cnt[snd_device]--;
+ return -EINVAL;
+ }
+ if (audio_extn_spkr_prot_start_processing(snd_device)) {
+ ALOGE("%s: spkr_start_processing failed", __func__);
+ return -EINVAL;
+ }
+ } else {
+ ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
+ /* due to the possibility of calibration overwrite between listen
+ and audio, notify listen hal before audio calibration is sent */
+ audio_extn_sound_trigger_update_device_status(snd_device,
+ ST_EVENT_SND_DEVICE_BUSY);
+ audio_extn_listen_update_device_status(snd_device,
+ LISTEN_EVENT_SND_DEVICE_BUSY);
+ if (platform_get_snd_device_acdb_id(snd_device) < 0) {
+ adev->snd_dev_ref_cnt[snd_device]--;
+ audio_extn_sound_trigger_update_device_status(snd_device,
+ ST_EVENT_SND_DEVICE_FREE);
+ audio_extn_listen_update_device_status(snd_device,
+ LISTEN_EVENT_SND_DEVICE_FREE);
+ return -EINVAL;
+ }
+ audio_extn_dev_arbi_acquire(snd_device);
+ audio_route_apply_and_update_path(adev->audio_route, device_name);
+ }
+ return 0;
+}
+
+int disable_snd_device(struct audio_device *adev,
+ snd_device_t snd_device)
+{
+ char device_name[DEVICE_NAME_MAX_SIZE] = {0};
+
+ if (snd_device < SND_DEVICE_MIN ||
+ snd_device >= SND_DEVICE_MAX) {
+ ALOGE("%s: Invalid sound device %d", __func__, snd_device);
+ return -EINVAL;
+ }
+ if (adev->snd_dev_ref_cnt[snd_device] <= 0) {
+ ALOGE("%s: device ref cnt is already 0", __func__);
+ return -EINVAL;
+ }
+
+ adev->snd_dev_ref_cnt[snd_device]--;
+
+ if(platform_get_snd_device_name_extn(adev->platform, snd_device, device_name) < 0) {
+ ALOGE("%s: Invalid sound device returned", __func__);
+ return -EINVAL;
+ }
+
+ if (adev->snd_dev_ref_cnt[snd_device] == 0) {
+ ALOGD("%s: snd_device(%d: %s)", __func__, snd_device, device_name);
+ /* exit usb play back thread */
+ if(SND_DEVICE_OUT_USB_HEADSET == snd_device ||
+ SND_DEVICE_OUT_SPEAKER_AND_USB_HEADSET == snd_device)
+ audio_extn_usb_stop_playback();
+
+ /* exit usb capture thread */
+ if(SND_DEVICE_IN_USB_HEADSET_MIC == snd_device)
+ audio_extn_usb_stop_capture();
+
+ if (SND_DEVICE_OUT_BT_A2DP == snd_device ||
+ (SND_DEVICE_OUT_SPEAKER_AND_BT_A2DP) == snd_device)
+ audio_extn_a2dp_stop_playback();
+
+ if ((snd_device == SND_DEVICE_OUT_SPEAKER ||
+ snd_device == SND_DEVICE_OUT_VOICE_SPEAKER) &&
+ audio_extn_spkr_prot_is_enabled()) {
+ audio_extn_spkr_prot_stop_processing(snd_device);
+ } else {
+ audio_route_reset_and_update_path(adev->audio_route, device_name);
+ audio_extn_dev_arbi_release(snd_device);
+ }
+
+ audio_extn_sound_trigger_update_device_status(snd_device,
+ ST_EVENT_SND_DEVICE_FREE);
+ audio_extn_listen_update_device_status(snd_device,
+ LISTEN_EVENT_SND_DEVICE_FREE);
+ }
+
+ return 0;
+}
+
+static void check_usecases_codec_backend(struct audio_device *adev,
+ struct audio_usecase *uc_info,
+ snd_device_t snd_device)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase;
+ bool switch_device[AUDIO_USECASE_MAX];
+ int i, num_uc_to_switch = 0;
+
+ /*
+ * This function is to make sure that all the usecases that are active on
+ * the hardware codec backend are always routed to any one device that is
+ * handled by the hardware codec.
+ * For example, if low-latency and deep-buffer usecases are currently active
+ * on speaker and out_set_parameters(headset) is received on low-latency
+ * output, then we have to make sure deep-buffer is also switched to headset,
+ * because of the limitation that both the devices cannot be enabled
+ * at the same time as they share the same backend.
+ */
+ /*
+ * This call is to check if we need to force routing for a particular stream
+ * If there is a backend configuration change for the device when a
+ * new stream starts, then ADM needs to be closed and re-opened with the new
+ * configuraion. This call check if we need to re-route all the streams
+ * associated with the backend. Touch tone + 24 bit playback.
+ */
+ bool force_routing = platform_check_and_set_codec_backend_cfg(adev, uc_info);
+
+ /* Disable all the usecases on the shared backend other than the
+ specified usecase */
+ for (i = 0; i < AUDIO_USECASE_MAX; i++)
+ switch_device[i] = false;
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type != PCM_CAPTURE &&
+ usecase != uc_info &&
+ (usecase->out_snd_device != snd_device || force_routing) &&
+ usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) {
+ ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
+ __func__, use_case_table[usecase->id],
+ platform_get_snd_device_name(usecase->out_snd_device));
+ disable_audio_route(adev, usecase);
+ switch_device[usecase->id] = true;
+ num_uc_to_switch++;
+ }
+ }
+
+ if (num_uc_to_switch) {
+ /* All streams have been de-routed. Disable the device */
+
+ /* Make sure the previous devices to be disabled first and then enable the
+ selected devices */
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (switch_device[usecase->id]) {
+ disable_snd_device(adev, usecase->out_snd_device);
+ }
+ }
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (switch_device[usecase->id]) {
+ enable_snd_device(adev, snd_device);
+ }
+ }
+
+ /* Re-route all the usecases on the shared backend other than the
+ specified usecase to new snd devices */
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ /* Update the out_snd_device only before enabling the audio route */
+ if (switch_device[usecase->id] ) {
+ usecase->out_snd_device = snd_device;
+ if (usecase->type != VOICE_CALL && usecase->type != VOIP_CALL)
+ enable_audio_route(adev, usecase);
+ }
+ }
+ }
+}
+
+static void check_and_route_capture_usecases(struct audio_device *adev,
+ struct audio_usecase *uc_info,
+ snd_device_t snd_device)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase;
+ bool switch_device[AUDIO_USECASE_MAX];
+ int i, num_uc_to_switch = 0;
+
+ /*
+ * This function is to make sure that all the active capture usecases
+ * are always routed to the same input sound device.
+ * For example, if audio-record and voice-call usecases are currently
+ * active on speaker(rx) and speaker-mic (tx) and out_set_parameters(earpiece)
+ * is received for voice call then we have to make sure that audio-record
+ * usecase is also switched to earpiece i.e. voice-dmic-ef,
+ * because of the limitation that two devices cannot be enabled
+ * at the same time if they share the same backend.
+ */
+ for (i = 0; i < AUDIO_USECASE_MAX; i++)
+ switch_device[i] = false;
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type != PCM_PLAYBACK &&
+ usecase != uc_info &&
+ usecase->in_snd_device != snd_device) {
+ ALOGV("%s: Usecase (%s) is active on (%s) - disabling ..",
+ __func__, use_case_table[usecase->id],
+ platform_get_snd_device_name(usecase->in_snd_device));
+ disable_audio_route(adev, usecase);
+ switch_device[usecase->id] = true;
+ num_uc_to_switch++;
+ }
+ }
+
+ if (num_uc_to_switch) {
+ /* All streams have been de-routed. Disable the device */
+
+ /* Make sure the previous devices to be disabled first and then enable the
+ selected devices */
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (switch_device[usecase->id]) {
+ disable_snd_device(adev, usecase->in_snd_device);
+ }
+ }
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (switch_device[usecase->id]) {
+ enable_snd_device(adev, snd_device);
+ }
+ }
+
+ /* Re-route all the usecases on the shared backend other than the
+ specified usecase to new snd devices */
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ /* Update the in_snd_device only before enabling the audio route */
+ if (switch_device[usecase->id] ) {
+ usecase->in_snd_device = snd_device;
+ if (usecase->type != VOICE_CALL && usecase->type != VOIP_CALL)
+ enable_audio_route(adev, usecase);
+ }
+ }
+ }
+}
+
+/* must be called with hw device mutex locked */
+static int read_hdmi_channel_masks(struct stream_out *out)
+{
+ int ret = 0;
+ int channels = platform_edid_get_max_channels(out->dev->platform);
+
+ switch (channels) {
+ /*
+ * Do not handle stereo output in Multi-channel cases
+ * Stereo case is handled in normal playback path
+ */
+ case 6:
+ ALOGV("%s: HDMI supports 5.1", __func__);
+ out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
+ break;
+ case 8:
+ ALOGV("%s: HDMI supports 5.1 and 7.1 channels", __func__);
+ out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_5POINT1;
+ out->supported_channel_masks[1] = AUDIO_CHANNEL_OUT_7POINT1;
+ break;
+ default:
+ ALOGE("HDMI does not support multi channel playback");
+ ret = -ENOSYS;
+ break;
+ }
+ return ret;
+}
+
+static audio_usecase_t get_voice_usecase_id_from_list(struct audio_device *adev)
+{
+ struct audio_usecase *usecase;
+ struct listnode *node;
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == VOICE_CALL) {
+ ALOGV("%s: usecase id %d", __func__, usecase->id);
+ return usecase->id;
+ }
+ }
+ return USECASE_INVALID;
+}
+
+struct audio_usecase *get_usecase_from_list(struct audio_device *adev,
+ audio_usecase_t uc_id)
+{
+ struct audio_usecase *usecase;
+ struct listnode *node;
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->id == uc_id)
+ return usecase;
+ }
+ return NULL;
+}
+
+int select_devices(struct audio_device *adev, audio_usecase_t uc_id)
+{
+ snd_device_t out_snd_device = SND_DEVICE_NONE;
+ snd_device_t in_snd_device = SND_DEVICE_NONE;
+ struct audio_usecase *usecase = NULL;
+ struct audio_usecase *vc_usecase = NULL;
+ struct audio_usecase *voip_usecase = NULL;
+ struct audio_usecase *hfp_usecase = NULL;
+ audio_usecase_t hfp_ucid;
+ struct listnode *node;
+ int status = 0;
+
+ usecase = get_usecase_from_list(adev, uc_id);
+ if (usecase == NULL) {
+ ALOGE("%s: Could not find the usecase(%d)", __func__, uc_id);
+ return -EINVAL;
+ }
+
+ if ((usecase->type == VOICE_CALL) ||
+ (usecase->type == VOIP_CALL) ||
+ (usecase->type == PCM_HFP_CALL)) {
+ out_snd_device = platform_get_output_snd_device(adev->platform,
+ usecase->stream.out->devices);
+ in_snd_device = platform_get_input_snd_device(adev->platform, usecase->stream.out->devices);
+ usecase->devices = usecase->stream.out->devices;
+ } else {
+ /*
+ * If the voice call is active, use the sound devices of voice call usecase
+ * so that it would not result any device switch. All the usecases will
+ * be switched to new device when select_devices() is called for voice call
+ * usecase. This is to avoid switching devices for voice call when
+ * check_usecases_codec_backend() is called below.
+ */
+ if (voice_is_in_call(adev) && adev->mode == AUDIO_MODE_IN_CALL) {
+ vc_usecase = get_usecase_from_list(adev,
+ get_voice_usecase_id_from_list(adev));
+ if ((vc_usecase) && ((vc_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) ||
+ (usecase->devices == AUDIO_DEVICE_IN_VOICE_CALL))) {
+ in_snd_device = vc_usecase->in_snd_device;
+ out_snd_device = vc_usecase->out_snd_device;
+ }
+ } else if (voice_extn_compress_voip_is_active(adev)) {
+ voip_usecase = get_usecase_from_list(adev, USECASE_COMPRESS_VOIP_CALL);
+ if ((voip_usecase) && ((voip_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+ (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND) &&
+ (voip_usecase->stream.out != adev->primary_output))) {
+ in_snd_device = voip_usecase->in_snd_device;
+ out_snd_device = voip_usecase->out_snd_device;
+ }
+ } else if (audio_extn_hfp_is_active(adev)) {
+ hfp_ucid = audio_extn_hfp_get_usecase();
+ hfp_usecase = get_usecase_from_list(adev, hfp_ucid);
+ if ((hfp_usecase) && (hfp_usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)) {
+ in_snd_device = hfp_usecase->in_snd_device;
+ out_snd_device = hfp_usecase->out_snd_device;
+ }
+ }
+ if (usecase->type == PCM_PLAYBACK) {
+ usecase->devices = usecase->stream.out->devices;
+ in_snd_device = SND_DEVICE_NONE;
+ if (out_snd_device == SND_DEVICE_NONE) {
+ out_snd_device = platform_get_output_snd_device(adev->platform,
+ usecase->stream.out->devices);
+ if (usecase->stream.out == adev->primary_output &&
+ adev->active_input &&
+ out_snd_device != usecase->out_snd_device) {
+ select_devices(adev, adev->active_input->usecase);
+ }
+ }
+ } else if (usecase->type == PCM_CAPTURE) {
+ usecase->devices = usecase->stream.in->device;
+ out_snd_device = SND_DEVICE_NONE;
+ if (in_snd_device == SND_DEVICE_NONE) {
+ audio_devices_t out_device = AUDIO_DEVICE_NONE;
+ if ((adev->active_input->source == AUDIO_SOURCE_VOICE_COMMUNICATION ||
+ (adev->mode == AUDIO_MODE_IN_COMMUNICATION &&
+ adev->active_input->source == AUDIO_SOURCE_MIC)) &&
+ adev->primary_output && !adev->primary_output->standby) {
+ out_device = adev->primary_output->devices;
+ platform_set_echo_reference(adev->platform, false);
+ } else if (usecase->id == USECASE_AUDIO_RECORD_AFE_PROXY) {
+ out_device = AUDIO_DEVICE_OUT_TELEPHONY_TX;
+ }
+ in_snd_device = platform_get_input_snd_device(adev->platform, out_device);
+ }
+ }
+ }
+
+ if (out_snd_device == usecase->out_snd_device &&
+ in_snd_device == usecase->in_snd_device) {
+ return 0;
+ }
+
+ ALOGD("%s: out_snd_device(%d: %s) in_snd_device(%d: %s)", __func__,
+ out_snd_device, platform_get_snd_device_name(out_snd_device),
+ in_snd_device, platform_get_snd_device_name(in_snd_device));
+
+ /*
+ * Limitation: While in call, to do a device switch we need to disable
+ * and enable both RX and TX devices though one of them is same as current
+ * device.
+ */
+ if ((usecase->type == VOICE_CALL) &&
+ (usecase->in_snd_device != SND_DEVICE_NONE) &&
+ (usecase->out_snd_device != SND_DEVICE_NONE)) {
+ status = platform_switch_voice_call_device_pre(adev->platform);
+ }
+
+ /* Disable current sound devices */
+ if (usecase->out_snd_device != SND_DEVICE_NONE) {
+ disable_audio_route(adev, usecase);
+ disable_snd_device(adev, usecase->out_snd_device);
+ }
+
+ if (usecase->in_snd_device != SND_DEVICE_NONE) {
+ disable_audio_route(adev, usecase);
+ disable_snd_device(adev, usecase->in_snd_device);
+ }
+
+ /* Applicable only on the targets that has external modem.
+ * New device information should be sent to modem before enabling
+ * the devices to reduce in-call device switch time.
+ */
+ if ((usecase->type == VOICE_CALL) &&
+ (usecase->in_snd_device != SND_DEVICE_NONE) &&
+ (usecase->out_snd_device != SND_DEVICE_NONE)) {
+ status = platform_switch_voice_call_enable_device_config(adev->platform,
+ out_snd_device,
+ in_snd_device);
+ }
+
+ /* Enable new sound devices */
+ if (out_snd_device != SND_DEVICE_NONE) {
+ if (usecase->devices & AUDIO_DEVICE_OUT_ALL_CODEC_BACKEND)
+ check_usecases_codec_backend(adev, usecase, out_snd_device);
+ enable_snd_device(adev, out_snd_device);
+ }
+
+ if (in_snd_device != SND_DEVICE_NONE) {
+ check_and_route_capture_usecases(adev, usecase, in_snd_device);
+ enable_snd_device(adev, in_snd_device);
+ }
+
+ if (usecase->type == VOICE_CALL || usecase->type == VOIP_CALL) {
+ status = platform_switch_voice_call_device_post(adev->platform,
+ out_snd_device,
+ in_snd_device);
+ enable_audio_route_for_voice_usecases(adev, usecase);
+ }
+
+ usecase->in_snd_device = in_snd_device;
+ usecase->out_snd_device = out_snd_device;
+
+ if (usecase->type == PCM_PLAYBACK) {
+ audio_extn_utils_update_stream_app_type_cfg(adev->platform,
+ &adev->streams_output_cfg_list,
+ usecase->stream.out->devices,
+ usecase->stream.out->flags,
+ usecase->stream.out->format,
+ usecase->stream.out->sample_rate,
+ usecase->stream.out->bit_width,
+ &usecase->stream.out->app_type_cfg);
+ ALOGI("%s Selected apptype: %d", __func__, usecase->stream.out->app_type_cfg.app_type);
+ }
+
+ enable_audio_route(adev, usecase);
+
+ /* Applicable only on the targets that has external modem.
+ * Enable device command should be sent to modem only after
+ * enabling voice call mixer controls
+ */
+ if (usecase->type == VOICE_CALL)
+ status = platform_switch_voice_call_usecase_route_post(adev->platform,
+ out_snd_device,
+ in_snd_device);
+ ALOGD("%s: done",__func__);
+
+ return status;
+}
+
+static int stop_input_stream(struct stream_in *in)
+{
+ int i, ret = 0;
+ struct audio_usecase *uc_info;
+ struct audio_device *adev = in->dev;
+
+ adev->active_input = NULL;
+
+ ALOGV("%s: enter: usecase(%d: %s)", __func__,
+ in->usecase, use_case_table[in->usecase]);
+ uc_info = get_usecase_from_list(adev, in->usecase);
+ if (uc_info == NULL) {
+ ALOGE("%s: Could not find the usecase (%d) in the list",
+ __func__, in->usecase);
+ return -EINVAL;
+ }
+
+ /* Close in-call recording streams */
+ voice_check_and_stop_incall_rec_usecase(adev, in);
+
+ /* 1. Disable stream specific mixer controls */
+ disable_audio_route(adev, uc_info);
+
+ /* 2. Disable the tx device */
+ disable_snd_device(adev, uc_info->in_snd_device);
+
+ list_remove(&uc_info->list);
+ free(uc_info);
+
+ ALOGV("%s: exit: status(%d)", __func__, ret);
+ return ret;
+}
+
+int start_input_stream(struct stream_in *in)
+{
+ /* 1. Enable output device and stream routing controls */
+ int ret = 0;
+ struct audio_usecase *uc_info;
+ struct audio_device *adev = in->dev;
+ int snd_card_status = get_snd_card_state(adev);
+
+ int usecase = platform_update_usecase_from_source(in->source,in->usecase);
+ if (get_usecase_from_list(adev, usecase) == NULL)
+ in->usecase = usecase;
+
+ ALOGD("%s: enter: stream(%p)usecase(%d: %s)",
+ __func__, &in->stream, in->usecase, use_case_table[in->usecase]);
+
+
+ if (SND_CARD_STATE_OFFLINE == snd_card_status) {
+ ALOGE("%s: sound card is not active/SSR returning error", __func__);
+ ret = -EIO;
+ goto error_config;
+ }
+
+ /* Check if source matches incall recording usecase criteria */
+ ret = voice_check_and_set_incall_rec_usecase(adev, in);
+ if (ret)
+ goto error_config;
+ else
+ ALOGV("%s: usecase(%d)", __func__, in->usecase);
+
+ if (get_usecase_from_list(adev, in->usecase) != NULL) {
+ ALOGE("%s: use case assigned already in use, stream(%p)usecase(%d: %s)",
+ __func__, &in->stream, in->usecase, use_case_table[in->usecase]);
+ goto error_config;
+ }
+
+ in->pcm_device_id = platform_get_pcm_device_id(in->usecase, PCM_CAPTURE);
+ if (in->pcm_device_id < 0) {
+ ALOGE("%s: Could not find PCM device id for the usecase(%d)",
+ __func__, in->usecase);
+ ret = -EINVAL;
+ goto error_config;
+ }
+
+ adev->active_input = in;
+ uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
+
+ if (!uc_info) {
+ ret = -ENOMEM;
+ goto error_config;
+ }
+
+ uc_info->id = in->usecase;
+ uc_info->type = PCM_CAPTURE;
+ uc_info->stream.in = in;
+ uc_info->devices = in->device;
+ uc_info->in_snd_device = SND_DEVICE_NONE;
+ uc_info->out_snd_device = SND_DEVICE_NONE;
+
+ list_add_tail(&adev->usecase_list, &uc_info->list);
+ audio_extn_perf_lock_acquire();
+ select_devices(adev, in->usecase);
+
+ ALOGV("%s: Opening PCM device card_id(%d) device_id(%d), channels %d",
+ __func__, adev->snd_card, in->pcm_device_id, in->config.channels);
+
+ unsigned int flags = PCM_IN;
+ unsigned int pcm_open_retry_count = 0;
+
+ if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY) {
+ flags |= PCM_MMAP | PCM_NOIRQ;
+ pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+ }
+
+ while (1) {
+ in->pcm = pcm_open(adev->snd_card, in->pcm_device_id,
+ flags, &in->config);
+ if (in->pcm == NULL || !pcm_is_ready(in->pcm)) {
+ ALOGE("%s: %s", __func__, pcm_get_error(in->pcm));
+ if (in->pcm != NULL) {
+ pcm_close(in->pcm);
+ in->pcm = NULL;
+ }
+ if (pcm_open_retry_count-- == 0) {
+ ret = -EIO;
+ goto error_open;
+ }
+ usleep(PROXY_OPEN_WAIT_TIME * 1000);
+ continue;
+ }
+ break;
+ }
+
+ ALOGV("%s: pcm_prepare", __func__);
+ ret = pcm_prepare(in->pcm);
+ if (ret < 0) {
+ ALOGE("%s: pcm_prepare returned %d", __func__, ret);
+ pcm_close(in->pcm);
+ in->pcm = NULL;
+ goto error_open;
+ }
+
+ audio_extn_perf_lock_release();
+
+ ALOGD("%s: exit", __func__);
+
+ return ret;
+
+error_open:
+ stop_input_stream(in);
+ audio_extn_perf_lock_release();
+
+error_config:
+ adev->active_input = NULL;
+ ALOGD("%s: exit: status(%d)", __func__, ret);
+
+ return ret;
+}
+
+/* must be called with out->lock locked */
+static int send_offload_cmd_l(struct stream_out* out, int command)
+{
+ struct offload_cmd *cmd = (struct offload_cmd *)calloc(1, sizeof(struct offload_cmd));
+
+ if (!cmd) {
+ ALOGE("failed to allocate mem for command 0x%x", command);
+ return -ENOMEM;
+ }
+
+ ALOGVV("%s %d", __func__, command);
+
+ cmd->cmd = command;
+ list_add_tail(&out->offload_cmd_list, &cmd->node);
+ pthread_cond_signal(&out->offload_cond);
+ return 0;
+}
+
+/* must be called iwth out->lock locked */
+static void stop_compressed_output_l(struct stream_out *out)
+{
+ out->offload_state = OFFLOAD_STATE_IDLE;
+ out->playback_started = 0;
+ out->send_new_metadata = 1;
+ if (out->compr != NULL) {
+ compress_stop(out->compr);
+ while (out->offload_thread_blocked) {
+ pthread_cond_wait(&out->cond, &out->lock);
+ }
+ }
+}
+
+bool is_offload_usecase(audio_usecase_t uc_id)
+{
+ unsigned int i;
+ for (i = 0; i < sizeof(offload_usecases)/sizeof(offload_usecases[0]); i++) {
+ if (uc_id == offload_usecases[i])
+ return true;
+ }
+ return false;
+}
+
+static audio_usecase_t get_offload_usecase(struct audio_device *adev)
+{
+ audio_usecase_t ret = USECASE_AUDIO_PLAYBACK_OFFLOAD;
+ unsigned int i, num_usecase = sizeof(offload_usecases)/sizeof(offload_usecases[0]);
+ char value[PROPERTY_VALUE_MAX] = {0};
+
+ property_get("audio.offload.multiple.enabled", value, NULL);
+ if (!(atoi(value) || !strncmp("true", value, 4)))
+ num_usecase = 1; /* If prop is not set, limit the num of offload usecases to 1 */
+
+ ALOGV("%s: num_usecase: %d", __func__, num_usecase);
+ for (i = 0; i < num_usecase; i++) {
+ if (!(adev->offload_usecases_state & (0x1<<i))) {
+ adev->offload_usecases_state |= 0x1 << i;
+ ret = offload_usecases[i];
+ break;
+ }
+ }
+ ALOGV("%s: offload usecase is %d", __func__, ret);
+ return ret;
+}
+
+static void free_offload_usecase(struct audio_device *adev,
+ audio_usecase_t uc_id)
+{
+ unsigned int i;
+ for (i = 0; i < sizeof(offload_usecases)/sizeof(offload_usecases[0]); i++) {
+ if (offload_usecases[i] == uc_id) {
+ adev->offload_usecases_state &= ~(0x1<<i);
+ break;
+ }
+ }
+ ALOGV("%s: free offload usecase %d", __func__, uc_id);
+}
+
+static void *offload_thread_loop(void *context)
+{
+ struct stream_out *out = (struct stream_out *) context;
+ struct listnode *item;
+ int ret = 0;
+
+ setpriority(PRIO_PROCESS, 0, ANDROID_PRIORITY_AUDIO);
+ set_sched_policy(0, SP_FOREGROUND);
+ prctl(PR_SET_NAME, (unsigned long)"Offload Callback", 0, 0, 0);
+
+ ALOGV("%s", __func__);
+ pthread_mutex_lock(&out->lock);
+ for (;;) {
+ struct offload_cmd *cmd = NULL;
+ stream_callback_event_t event;
+ bool send_callback = false;
+
+ ALOGVV("%s offload_cmd_list %d out->offload_state %d",
+ __func__, list_empty(&out->offload_cmd_list),
+ out->offload_state);
+ if (list_empty(&out->offload_cmd_list)) {
+ ALOGV("%s SLEEPING", __func__);
+ pthread_cond_wait(&out->offload_cond, &out->lock);
+ ALOGV("%s RUNNING", __func__);
+ continue;
+ }
+
+ item = list_head(&out->offload_cmd_list);
+ cmd = node_to_item(item, struct offload_cmd, node);
+ list_remove(item);
+
+ ALOGVV("%s STATE %d CMD %d out->compr %p",
+ __func__, out->offload_state, cmd->cmd, out->compr);
+
+ if (cmd->cmd == OFFLOAD_CMD_EXIT) {
+ free(cmd);
+ break;
+ }
+
+ if (out->compr == NULL) {
+ ALOGE("%s: Compress handle is NULL", __func__);
+ pthread_cond_signal(&out->cond);
+ continue;
+ }
+ out->offload_thread_blocked = true;
+ pthread_mutex_unlock(&out->lock);
+ send_callback = false;
+ switch(cmd->cmd) {
+ case OFFLOAD_CMD_WAIT_FOR_BUFFER:
+ ALOGD("copl(%p):calling compress_wait", out);
+ compress_wait(out->compr, -1);
+ ALOGD("copl(%p):out of compress_wait", out);
+ send_callback = true;
+ event = STREAM_CBK_EVENT_WRITE_READY;
+ break;
+ case OFFLOAD_CMD_PARTIAL_DRAIN:
+ ret = compress_next_track(out->compr);
+ if(ret == 0) {
+ ALOGD("copl(%p):calling compress_partial_drain", out);
+ ret = compress_partial_drain(out->compr);
+ ALOGD("copl(%p):out of compress_partial_drain", out);
+ if (ret < 0)
+ ret = -errno;
+ }
+ else if (ret == -ETIMEDOUT)
+ compress_drain(out->compr);
+ else
+ ALOGE("%s: Next track returned error %d",__func__, ret);
+
+ if (ret != -ENETRESET) {
+ send_callback = true;
+ event = STREAM_CBK_EVENT_DRAIN_READY;
+ ALOGV("copl(%p):send drain callback, ret %d", out, ret);
+ } else
+ ALOGE("%s: Block drain ready event during SSR", __func__);
+ break;
+ case OFFLOAD_CMD_DRAIN:
+ ALOGD("copl(%p):calling compress_drain", out);
+ compress_drain(out->compr);
+ ALOGD("copl(%p):calling compress_drain", out);
+ send_callback = true;
+ event = STREAM_CBK_EVENT_DRAIN_READY;
+ break;
+ default:
+ ALOGE("%s unknown command received: %d", __func__, cmd->cmd);
+ break;
+ }
+ pthread_mutex_lock(&out->lock);
+ out->offload_thread_blocked = false;
+ pthread_cond_signal(&out->cond);
+ if (send_callback) {
+ out->offload_callback(event, NULL, out->offload_cookie);
+ }
+ free(cmd);
+ }
+
+ pthread_cond_signal(&out->cond);
+ while (!list_empty(&out->offload_cmd_list)) {
+ item = list_head(&out->offload_cmd_list);
+ list_remove(item);
+ free(node_to_item(item, struct offload_cmd, node));
+ }
+ pthread_mutex_unlock(&out->lock);
+
+ return NULL;
+}
+
+static int create_offload_callback_thread(struct stream_out *out)
+{
+ pthread_cond_init(&out->offload_cond, (const pthread_condattr_t *) NULL);
+ list_init(&out->offload_cmd_list);
+ pthread_create(&out->offload_thread, (const pthread_attr_t *) NULL,
+ offload_thread_loop, out);
+ return 0;
+}
+
+static int destroy_offload_callback_thread(struct stream_out *out)
+{
+ pthread_mutex_lock(&out->lock);
+ stop_compressed_output_l(out);
+ send_offload_cmd_l(out, OFFLOAD_CMD_EXIT);
+
+ pthread_mutex_unlock(&out->lock);
+ pthread_join(out->offload_thread, (void **) NULL);
+ pthread_cond_destroy(&out->offload_cond);
+
+ return 0;
+}
+
+static bool allow_hdmi_channel_config(struct audio_device *adev)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase;
+ bool ret = true;
+
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ /*
+ * If voice call is already existing, do not proceed further to avoid
+ * disabling/enabling both RX and TX devices, CSD calls, etc.
+ * Once the voice call done, the HDMI channels can be configured to
+ * max channels of remaining use cases.
+ */
+ if (usecase->id == USECASE_VOICE_CALL) {
+ ALOGD("%s: voice call is active, no change in HDMI channels",
+ __func__);
+ ret = false;
+ break;
+ } else if (usecase->id == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
+ ALOGD("%s: multi channel playback is active, "
+ "no change in HDMI channels", __func__);
+ ret = false;
+ break;
+ } else if (is_offload_usecase(usecase->id) &&
+ audio_channel_count_from_out_mask(usecase->stream.out->channel_mask) > 2) {
+ ALOGD("%s: multi-channel(%x) compress offload playback is active, "
+ "no change in HDMI channels", __func__, usecase->stream.out->channel_mask);
+ ret = false;
+ break;
+ }
+ }
+ }
+ return ret;
+}
+
+static int check_and_set_hdmi_channels(struct audio_device *adev,
+ unsigned int channels)
+{
+ struct listnode *node;
+ struct audio_usecase *usecase;
+
+ /* Check if change in HDMI channel config is allowed */
+ if (!allow_hdmi_channel_config(adev))
+ return 0;
+
+ if (channels == adev->cur_hdmi_channels) {
+ ALOGD("%s: Requested channels are same as current channels(%d)", __func__, channels);
+ return 0;
+ }
+
+ platform_set_hdmi_channels(adev->platform, channels);
+ adev->cur_hdmi_channels = channels;
+
+ /*
+ * Deroute all the playback streams routed to HDMI so that
+ * the back end is deactivated. Note that backend will not
+ * be deactivated if any one stream is connected to it.
+ */
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == PCM_PLAYBACK &&
+ usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ disable_audio_route(adev, usecase);
+ }
+ }
+
+ /*
+ * Enable all the streams disabled above. Now the HDMI backend
+ * will be activated with new channel configuration
+ */
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == PCM_PLAYBACK &&
+ usecase->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ enable_audio_route(adev, usecase);
+ }
+ }
+
+ return 0;
+}
+
+static int stop_output_stream(struct stream_out *out)
+{
+ int i, ret = 0;
+ struct audio_usecase *uc_info;
+ struct audio_device *adev = out->dev;
+
+ ALOGV("%s: enter: usecase(%d: %s)", __func__,
+ out->usecase, use_case_table[out->usecase]);
+ uc_info = get_usecase_from_list(adev, out->usecase);
+ if (uc_info == NULL) {
+ ALOGE("%s: Could not find the usecase (%d) in the list",
+ __func__, out->usecase);
+ return -EINVAL;
+ }
+
+ if (is_offload_usecase(out->usecase)) {
+ if (adev->visualizer_stop_output != NULL)
+ adev->visualizer_stop_output(out->handle, out->pcm_device_id);
+ if (adev->offload_effects_stop_output != NULL)
+ adev->offload_effects_stop_output(out->handle, out->pcm_device_id);
+ }
+
+ /* 1. Get and set stream specific mixer controls */
+ disable_audio_route(adev, uc_info);
+
+ /* 2. Disable the rx device */
+ disable_snd_device(adev, uc_info->out_snd_device);
+
+ list_remove(&uc_info->list);
+ free(uc_info);
+
+ /* Must be called after removing the usecase from list */
+ if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
+ check_and_set_hdmi_channels(adev, DEFAULT_HDMI_OUT_CHANNELS);
+
+ ALOGV("%s: exit: status(%d)", __func__, ret);
+ return ret;
+}
+
+int start_output_stream(struct stream_out *out)
+{
+ int ret = 0;
+ int sink_channels = 0;
+ char prop_value[PROPERTY_VALUE_MAX] = {0};
+ struct audio_usecase *uc_info;
+ struct audio_device *adev = out->dev;
+ int snd_card_status = get_snd_card_state(adev);
+
+ if ((out->usecase < 0) || (out->usecase >= AUDIO_USECASE_MAX)) {
+ ret = -EINVAL;
+ goto error_config;
+ }
+
+ ALOGD("%s: enter: stream(%p)usecase(%d: %s) devices(%#x)",
+ __func__, &out->stream, out->usecase, use_case_table[out->usecase],
+ out->devices);
+
+ if (SND_CARD_STATE_OFFLINE == snd_card_status) {
+ ALOGE("%s: sound card is not active/SSR returning error", __func__);
+ ret = -EIO;
+ goto error_config;
+ }
+
+ out->pcm_device_id = platform_get_pcm_device_id(out->usecase, PCM_PLAYBACK);
+ if (out->pcm_device_id < 0) {
+ ALOGE("%s: Invalid PCM device id(%d) for the usecase(%d)",
+ __func__, out->pcm_device_id, out->usecase);
+ ret = -EINVAL;
+ goto error_config;
+ }
+
+ uc_info = (struct audio_usecase *)calloc(1, sizeof(struct audio_usecase));
+
+ if (!uc_info) {
+ ret = -ENOMEM;
+ goto error_config;
+ }
+
+ uc_info->id = out->usecase;
+ uc_info->type = PCM_PLAYBACK;
+ uc_info->stream.out = out;
+ uc_info->devices = out->devices;
+ uc_info->in_snd_device = SND_DEVICE_NONE;
+ uc_info->out_snd_device = SND_DEVICE_NONE;
+
+ /* This must be called before adding this usecase to the list */
+ if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) {
+ property_get("audio.use.hdmi.sink.cap", prop_value, NULL);
+ if (!strncmp("true", prop_value, 4)) {
+ sink_channels = platform_edid_get_max_channels(out->dev->platform);
+ ALOGD("%s: set HDMI channel count[%d] based on sink capability", __func__, sink_channels);
+ check_and_set_hdmi_channels(adev, sink_channels);
+ } else {
+ if (is_offload_usecase(out->usecase))
+ check_and_set_hdmi_channels(adev, out->compr_config.codec->ch_in);
+ else
+ check_and_set_hdmi_channels(adev, out->config.channels);
+ }
+ }
+
+ list_add_tail(&adev->usecase_list, &uc_info->list);
+
+ select_devices(adev, out->usecase);
+
+ ALOGV("%s: Opening PCM device card_id(%d) device_id(%d) format(%#x)",
+ __func__, adev->snd_card, out->pcm_device_id, out->config.format);
+ if (!is_offload_usecase(out->usecase)) {
+ unsigned int flags = PCM_OUT;
+ unsigned int pcm_open_retry_count = 0;
+ if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY) {
+ flags |= PCM_MMAP | PCM_NOIRQ;
+ pcm_open_retry_count = PROXY_OPEN_RETRY_COUNT;
+ } else
+ flags |= PCM_MONOTONIC;
+
+ while (1) {
+ out->pcm = pcm_open(adev->snd_card, out->pcm_device_id,
+ flags, &out->config);
+ if (out->pcm == NULL || !pcm_is_ready(out->pcm)) {
+ ALOGE("%s: %s", __func__, pcm_get_error(out->pcm));
+ if (out->pcm != NULL) {
+ pcm_close(out->pcm);
+ out->pcm = NULL;
+ }
+ if (pcm_open_retry_count-- == 0) {
+ ret = -EIO;
+ goto error_open;
+ }
+ usleep(PROXY_OPEN_WAIT_TIME * 1000);
+ continue;
+ }
+ break;
+ }
+
+ ALOGV("%s: pcm_prepare", __func__);
+ if (pcm_is_ready(out->pcm)) {
+ ret = pcm_prepare(out->pcm);
+ if (ret < 0) {
+ ALOGE("%s: pcm_prepare returned %d", __func__, ret);
+ pcm_close(out->pcm);
+ out->pcm = NULL;
+ goto error_open;
+ }
+ }
+ } else {
+ out->pcm = NULL;
+ out->compr = compress_open(adev->snd_card,
+ out->pcm_device_id,
+ COMPRESS_IN, &out->compr_config);
+ if (out->compr && !is_compress_ready(out->compr)) {
+ ALOGE("%s: %s", __func__, compress_get_error(out->compr));
+ compress_close(out->compr);
+ out->compr = NULL;
+ ret = -EIO;
+ goto error_open;
+ }
+ if (out->offload_callback)
+ compress_nonblock(out->compr, out->non_blocking);
+
+#ifdef DS1_DOLBY_DDP_ENABLED
+ if (audio_extn_is_dolby_format(out->format))
+ audio_extn_dolby_send_ddp_endp_params(adev);
+#endif
+
+ if (adev->visualizer_start_output != NULL)
+ adev->visualizer_start_output(out->handle, out->pcm_device_id);
+ if (adev->offload_effects_start_output != NULL)
+ adev->offload_effects_start_output(out->handle, out->pcm_device_id);
+ }
+
+ ALOGD("%s: exit", __func__);
+
+ return 0;
+error_open:
+ stop_output_stream(out);
+error_config:
+ return ret;
+}
+
+static int check_input_parameters(uint32_t sample_rate,
+ audio_format_t format,
+ int channel_count)
+{
+ int ret = 0;
+
+ if ((format != AUDIO_FORMAT_PCM_16_BIT) &&
+ !voice_extn_compress_voip_is_format_supported(format) &&
+ !audio_extn_compr_cap_format_supported(format)) ret = -EINVAL;
+
+ switch (channel_count) {
+ case 1:
+ case 2:
+ case 6:
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ switch (sample_rate) {
+ case 8000:
+ case 11025:
+ case 12000:
+ case 16000:
+ case 22050:
+ case 24000:
+ case 32000:
+ case 44100:
+ case 48000:
+ break;
+ default:
+ ret = -EINVAL;
+ }
+
+ return ret;
+}
+
+static size_t get_input_buffer_size(uint32_t sample_rate,
+ audio_format_t format,
+ int channel_count,
+ bool is_low_latency)
+{
+ size_t size = 0;
+
+ if (check_input_parameters(sample_rate, format, channel_count) != 0)
+ return 0;
+
+ size = (sample_rate * AUDIO_CAPTURE_PERIOD_DURATION_MSEC) / 1000;
+ if (is_low_latency)
+ size = configured_low_latency_capture_period_size;
+ /* ToDo: should use frame_size computed based on the format and
+ channel_count here. */
+ size *= sizeof(short) * channel_count;
+
+ /* make sure the size is multiple of 32 bytes
+ * At 48 kHz mono 16-bit PCM:
+ * 5.000 ms = 240 frames = 15*16*1*2 = 480, a whole multiple of 32 (15)
+ * 3.333 ms = 160 frames = 10*16*1*2 = 320, a whole multiple of 32 (10)
+ */
+ size += 0x1f;
+ size &= ~0x1f;
+
+ return size;
+}
+
+static uint32_t out_get_sample_rate(const struct audio_stream *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+
+ return out->sample_rate;
+}
+
+static int out_set_sample_rate(struct audio_stream *stream __unused,
+ uint32_t rate __unused)
+{
+ return -ENOSYS;
+}
+
+static size_t out_get_buffer_size(const struct audio_stream *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+
+ if (is_offload_usecase(out->usecase))
+ return out->compr_config.fragment_size;
+ else if(out->usecase == USECASE_COMPRESS_VOIP_CALL)
+ return voice_extn_compress_voip_out_get_buffer_size(out);
+
+ return out->config.period_size *
+ audio_stream_out_frame_size((const struct audio_stream_out *)stream);
+}
+
+static uint32_t out_get_channels(const struct audio_stream *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+
+ return out->channel_mask;
+}
+
+static audio_format_t out_get_format(const struct audio_stream *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+
+ return out->format;
+}
+
+static int out_set_format(struct audio_stream *stream __unused,
+ audio_format_t format __unused)
+{
+ return -ENOSYS;
+}
+
+static int out_standby(struct audio_stream *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct audio_device *adev = out->dev;
+
+ ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
+ stream, out->usecase, use_case_table[out->usecase]);
+ if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
+ /* Ignore standby in case of voip call because the voip output
+ * stream is closed in adev_close_output_stream()
+ */
+ ALOGD("%s: Ignore Standby in VOIP call", __func__);
+ return 0;
+ }
+
+ pthread_mutex_lock(&out->lock);
+ if (!out->standby) {
+ pthread_mutex_lock(&adev->lock);
+ out->standby = true;
+ if (!is_offload_usecase(out->usecase)) {
+ if (out->pcm) {
+ pcm_close(out->pcm);
+ out->pcm = NULL;
+ }
+ } else {
+ ALOGD("copl(%p):standby", out);
+ stop_compressed_output_l(out);
+ out->gapless_mdata.encoder_delay = 0;
+ out->gapless_mdata.encoder_padding = 0;
+ if (out->compr != NULL) {
+ compress_close(out->compr);
+ out->compr = NULL;
+ }
+ }
+ stop_output_stream(out);
+ pthread_mutex_unlock(&adev->lock);
+ }
+ pthread_mutex_unlock(&out->lock);
+ ALOGV("%s: exit", __func__);
+ return 0;
+}
+
+static int out_dump(const struct audio_stream *stream __unused,
+ int fd __unused)
+{
+ return 0;
+}
+
+static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms)
+{
+ int ret = 0;
+ char value[32];
+ bool is_meta_data_params = false;
+
+ if (!out || !parms) {
+ ALOGE("%s: return invalid ",__func__);
+ return -EINVAL;
+ }
+
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_FORMAT, value, sizeof(value));
+ if (ret >= 0) {
+ if (atoi(value) == SND_AUDIOSTREAMFORMAT_MP4ADTS) {
+ out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_MP4ADTS;
+ ALOGV("ADTS format is set in offload mode");
+ }
+ out->send_new_metadata = 1;
+ }
+
+ ret = audio_extn_parse_compress_metadata(out, parms);
+
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_SAMPLE_RATE, value, sizeof(value));
+ if(ret >= 0)
+ is_meta_data_params = true;
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_NUM_CHANNEL, value, sizeof(value));
+ if(ret >= 0)
+ is_meta_data_params = true;
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_AVG_BIT_RATE, value, sizeof(value));
+ if(ret >= 0)
+ is_meta_data_params = true;
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value));
+ if (ret >= 0) {
+ is_meta_data_params = true;
+ out->gapless_mdata.encoder_delay = atoi(value); //whats a good limit check?
+ }
+ ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value));
+ if (ret >= 0) {
+ is_meta_data_params = true;
+ out->gapless_mdata.encoder_padding = atoi(value);
+ }
+
+ if(!is_meta_data_params) {
+ ALOGV("%s: Not gapless meta data params", __func__);
+ return 0;
+ }
+ out->send_new_metadata = 1;
+ ALOGV("%s new encoder delay %u and padding %u", __func__,
+ out->gapless_mdata.encoder_delay, out->gapless_mdata.encoder_padding);
+
+ return 0;
+}
+
+static bool output_drives_call(struct audio_device *adev, struct stream_out *out)
+{
+ return out == adev->primary_output || out == adev->voice_tx_output;
+}
+
+static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct audio_device *adev = out->dev;
+ struct audio_usecase *usecase;
+ struct listnode *node;
+ struct str_parms *parms;
+ char value[32];
+ int ret = 0, val = 0, err;
+ bool select_new_device = false;
+
+ ALOGD("%s: enter: usecase(%d: %s) kvpairs: %s",
+ __func__, out->usecase, use_case_table[out->usecase], kvpairs);
+ parms = str_parms_create_str(kvpairs);
+ if (!parms)
+ goto error;
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+ if (err >= 0) {
+ val = atoi(value);
+ pthread_mutex_lock(&out->lock);
+ pthread_mutex_lock(&adev->lock);
+
+ /*
+ * When HDMI cable is unplugged/usb hs is disconnected the
+ * music playback is paused and the policy manager sends routing=0
+ * But the audioflingercontinues to write data until standby time
+ * (3sec). As the HDMI core is turned off, the write gets blocked.
+ * Avoid this by routing audio to speaker until standby.
+ */
+ if ((out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL ||
+ out->devices == AUDIO_DEVICE_OUT_ANLG_DOCK_HEADSET) &&
+ val == AUDIO_DEVICE_NONE) {
+ val = AUDIO_DEVICE_OUT_SPEAKER;
+ }
+
+ /*
+ * select_devices() call below switches all the usecases on the same
+ * backend to the new device. Refer to check_usecases_codec_backend() in
+ * the select_devices(). But how do we undo this?
+ *
+ * For example, music playback is active on headset (deep-buffer usecase)
+ * and if we go to ringtones and select a ringtone, low-latency usecase
+ * will be started on headset+speaker. As we can't enable headset+speaker
+ * and headset devices at the same time, select_devices() switches the music
+ * playback to headset+speaker while starting low-lateny usecase for ringtone.
+ * So when the ringtone playback is completed, how do we undo the same?
+ *
+ * We are relying on the out_set_parameters() call on deep-buffer output,
+ * once the ringtone playback is ended.
+ * NOTE: We should not check if the current devices are same as new devices.
+ * Because select_devices() must be called to switch back the music
+ * playback to headset.
+ */
+ if (val != 0) {
+ out->devices = val;
+
+ if (!out->standby)
+ select_devices(adev, out->usecase);
+
+ if ((adev->mode == AUDIO_MODE_IN_CALL) &&
+ output_drives_call(adev, out)) {
+ adev->current_call_output = out;
+ if (!voice_is_in_call(adev))
+ ret = voice_start_call(adev);
+ else
+ voice_update_devices_for_all_voice_usecases(adev);
+ }
+ }
+
+ pthread_mutex_unlock(&adev->lock);
+ pthread_mutex_unlock(&out->lock);
+ }
+
+ if (out == adev->primary_output) {
+ pthread_mutex_lock(&adev->lock);
+ audio_extn_set_parameters(adev, parms);
+ pthread_mutex_unlock(&adev->lock);
+ }
+ if (is_offload_usecase(out->usecase)) {
+ pthread_mutex_lock(&out->lock);
+ parse_compress_metadata(out, parms);
+ pthread_mutex_unlock(&out->lock);
+ }
+
+ str_parms_destroy(parms);
+error:
+ ALOGV("%s: exit: code(%d)", __func__, ret);
+ return ret;
+}
+
+static char* out_get_parameters(const struct audio_stream *stream, const char *keys)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct str_parms *query = str_parms_create_str(keys);
+ char *str;
+ char value[256];
+ struct str_parms *reply = str_parms_create();
+ size_t i, j;
+ int ret;
+ bool first = true;
+
+ if (!query || !reply) {
+ ALOGE("out_get_parameters: failed to allocate mem for query or reply");
+ return NULL;
+ }
+
+ ALOGV("%s: enter: keys - %s", __func__, keys);
+ ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value));
+ if (ret >= 0) {
+ value[0] = '\0';
+ i = 0;
+ while (out->supported_channel_masks[i] != 0) {
+ for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) {
+ if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) {
+ if (!first) {
+ strcat(value, "|");
+ }
+ strcat(value, out_channels_name_to_enum_table[j].name);
+ first = false;
+ break;
+ }
+ }
+ i++;
+ }
+ str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value);
+ str = str_parms_to_str(reply);
+ } else {
+ voice_extn_out_get_parameters(out, query, reply);
+ str = str_parms_to_str(reply);
+ if (!strncmp(str, "", sizeof(""))) {
+ free(str);
+ str = strdup(keys);
+ }
+ }
+ str_parms_destroy(query);
+ str_parms_destroy(reply);
+ ALOGV("%s: exit: returns - %s", __func__, str);
+ return str;
+}
+
+static uint32_t out_get_latency(const struct audio_stream_out *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+
+ if (is_offload_usecase(out->usecase))
+ return COMPRESS_OFFLOAD_PLAYBACK_LATENCY;
+
+ return (out->config.period_count * out->config.period_size * 1000) /
+ (out->config.rate);
+}
+
+static int out_set_volume(struct audio_stream_out *stream, float left,
+ float right)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int volume[2];
+
+ if (out->usecase == USECASE_AUDIO_PLAYBACK_MULTI_CH) {
+ /* only take left channel into account: the API is for stereo anyway */
+ out->muted = (left == 0.0f);
+ return 0;
+ } else if (is_offload_usecase(out->usecase)) {
+ char mixer_ctl_name[128];
+ struct audio_device *adev = out->dev;
+ struct mixer_ctl *ctl;
+ int pcm_device_id = platform_get_pcm_device_id(out->usecase,
+ PCM_PLAYBACK);
+
+ snprintf(mixer_ctl_name, sizeof(mixer_ctl_name),
+ "Compress Playback %d Volume", pcm_device_id);
+ ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name);
+ if (!ctl) {
+ ALOGE("%s: Could not get ctl for mixer cmd - %s",
+ __func__, mixer_ctl_name);
+ return -EINVAL;
+ }
+ volume[0] = (int)(left * COMPRESS_PLAYBACK_VOLUME_MAX);
+ volume[1] = (int)(right * COMPRESS_PLAYBACK_VOLUME_MAX);
+ mixer_ctl_set_array(ctl, volume, sizeof(volume)/sizeof(volume[0]));
+ return 0;
+ }
+
+ return -ENOSYS;
+}
+
+static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
+ size_t bytes)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct audio_device *adev = out->dev;
+ int snd_scard_state = get_snd_card_state(adev);
+ ssize_t ret = 0;
+
+ pthread_mutex_lock(&out->lock);
+
+ if (SND_CARD_STATE_OFFLINE == snd_scard_state) {
+ // increase written size during SSR to avoid mismatch
+ // with the written frames count in AF
+ if (!is_offload_usecase(out->usecase))
+ out->written += bytes / (out->config.channels * sizeof(short));
+
+ if (out->pcm) {
+ ALOGD(" %s: sound card is not active/SSR state", __func__);
+ ret= -EIO;
+ goto exit;
+ } else if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) {
+ //during SSR for compress usecase we should return error to flinger
+ ALOGD(" copl %s: sound card is not active/SSR state", __func__);
+ pthread_mutex_unlock(&out->lock);
+ return -ENETRESET;
+ }
+ }
+
+ if (out->standby) {
+ out->standby = false;
+ pthread_mutex_lock(&adev->lock);
+ if (out->usecase == USECASE_COMPRESS_VOIP_CALL)
+ ret = voice_extn_compress_voip_start_output_stream(out);
+ else
+ ret = start_output_stream(out);
+ pthread_mutex_unlock(&adev->lock);
+ /* ToDo: If use case is compress offload should return 0 */
+ if (ret != 0) {
+ out->standby = true;
+ goto exit;
+ }
+ }
+
+ if (is_offload_usecase(out->usecase)) {
+ ALOGD("copl(%p): writing buffer (%zu bytes) to compress device", out, bytes);
+ if (out->send_new_metadata) {
+ ALOGD("copl(%p):send new gapless metadata", out);
+ compress_set_gapless_metadata(out->compr, &out->gapless_mdata);
+ out->send_new_metadata = 0;
+ }
+
+ ret = compress_write(out->compr, buffer, bytes);
+ if (ret < 0)
+ ret = -errno;
+ ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret);
+ if (ret >= 0 && ret < (ssize_t)bytes) {
+ ALOGD("No space available in compress driver, post msg to cb thread");
+ send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER);
+ } else if (-ENETRESET == ret) {
+ ALOGE("copl %s: received sound card offline state on compress write", __func__);
+ set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
+ pthread_mutex_unlock(&out->lock);
+ out_standby(&out->stream.common);
+ return ret;
+ }
+ if (!out->playback_started && ret >= 0) {
+ compress_start(out->compr);
+ out->playback_started = 1;
+ out->offload_state = OFFLOAD_STATE_PLAYING;
+ }
+ pthread_mutex_unlock(&out->lock);
+ return ret;
+ } else {
+ if (out->pcm) {
+ if (out->muted)
+ memset((void *)buffer, 0, bytes);
+ ALOGVV("%s: writing buffer (%d bytes) to pcm device", __func__, bytes);
+ if (out->usecase == USECASE_AUDIO_PLAYBACK_AFE_PROXY)
+ ret = pcm_mmap_write(out->pcm, (void *)buffer, bytes);
+ else
+ ret = pcm_write(out->pcm, (void *)buffer, bytes);
+ if (ret < 0)
+ ret = -errno;
+ else if (ret == 0)
+ out->written += bytes / (out->config.channels * sizeof(short));
+ }
+ }
+
+exit:
+ /* ToDo: There may be a corner case when SSR happens back to back during
+ start/stop. Need to post different error to handle that. */
+ if (-ENETRESET == ret) {
+ set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
+ }
+
+ pthread_mutex_unlock(&out->lock);
+
+ if (ret != 0) {
+ if (out->pcm)
+ ALOGE("%s: error %ld - %s", __func__, ret, pcm_get_error(out->pcm));
+ if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
+ pthread_mutex_lock(&adev->lock);
+ voice_extn_compress_voip_close_output_stream(&out->stream.common);
+ pthread_mutex_unlock(&adev->lock);
+ out->standby = true;
+ }
+ out_standby(&out->stream.common);
+ usleep(bytes * 1000000 / audio_stream_out_frame_size(stream) /
+ out_get_sample_rate(&out->stream.common));
+
+ }
+ return bytes;
+}
+
+static int out_get_render_position(const struct audio_stream_out *stream,
+ uint32_t *dsp_frames)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct audio_device *adev = out->dev;
+
+ if (dsp_frames == NULL)
+ return -EINVAL;
+
+ *dsp_frames = 0;
+ if (is_offload_usecase(out->usecase)) {
+ ssize_t ret = 0;
+ pthread_mutex_lock(&out->lock);
+ if (out->compr != NULL) {
+ ret = compress_get_tstamp(out->compr, (unsigned long *)dsp_frames,
+ &out->sample_rate);
+ if (ret < 0)
+ ret = -errno;
+ ALOGVV("%s rendered frames %d sample_rate %d",
+ __func__, *dsp_frames, out->sample_rate);
+ }
+ pthread_mutex_unlock(&out->lock);
+ if (-ENETRESET == ret) {
+ ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
+ set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
+ return -EINVAL;
+ } else if(ret < 0) {
+ ALOGE(" ERROR: Unable to get time stamp from compress driver");
+ return -EINVAL;
+ } else if (get_snd_card_state(adev) == SND_CARD_STATE_OFFLINE){
+ /*
+ * Handle corner case where compress session is closed during SSR
+ * and timestamp is queried
+ */
+ ALOGE(" ERROR: sound card not active, return error");
+ return -EINVAL;
+ } else {
+ return 0;
+ }
+ } else if (audio_is_linear_pcm(out->format)) {
+ *dsp_frames = out->written;
+ return 0;
+ } else
+ return -EINVAL;
+}
+
+static int out_add_audio_effect(const struct audio_stream *stream __unused,
+ effect_handle_t effect __unused)
+{
+ return 0;
+}
+
+static int out_remove_audio_effect(const struct audio_stream *stream __unused,
+ effect_handle_t effect __unused)
+{
+ return 0;
+}
+
+static int out_get_next_write_timestamp(const struct audio_stream_out *stream __unused,
+ int64_t *timestamp __unused)
+{
+ return -EINVAL;
+}
+
+static int out_get_presentation_position(const struct audio_stream_out *stream,
+ uint64_t *frames, struct timespec *timestamp)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int ret = -1;
+ unsigned long dsp_frames;
+
+ pthread_mutex_lock(&out->lock);
+
+ if (is_offload_usecase(out->usecase)) {
+ if (out->compr != NULL) {
+ ret = compress_get_tstamp(out->compr, &dsp_frames,
+ &out->sample_rate);
+ ALOGVV("%s rendered frames %ld sample_rate %d",
+ __func__, dsp_frames, out->sample_rate);
+ *frames = dsp_frames;
+ if (ret < 0)
+ ret = -errno;
+ if (-ENETRESET == ret) {
+ ALOGE(" ERROR: sound card not active Unable to get time stamp from compress driver");
+ set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
+ ret = -EINVAL;
+ } else
+ ret = 0;
+
+ /* this is the best we can do */
+ clock_gettime(CLOCK_MONOTONIC, timestamp);
+ }
+ } else {
+ if (out->pcm) {
+ unsigned int avail;
+ if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
+ size_t kernel_buffer_size = out->config.period_size * out->config.period_count;
+ int64_t signed_frames = out->written - kernel_buffer_size + avail;
+ // This adjustment accounts for buffering after app processor.
+ // It is based on estimated DSP latency per use case, rather than exact.
+ signed_frames -=
+ (platform_render_latency(out->usecase) * out->sample_rate / 1000000LL);
+
+ // It would be unusual for this value to be negative, but check just in case ...
+ if (signed_frames >= 0) {
+ *frames = signed_frames;
+ ret = 0;
+ }
+ }
+ }
+ }
+
+ pthread_mutex_unlock(&out->lock);
+
+ return ret;
+}
+
+static int out_set_callback(struct audio_stream_out *stream,
+ stream_callback_t callback, void *cookie)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+
+ ALOGV("%s", __func__);
+ pthread_mutex_lock(&out->lock);
+ out->offload_callback = callback;
+ out->offload_cookie = cookie;
+ pthread_mutex_unlock(&out->lock);
+ return 0;
+}
+
+static int out_pause(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = -ENOSYS;
+ ALOGV("%s", __func__);
+ if (is_offload_usecase(out->usecase)) {
+ ALOGD("copl(%p):pause compress driver", out);
+ pthread_mutex_lock(&out->lock);
+ if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) {
+ struct audio_device *adev = out->dev;
+ int snd_scard_state = get_snd_card_state(adev);
+
+ if (SND_CARD_STATE_ONLINE == snd_scard_state)
+ status = compress_pause(out->compr);
+
+ out->offload_state = OFFLOAD_STATE_PAUSED;
+ }
+ pthread_mutex_unlock(&out->lock);
+ }
+ return status;
+}
+
+static int out_resume(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = -ENOSYS;
+ ALOGV("%s", __func__);
+ if (is_offload_usecase(out->usecase)) {
+ ALOGD("copl(%p):resume compress driver", out);
+ status = 0;
+ pthread_mutex_lock(&out->lock);
+ if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) {
+ struct audio_device *adev = out->dev;
+ int snd_scard_state = get_snd_card_state(adev);
+
+ if (SND_CARD_STATE_ONLINE == snd_scard_state)
+ status = compress_resume(out->compr);
+
+ out->offload_state = OFFLOAD_STATE_PLAYING;
+ }
+ pthread_mutex_unlock(&out->lock);
+ }
+ return status;
+}
+
+static int out_drain(struct audio_stream_out* stream, audio_drain_type_t type )
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ int status = -ENOSYS;
+ ALOGV("%s", __func__);
+ if (is_offload_usecase(out->usecase)) {
+ pthread_mutex_lock(&out->lock);
+ if (type == AUDIO_DRAIN_EARLY_NOTIFY)
+ status = send_offload_cmd_l(out, OFFLOAD_CMD_PARTIAL_DRAIN);
+ else
+ status = send_offload_cmd_l(out, OFFLOAD_CMD_DRAIN);
+ pthread_mutex_unlock(&out->lock);
+ }
+ return status;
+}
+
+static int out_flush(struct audio_stream_out* stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ ALOGV("%s", __func__);
+ if (is_offload_usecase(out->usecase)) {
+ ALOGD("copl(%p):calling compress flush", out);
+ pthread_mutex_lock(&out->lock);
+ stop_compressed_output_l(out);
+ pthread_mutex_unlock(&out->lock);
+ ALOGD("copl(%p):out of compress flush", out);
+ return 0;
+ }
+ return -ENOSYS;
+}
+
+/** audio_stream_in implementation **/
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+
+ return in->config.rate;
+}
+
+static int in_set_sample_rate(struct audio_stream *stream __unused,
+ uint32_t rate __unused)
+{
+ return -ENOSYS;
+}
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+
+ if(in->usecase == USECASE_COMPRESS_VOIP_CALL)
+ return voice_extn_compress_voip_in_get_buffer_size(in);
+ else if(audio_extn_compr_cap_usecase_supported(in->usecase))
+ return audio_extn_compr_cap_get_buffer_size(in->config.format);
+
+ return in->config.period_size *
+ audio_stream_in_frame_size((const struct audio_stream_in *)stream);
+}
+
+static uint32_t in_get_channels(const struct audio_stream *stream)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+
+ return in->channel_mask;
+}
+
+static audio_format_t in_get_format(const struct audio_stream *stream)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+
+ return in->format;
+}
+
+static int in_set_format(struct audio_stream *stream __unused,
+ audio_format_t format __unused)
+{
+ return -ENOSYS;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+ struct audio_device *adev = in->dev;
+ int status = 0;
+ ALOGD("%s: enter: stream (%p) usecase(%d: %s)", __func__,
+ stream, in->usecase, use_case_table[in->usecase]);
+
+ if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
+ /* Ignore standby in case of voip call because the voip input
+ * stream is closed in adev_close_input_stream()
+ */
+ ALOGV("%s: Ignore Standby in VOIP call", __func__);
+ return status;
+ }
+
+ pthread_mutex_lock(&in->lock);
+ if (!in->standby && in->is_st_session) {
+ ALOGD("%s: sound trigger pcm stop lab", __func__);
+ audio_extn_sound_trigger_stop_lab(in);
+ in->standby = 1;
+ }
+
+ if (!in->standby) {
+ pthread_mutex_lock(&adev->lock);
+ in->standby = true;
+ if (in->pcm) {
+ pcm_close(in->pcm);
+ in->pcm = NULL;
+ }
+ status = stop_input_stream(in);
+ pthread_mutex_unlock(&adev->lock);
+ }
+ pthread_mutex_unlock(&in->lock);
+ ALOGV("%s: exit: status(%d)", __func__, status);
+ return status;
+}
+
+static int in_dump(const struct audio_stream *stream __unused,
+ int fd __unused)
+{
+ return 0;
+}
+
+static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+ struct audio_device *adev = in->dev;
+ struct str_parms *parms;
+ char *str;
+ char value[32];
+ int ret = 0, val = 0, err;
+
+ ALOGD("%s: enter: kvpairs=%s", __func__, kvpairs);
+ parms = str_parms_create_str(kvpairs);
+
+ if (!parms)
+ goto error;
+ pthread_mutex_lock(&in->lock);
+ pthread_mutex_lock(&adev->lock);
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_INPUT_SOURCE, value, sizeof(value));
+ if (err >= 0) {
+ val = atoi(value);
+ /* no audio source uses val == 0 */
+ if ((in->source != val) && (val != 0)) {
+ in->source = val;
+ if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
+ (in->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
+ (voice_extn_compress_voip_is_format_supported(in->format)) &&
+ (in->config.rate == 8000 || in->config.rate == 16000) &&
+ (audio_channel_count_from_in_mask(in->channel_mask) == 1)) {
+ err = voice_extn_compress_voip_open_input_stream(in);
+ if (err != 0) {
+ ALOGE("%s: Compress voip input cannot be opened, error:%d",
+ __func__, err);
+ }
+ }
+ }
+ }
+
+ err = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value));
+ if (err >= 0) {
+ val = atoi(value);
+ if (((int)in->device != val) && (val != 0)) {
+ in->device = val;
+ /* If recording is in progress, change the tx device to new device */
+ if (!in->standby && !in->is_st_session)
+ ret = select_devices(adev, in->usecase);
+ }
+ }
+
+done:
+ pthread_mutex_unlock(&adev->lock);
+ pthread_mutex_unlock(&in->lock);
+
+ str_parms_destroy(parms);
+error:
+ ALOGV("%s: exit: status(%d)", __func__, ret);
+ return ret;
+}
+
+static char* in_get_parameters(const struct audio_stream *stream,
+ const char *keys)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+ struct str_parms *query = str_parms_create_str(keys);
+ char *str;
+ char value[256];
+ struct str_parms *reply = str_parms_create();
+
+ if (!query || !reply) {
+ ALOGE("in_get_parameters: failed to create query or reply");
+ return NULL;
+ }
+
+ ALOGV("%s: enter: keys - %s", __func__, keys);
+
+ voice_extn_in_get_parameters(in, query, reply);
+
+ str = str_parms_to_str(reply);
+ str_parms_destroy(query);
+ str_parms_destroy(reply);
+
+ ALOGV("%s: exit: returns - %s", __func__, str);
+ return str;
+}
+
+static int in_set_gain(struct audio_stream_in *stream __unused,
+ float gain __unused)
+{
+ return 0;
+}
+
+static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
+ size_t bytes)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+ struct audio_device *adev = in->dev;
+ int i, ret = -1;
+ int snd_scard_state = get_snd_card_state(adev);
+
+ pthread_mutex_lock(&in->lock);
+
+ if (in->pcm) {
+ if(SND_CARD_STATE_OFFLINE == snd_scard_state) {
+ ALOGD(" %s: sound card is not active/SSR state", __func__);
+ ret= -EIO;;
+ goto exit;
+ }
+ }
+
+ if (in->standby) {
+ if (!in->is_st_session) {
+ pthread_mutex_lock(&adev->lock);
+ if (in->usecase == USECASE_COMPRESS_VOIP_CALL)
+ ret = voice_extn_compress_voip_start_input_stream(in);
+ else
+ ret = start_input_stream(in);
+ pthread_mutex_unlock(&adev->lock);
+ if (ret != 0) {
+ goto exit;
+ }
+ }
+ in->standby = 0;
+ }
+
+ if (in->pcm) {
+ if (audio_extn_ssr_get_enabled() &&
+ audio_channel_count_from_in_mask(in->channel_mask) == 6)
+ ret = audio_extn_ssr_read(stream, buffer, bytes);
+ else if (audio_extn_compr_cap_usecase_supported(in->usecase))
+ ret = audio_extn_compr_cap_read(in, buffer, bytes);
+ else if (in->usecase == USECASE_AUDIO_RECORD_AFE_PROXY)
+ ret = pcm_mmap_read(in->pcm, buffer, bytes);
+ else
+ ret = pcm_read(in->pcm, buffer, bytes);
+ if (ret < 0)
+ ret = -errno;
+ }
+
+ /*
+ * Instead of writing zeroes here, we could trust the hardware
+ * to always provide zeroes when muted.
+ */
+ if (ret == 0 && voice_get_mic_mute(adev) && !voice_is_in_call_rec_stream(in))
+ memset(buffer, 0, bytes);
+
+exit:
+ /* ToDo: There may be a corner case when SSR happens back to back during
+ start/stop. Need to post different error to handle that. */
+ if (-ENETRESET == ret) {
+ set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
+ }
+ pthread_mutex_unlock(&in->lock);
+
+ if (ret != 0) {
+ if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
+ pthread_mutex_lock(&adev->lock);
+ voice_extn_compress_voip_close_input_stream(&in->stream.common);
+ pthread_mutex_unlock(&adev->lock);
+ in->standby = true;
+ }
+ memset(buffer, 0, bytes);
+ in_standby(&in->stream.common);
+ ALOGV("%s: read failed status %d- sleeping for buffer duration", __func__, ret);
+ usleep(bytes * 1000000 / audio_stream_in_frame_size(stream) /
+ in_get_sample_rate(&in->stream.common));
+ }
+ return bytes;
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream __unused)
+{
+ return 0;
+}
+
+static int add_remove_audio_effect(const struct audio_stream *stream,
+ effect_handle_t effect,
+ bool enable)
+{
+ struct stream_in *in = (struct stream_in *)stream;
+ int status = 0;
+ effect_descriptor_t desc;
+
+ status = (*effect)->get_descriptor(effect, &desc);
+ if (status != 0)
+ return status;
+
+ pthread_mutex_lock(&in->lock);
+ pthread_mutex_lock(&in->dev->lock);
+ if ((in->source == AUDIO_SOURCE_VOICE_COMMUNICATION) &&
+ in->enable_aec != enable &&
+ (memcmp(&desc.type, FX_IID_AEC, sizeof(effect_uuid_t)) == 0)) {
+ in->enable_aec = enable;
+ if (!in->standby)
+ select_devices(in->dev, in->usecase);
+ }
+ if (in->enable_ns != enable &&
+ (memcmp(&desc.type, FX_IID_NS, sizeof(effect_uuid_t)) == 0)) {
+ in->enable_ns = enable;
+ if (!in->standby)
+ select_devices(in->dev, in->usecase);
+ }
+ pthread_mutex_unlock(&in->dev->lock);
+ pthread_mutex_unlock(&in->lock);
+
+ return 0;
+}
+
+static int in_add_audio_effect(const struct audio_stream *stream,
+ effect_handle_t effect)
+{
+ ALOGV("%s: effect %p", __func__, effect);
+ return add_remove_audio_effect(stream, effect, true);
+}
+
+static int in_remove_audio_effect(const struct audio_stream *stream,
+ effect_handle_t effect)
+{
+ ALOGV("%s: effect %p", __func__, effect);
+ return add_remove_audio_effect(stream, effect, false);
+}
+
+static int adev_open_output_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ struct audio_stream_out **stream_out,
+ const char *address __unused)
+{
+ struct audio_device *adev = (struct audio_device *)dev;
+ struct stream_out *out;
+ int i, ret = 0;
+ audio_format_t format;
+
+ *stream_out = NULL;
+
+ if ((flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) &&
+ (SND_CARD_STATE_OFFLINE == get_snd_card_state(adev))) {
+ ALOGE(" sound card is not active rejecting compress output open request");
+ return -EINVAL;
+ }
+
+ out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
+
+ ALOGD("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)\
+ stream_handle(%p)",__func__, config->sample_rate, config->channel_mask,
+ devices, flags, &out->stream);
+
+
+ if (!out) {
+ return -ENOMEM;
+ }
+
+ if (devices == AUDIO_DEVICE_NONE)
+ devices = AUDIO_DEVICE_OUT_SPEAKER;
+
+ out->flags = flags;
+ out->devices = devices;
+ out->dev = adev;
+ format = out->format = config->format;
+ out->sample_rate = config->sample_rate;
+ out->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ out->supported_channel_masks[0] = AUDIO_CHANNEL_OUT_STEREO;
+ out->handle = handle;
+ out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+
+ /* Init use case and pcm_config */
+ if ((out->flags == AUDIO_OUTPUT_FLAG_DIRECT) &&
+ (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL ||
+ out->devices & AUDIO_DEVICE_OUT_PROXY)) {
+
+ pthread_mutex_lock(&adev->lock);
+ if (out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL)
+ ret = read_hdmi_channel_masks(out);
+
+ if (out->devices & AUDIO_DEVICE_OUT_PROXY)
+ ret = audio_extn_read_afe_proxy_channel_masks(out);
+ pthread_mutex_unlock(&adev->lock);
+ if (ret != 0)
+ goto error_open;
+
+ if (config->sample_rate == 0)
+ config->sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
+ if (config->channel_mask == 0)
+ config->channel_mask = AUDIO_CHANNEL_OUT_5POINT1;
+
+ out->channel_mask = config->channel_mask;
+ out->sample_rate = config->sample_rate;
+ out->usecase = USECASE_AUDIO_PLAYBACK_MULTI_CH;
+ out->config = pcm_config_hdmi_multi;
+ out->config.rate = config->sample_rate;
+ out->config.channels = audio_channel_count_from_out_mask(out->channel_mask);
+ out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels * 2);
+ } else if ((out->dev->mode == AUDIO_MODE_IN_COMMUNICATION) &&
+ (out->flags == (AUDIO_OUTPUT_FLAG_DIRECT | AUDIO_OUTPUT_FLAG_VOIP_RX)) &&
+ (voice_extn_compress_voip_is_config_supported(config))) {
+ ret = voice_extn_compress_voip_open_output_stream(out);
+ if (ret != 0) {
+ ALOGE("%s: Compress voip output cannot be opened, error:%d",
+ __func__, ret);
+ goto error_open;
+ }
+ } else if (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_OFFLOAD) {
+ if (config->offload_info.version != AUDIO_INFO_INITIALIZER.version ||
+ config->offload_info.size != AUDIO_INFO_INITIALIZER.size) {
+ ALOGE("%s: Unsupported Offload information", __func__);
+ ret = -EINVAL;
+ goto error_open;
+ }
+ if (!is_supported_format(config->offload_info.format) &&
+ !audio_extn_is_dolby_format(config->offload_info.format)) {
+ ALOGE("%s: Unsupported audio format", __func__);
+ ret = -EINVAL;
+ goto error_open;
+ }
+
+ out->compr_config.codec = (struct snd_codec *)
+ calloc(1, sizeof(struct snd_codec));
+
+ if (!out->compr_config.codec) {
+ ret = -ENOMEM;
+ goto error_open;
+ }
+
+ out->usecase = get_offload_usecase(adev);
+ if (config->offload_info.channel_mask)
+ out->channel_mask = config->offload_info.channel_mask;
+ else if (config->channel_mask) {
+ out->channel_mask = config->channel_mask;
+ config->offload_info.channel_mask = config->channel_mask;
+ }
+ format = out->format = config->offload_info.format;
+ out->sample_rate = config->offload_info.sample_rate;
+
+ out->stream.set_callback = out_set_callback;
+ out->stream.pause = out_pause;
+ out->stream.resume = out_resume;
+ out->stream.drain = out_drain;
+ out->stream.flush = out_flush;
+ out->bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+
+ if (audio_extn_is_dolby_format(config->offload_info.format))
+ out->compr_config.codec->id =
+ audio_extn_dolby_get_snd_codec_id(adev, out,
+ config->offload_info.format);
+ else
+ out->compr_config.codec->id =
+ get_snd_codec_id(config->offload_info.format);
+ if (audio_is_offload_pcm(config->offload_info.format)) {
+ out->compr_config.fragment_size =
+ platform_get_pcm_offload_buffer_size(&config->offload_info);
+ } else {
+ out->compr_config.fragment_size =
+ platform_get_compress_offload_buffer_size(&config->offload_info);
+ }
+ out->compr_config.fragments = COMPRESS_OFFLOAD_NUM_FRAGMENTS;
+ out->compr_config.codec->sample_rate =
+ compress_get_alsa_rate(config->offload_info.sample_rate);
+ out->compr_config.codec->bit_rate =
+ config->offload_info.bit_rate;
+ out->compr_config.codec->ch_in =
+ audio_channel_count_from_out_mask(config->channel_mask);
+ out->compr_config.codec->ch_out = out->compr_config.codec->ch_in;
+ out->bit_width = PCM_OUTPUT_BIT_WIDTH;
+
+ if (config->offload_info.format == AUDIO_FORMAT_AAC)
+ out->compr_config.codec->format = SND_AUDIOSTREAMFORMAT_RAW;
+ if (config->offload_info.format == AUDIO_FORMAT_PCM_16_BIT_OFFLOAD)
+ out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
+ if(config->offload_info.format == AUDIO_FORMAT_PCM_24_BIT_OFFLOAD)
+ out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
+
+ if (out->bit_width == 24) {
+ out->compr_config.codec->format = SNDRV_PCM_FORMAT_S24_LE;
+ }
+
+ if (config->offload_info.format == AUDIO_FORMAT_FLAC)
+ out->compr_config.codec->options.flac_dec.sample_size = PCM_OUTPUT_BIT_WIDTH;
+
+ if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
+ out->non_blocking = 1;
+
+ out->send_new_metadata = 1;
+ out->offload_state = OFFLOAD_STATE_IDLE;
+ out->playback_started = 0;
+
+ create_offload_callback_thread(out);
+ ALOGV("%s: offloaded output offload_info version %04x bit rate %d",
+ __func__, config->offload_info.version,
+ config->offload_info.bit_rate);
+ //Decide if we need to use gapless mode by default
+ check_and_set_gapless_mode(adev);
+
+ } else if (out->flags & AUDIO_OUTPUT_FLAG_INCALL_MUSIC) {
+ ret = voice_check_and_set_incall_music_usecase(adev, out);
+ if (ret != 0) {
+ ALOGE("%s: Incall music delivery usecase cannot be set error:%d",
+ __func__, ret);
+ goto error_open;
+ }
+ } else if (out->devices == AUDIO_DEVICE_OUT_TELEPHONY_TX) {
+ if (config->sample_rate == 0)
+ config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+ if (config->sample_rate != 48000 && config->sample_rate != 16000 &&
+ config->sample_rate != 8000) {
+ config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+ ret = -EINVAL;
+ goto error_open;
+ }
+ out->sample_rate = config->sample_rate;
+ out->config.rate = config->sample_rate;
+ if (config->format == AUDIO_FORMAT_DEFAULT)
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ ret = -EINVAL;
+ goto error_open;
+ }
+ out->format = config->format;
+ out->usecase = USECASE_AUDIO_PLAYBACK_AFE_PROXY;
+ out->config = pcm_config_afe_proxy_playback;
+ adev->voice_tx_output = out;
+ } else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) {
+ format = AUDIO_FORMAT_PCM_16_BIT;
+ out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY;
+ out->config = pcm_config_low_latency;
+ out->sample_rate = out->config.rate;
+ } else {
+ /* primary path is the default path selected if no other outputs are available/suitable */
+ format = AUDIO_FORMAT_PCM_16_BIT;
+ out->usecase = USECASE_AUDIO_PLAYBACK_PRIMARY;
+ out->config = pcm_config_deep_buffer;
+ out->sample_rate = out->config.rate;
+ }
+
+ ALOGV("%s devices %d,flags %x, format %x, out->sample_rate %d, out->bit_width %d",
+ __func__, devices, flags, format, out->sample_rate, out->bit_width);
+ audio_extn_utils_update_stream_app_type_cfg(adev->platform,
+ &adev->streams_output_cfg_list,
+ devices, flags, format, out->sample_rate,
+ out->bit_width, &out->app_type_cfg);
+ if ((out->usecase == USECASE_AUDIO_PLAYBACK_PRIMARY) ||
+ (flags & AUDIO_OUTPUT_FLAG_PRIMARY)) {
+ /* Ensure the default output is not selected twice */
+ if(adev->primary_output == NULL)
+ adev->primary_output = out;
+ else {
+ ALOGE("%s: Primary output is already opened", __func__);
+ ret = -EEXIST;
+ goto error_open;
+ }
+ }
+
+ /* Check if this usecase is already existing */
+ pthread_mutex_lock(&adev->lock);
+ if ((get_usecase_from_list(adev, out->usecase) != NULL) &&
+ (out->usecase != USECASE_COMPRESS_VOIP_CALL)) {
+ ALOGE("%s: Usecase (%d) is already present", __func__, out->usecase);
+ pthread_mutex_unlock(&adev->lock);
+ ret = -EEXIST;
+ goto error_open;
+ }
+ pthread_mutex_unlock(&adev->lock);
+
+ out->stream.common.get_sample_rate = out_get_sample_rate;
+ out->stream.common.set_sample_rate = out_set_sample_rate;
+ out->stream.common.get_buffer_size = out_get_buffer_size;
+ out->stream.common.get_channels = out_get_channels;
+ out->stream.common.get_format = out_get_format;
+ out->stream.common.set_format = out_set_format;
+ out->stream.common.standby = out_standby;
+ out->stream.common.dump = out_dump;
+ out->stream.common.set_parameters = out_set_parameters;
+ out->stream.common.get_parameters = out_get_parameters;
+ out->stream.common.add_audio_effect = out_add_audio_effect;
+ out->stream.common.remove_audio_effect = out_remove_audio_effect;
+ out->stream.get_latency = out_get_latency;
+ out->stream.set_volume = out_set_volume;
+ out->stream.write = out_write;
+ out->stream.get_render_position = out_get_render_position;
+ out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
+ out->stream.get_presentation_position = out_get_presentation_position;
+
+ out->standby = 1;
+ /* out->muted = false; by calloc() */
+ /* out->written = 0; by calloc() */
+
+ pthread_mutex_init(&out->lock, (const pthread_mutexattr_t *) NULL);
+ pthread_cond_init(&out->cond, (const pthread_condattr_t *) NULL);
+
+ config->format = out->stream.common.get_format(&out->stream.common);
+ config->channel_mask = out->stream.common.get_channels(&out->stream.common);
+ config->sample_rate = out->stream.common.get_sample_rate(&out->stream.common);
+
+ *stream_out = &out->stream;
+ ALOGD("%s: Stream (%p) picks up usecase (%s)", __func__, &out->stream,
+ use_case_table[out->usecase]);
+ ALOGV("%s: exit", __func__);
+ return 0;
+
+error_open:
+ free(out);
+ *stream_out = NULL;
+ ALOGD("%s: exit: ret %d", __func__, ret);
+ return ret;
+}
+
+static void adev_close_output_stream(struct audio_hw_device *dev __unused,
+ struct audio_stream_out *stream)
+{
+ struct stream_out *out = (struct stream_out *)stream;
+ struct audio_device *adev = out->dev;
+ int ret = 0;
+
+ ALOGD("%s: enter:stream_handle(%p)",__func__, out);
+
+ if (out->usecase == USECASE_COMPRESS_VOIP_CALL) {
+ pthread_mutex_lock(&adev->lock);
+ ret = voice_extn_compress_voip_close_output_stream(&stream->common);
+ pthread_mutex_unlock(&adev->lock);
+ if(ret != 0)
+ ALOGE("%s: Compress voip output cannot be closed, error:%d",
+ __func__, ret);
+ } else
+ out_standby(&stream->common);
+
+ if (is_offload_usecase(out->usecase)) {
+ destroy_offload_callback_thread(out);
+ free_offload_usecase(adev, out->usecase);
+ if (out->compr_config.codec != NULL)
+ free(out->compr_config.codec);
+ }
+
+ if (adev->voice_tx_output == out)
+ adev->voice_tx_output = NULL;
+
+ pthread_cond_destroy(&out->cond);
+ pthread_mutex_destroy(&out->lock);
+ free(stream);
+ ALOGV("%s: exit", __func__);
+}
+
+static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
+{
+ struct audio_device *adev = (struct audio_device *)dev;
+ struct str_parms *parms;
+ char *str;
+ char value[32];
+ int val;
+ int ret;
+ int status = 0;
+
+ ALOGD("%s: enter: %s", __func__, kvpairs);
+ parms = str_parms_create_str(kvpairs);
+
+ if (!parms)
+ goto error;
+ ret = str_parms_get_str(parms, "SND_CARD_STATUS", value, sizeof(value));
+ if (ret >= 0) {
+ char *snd_card_status = value+2;
+ if (strstr(snd_card_status, "OFFLINE")) {
+ struct listnode *node;
+ struct audio_usecase *usecase;
+
+ ALOGD("Received sound card OFFLINE status");
+ set_snd_card_state(adev,SND_CARD_STATE_OFFLINE);
+
+ pthread_mutex_lock(&adev->lock);
+ //close compress session on OFFLINE status
+ usecase = get_usecase_from_list(adev,USECASE_AUDIO_PLAYBACK_OFFLOAD);
+ if (usecase && usecase->stream.out) {
+ ALOGD(" %s closing compress session on OFFLINE state", __func__);
+
+ struct stream_out *out = usecase->stream.out;
+
+ pthread_mutex_unlock(&adev->lock);
+ out_standby(&out->stream.common);
+ } else
+ pthread_mutex_unlock(&adev->lock);
+ } else if (strstr(snd_card_status, "ONLINE")) {
+ ALOGD("Received sound card ONLINE status");
+ set_snd_card_state(adev,SND_CARD_STATE_ONLINE);
+ if (!platform_is_acdb_initialized(adev->platform)) {
+ ret = platform_acdb_init(adev->platform);
+ if(ret)
+ ALOGE("acdb initialization is failed");
+
+ }
+ }
+ }
+
+ pthread_mutex_lock(&adev->lock);
+ status = voice_set_parameters(adev, parms);
+ if (status != 0)
+ goto done;
+
+ status = platform_set_parameters(adev->platform, parms);
+ if (status != 0)
+ goto done;
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_NREC, value, sizeof(value));
+ if (ret >= 0) {
+ /* When set to false, HAL should disable EC and NS */
+ if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
+ adev->bluetooth_nrec = true;
+ else
+ adev->bluetooth_nrec = false;
+ }
+
+ ret = str_parms_get_str(parms, "screen_state", value, sizeof(value));
+ if (ret >= 0) {
+ if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
+ adev->screen_off = false;
+ else
+ adev->screen_off = true;
+ }
+
+ ret = str_parms_get_int(parms, "rotation", &val);
+ if (ret >= 0) {
+ bool reverse_speakers = false;
+ switch(val) {
+ // FIXME: note that the code below assumes that the speakers are in the correct placement
+ // relative to the user when the device is rotated 90deg from its default rotation. This
+ // assumption is device-specific, not platform-specific like this code.
+ case 270:
+ reverse_speakers = true;
+ break;
+ case 0:
+ case 90:
+ case 180:
+ break;
+ default:
+ ALOGE("%s: unexpected rotation of %d", __func__, val);
+ status = -EINVAL;
+ }
+ if (status == 0) {
+ if (adev->speaker_lr_swap != reverse_speakers) {
+ adev->speaker_lr_swap = reverse_speakers;
+ // only update the selected device if there is active pcm playback
+ struct audio_usecase *usecase;
+ struct listnode *node;
+ list_for_each(node, &adev->usecase_list) {
+ usecase = node_to_item(node, struct audio_usecase, list);
+ if (usecase->type == PCM_PLAYBACK) {
+ select_devices(adev, usecase->id);
+ break;
+ }
+ }
+ }
+ }
+ }
+
+ ret = str_parms_get_str(parms, AUDIO_PARAMETER_KEY_BT_SCO_WB, value, sizeof(value));
+ if (ret >= 0) {
+ if (strcmp(value, AUDIO_PARAMETER_VALUE_ON) == 0)
+ adev->bt_wb_speech_enabled = true;
+ else
+ adev->bt_wb_speech_enabled = false;
+ }
+
+ audio_extn_set_parameters(adev, parms);
+
+done:
+ str_parms_destroy(parms);
+ pthread_mutex_unlock(&adev->lock);
+error:
+ ALOGV("%s: exit with code(%d)", __func__, status);
+ return status;
+}
+
+static char* adev_get_parameters(const struct audio_hw_device *dev,
+ const char *keys)
+{
+ struct audio_device *adev = (struct audio_device *)dev;
+ struct str_parms *reply = str_parms_create();
+ struct str_parms *query = str_parms_create_str(keys);
+ char *str;
+ char value[256] = {0};
+ int ret = 0;
+
+ if (!query || !reply) {
+ ALOGE("adev_get_parameters: failed to create query or reply");
+ return NULL;
+ }
+
+ ret = str_parms_get_str(query, "SND_CARD_STATUS", value,
+ sizeof(value));
+ if (ret >=0) {
+ int val = 1;
+ pthread_mutex_lock(&adev->snd_card_status.lock);
+ if (SND_CARD_STATE_OFFLINE == adev->snd_card_status.state)
+ val = 0;
+ pthread_mutex_unlock(&adev->snd_card_status.lock);
+ str_parms_add_int(reply, "SND_CARD_STATUS", val);
+ goto exit;
+ }
+
+ pthread_mutex_lock(&adev->lock);
+ audio_extn_get_parameters(adev, query, reply);
+ voice_get_parameters(adev, query, reply);
+ platform_get_parameters(adev->platform, query, reply);
+ pthread_mutex_unlock(&adev->lock);
+
+exit:
+ str = str_parms_to_str(reply);
+ str_parms_destroy(query);
+ str_parms_destroy(reply);
+
+ ALOGV("%s: exit: returns - %s", __func__, str);
+ return str;
+}
+
+static int adev_init_check(const struct audio_hw_device *dev __unused)
+{
+ return 0;
+}
+
+static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
+{
+ int ret;
+ struct audio_device *adev = (struct audio_device *)dev;
+ pthread_mutex_lock(&adev->lock);
+ /* cache volume */
+ ret = voice_set_volume(adev, volume);
+ pthread_mutex_unlock(&adev->lock);
+ return ret;
+}
+
+static int adev_set_master_volume(struct audio_hw_device *dev __unused,
+ float volume __unused)
+{
+ return -ENOSYS;
+}
+
+static int adev_get_master_volume(struct audio_hw_device *dev __unused,
+ float *volume __unused)
+{
+ return -ENOSYS;
+}
+
+static int adev_set_master_mute(struct audio_hw_device *dev __unused,
+ bool muted __unused)
+{
+ return -ENOSYS;
+}
+
+static int adev_get_master_mute(struct audio_hw_device *dev __unused,
+ bool *muted __unused)
+{
+ return -ENOSYS;
+}
+
+static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
+{
+ struct audio_device *adev = (struct audio_device *)dev;
+
+ pthread_mutex_lock(&adev->lock);
+ if (adev->mode != mode) {
+ ALOGD("%s: mode %d\n", __func__, mode);
+ adev->mode = mode;
+ if ((mode == AUDIO_MODE_NORMAL || mode == AUDIO_MODE_IN_COMMUNICATION) &&
+ voice_is_in_call(adev)) {
+ voice_stop_call(adev);
+ adev->current_call_output = NULL;
+ }
+ }
+ pthread_mutex_unlock(&adev->lock);
+ return 0;
+}
+
+static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
+{
+ int ret;
+
+ pthread_mutex_lock(&adev->lock);
+ ALOGD("%s state %d\n", __func__, state);
+ ret = voice_set_mic_mute((struct audio_device *)dev, state);
+ pthread_mutex_unlock(&adev->lock);
+
+ return ret;
+}
+
+static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
+{
+ *state = voice_get_mic_mute((struct audio_device *)dev);
+ return 0;
+}
+
+static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev __unused,
+ const struct audio_config *config)
+{
+ int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
+
+ return get_input_buffer_size(config->sample_rate, config->format, channel_count,
+ false /* is_low_latency: since we don't know, be conservative */);
+}
+
+static int adev_open_input_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle __unused,
+ audio_devices_t devices,
+ struct audio_config *config,
+ struct audio_stream_in **stream_in,
+ audio_input_flags_t flags __unused,
+ const char *address __unused,
+ audio_source_t source __unused)
+{
+ struct audio_device *adev = (struct audio_device *)dev;
+ struct stream_in *in;
+ int ret = 0, buffer_size, frame_size;
+ int channel_count = audio_channel_count_from_in_mask(config->channel_mask);
+ bool is_low_latency = false;
+
+ *stream_in = NULL;
+ if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
+ return -EINVAL;
+
+ in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
+
+ if (!in) {
+ ALOGE("failed to allocate input stream");
+ return -ENOMEM;
+ }
+
+ ALOGD("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x)\
+ stream_handle(%p) io_handle(%d)",__func__, config->sample_rate, config->channel_mask,
+ devices, &in->stream, handle);
+
+ pthread_mutex_init(&in->lock, (const pthread_mutexattr_t *) NULL);
+
+ in->stream.common.get_sample_rate = in_get_sample_rate;
+ in->stream.common.set_sample_rate = in_set_sample_rate;
+ in->stream.common.get_buffer_size = in_get_buffer_size;
+ in->stream.common.get_channels = in_get_channels;
+ in->stream.common.get_format = in_get_format;
+ in->stream.common.set_format = in_set_format;
+ in->stream.common.standby = in_standby;
+ in->stream.common.dump = in_dump;
+ in->stream.common.set_parameters = in_set_parameters;
+ in->stream.common.get_parameters = in_get_parameters;
+ in->stream.common.add_audio_effect = in_add_audio_effect;
+ in->stream.common.remove_audio_effect = in_remove_audio_effect;
+ in->stream.set_gain = in_set_gain;
+ in->stream.read = in_read;
+ in->stream.get_input_frames_lost = in_get_input_frames_lost;
+
+ in->device = devices;
+ in->source = AUDIO_SOURCE_DEFAULT;
+ in->dev = adev;
+ in->standby = 1;
+ in->channel_mask = config->channel_mask;
+ in->capture_handle = handle;
+
+ /* Update config params with the requested sample rate and channels */
+ in->usecase = USECASE_AUDIO_RECORD;
+ if (config->sample_rate == LOW_LATENCY_CAPTURE_SAMPLE_RATE &&
+ (flags & AUDIO_INPUT_FLAG_FAST) != 0) {
+ is_low_latency = true;
+#if LOW_LATENCY_CAPTURE_USE_CASE
+ in->usecase = USECASE_AUDIO_RECORD_LOW_LATENCY;
+#endif
+ }
+ in->config = pcm_config_audio_capture;
+ in->config.rate = config->sample_rate;
+ in->format = config->format;
+
+ if (in->device == AUDIO_DEVICE_IN_TELEPHONY_RX) {
+ if (config->sample_rate == 0)
+ config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+ if (config->sample_rate != 48000 && config->sample_rate != 16000 &&
+ config->sample_rate != 8000) {
+ config->sample_rate = AFE_PROXY_SAMPLING_RATE;
+ ret = -EINVAL;
+ goto err_open;
+ }
+ if (config->format == AUDIO_FORMAT_DEFAULT)
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ if (config->format != AUDIO_FORMAT_PCM_16_BIT) {
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ ret = -EINVAL;
+ goto err_open;
+ }
+
+ in->usecase = USECASE_AUDIO_RECORD_AFE_PROXY;
+ in->config = pcm_config_afe_proxy_record;
+ in->config.channels = channel_count;
+ in->config.rate = config->sample_rate;
+ } else if (channel_count == 6) {
+ if(audio_extn_ssr_get_enabled()) {
+ if(audio_extn_ssr_init(in)) {
+ ALOGE("%s: audio_extn_ssr_init failed", __func__);
+ ret = -EINVAL;
+ goto err_open;
+ }
+ } else {
+ ALOGW("%s: surround sound recording is not supported", __func__);
+ }
+ } else if (audio_extn_compr_cap_enabled() &&
+ audio_extn_compr_cap_format_supported(config->format) &&
+ (in->dev->mode != AUDIO_MODE_IN_COMMUNICATION)) {
+ audio_extn_compr_cap_init(in);
+ } else {
+ in->config.channels = channel_count;
+ frame_size = audio_stream_in_frame_size(&in->stream);
+ buffer_size = get_input_buffer_size(config->sample_rate,
+ config->format,
+ channel_count,
+ is_low_latency);
+ in->config.period_size = buffer_size / frame_size;
+ }
+
+ /* This stream could be for sound trigger lab,
+ get sound trigger pcm if present */
+ audio_extn_sound_trigger_check_and_get_session(in);
+ audio_extn_perf_lock_init();
+
+ *stream_in = &in->stream;
+ ALOGV("%s: exit", __func__);
+ return ret;
+
+err_open:
+ free(in);
+ *stream_in = NULL;
+ return ret;
+}
+
+static void adev_close_input_stream(struct audio_hw_device *dev,
+ struct audio_stream_in *stream)
+{
+ int ret;
+ struct stream_in *in = (struct stream_in *)stream;
+ struct audio_device *adev = (struct audio_device *)dev;
+
+ ALOGD("%s: enter:stream_handle(%p)",__func__, in);
+
+ /* Disable echo reference while closing input stream */
+ platform_set_echo_reference(adev->platform, false);
+
+ if (in->usecase == USECASE_COMPRESS_VOIP_CALL) {
+ pthread_mutex_lock(&adev->lock);
+ ret = voice_extn_compress_voip_close_input_stream(&stream->common);
+ pthread_mutex_unlock(&adev->lock);
+ if (ret != 0)
+ ALOGE("%s: Compress voip input cannot be closed, error:%d",
+ __func__, ret);
+ } else
+ in_standby(&stream->common);
+
+ if (audio_extn_ssr_get_enabled() &&
+ (audio_channel_count_from_in_mask(in->channel_mask) == 6)) {
+ audio_extn_ssr_deinit();
+ }
+
+ if(audio_extn_compr_cap_enabled() &&
+ audio_extn_compr_cap_format_supported(in->config.format))
+ audio_extn_compr_cap_deinit();
+
+ free(stream);
+ return;
+}
+
+static int adev_dump(const audio_hw_device_t *device __unused,
+ int fd __unused)
+{
+ return 0;
+}
+
+static int adev_close(hw_device_t *device)
+{
+ struct audio_device *adev = (struct audio_device *)device;
+
+ if (!adev)
+ return 0;
+
+ pthread_mutex_lock(&adev_init_lock);
+
+ if ((--audio_device_ref_count) == 0) {
+ audio_extn_sound_trigger_deinit(adev);
+ audio_extn_listen_deinit(adev);
+ audio_extn_utils_release_streams_output_cfg_list(&adev->streams_output_cfg_list);
+ audio_route_free(adev->audio_route);
+ free(adev->snd_dev_ref_cnt);
+ platform_deinit(adev->platform);
+ free(device);
+ adev = NULL;
+ }
+ pthread_mutex_unlock(&adev_init_lock);
+ return 0;
+}
+
+/* This returns 1 if the input parameter looks at all plausible as a low latency period size,
+ * or 0 otherwise. A return value of 1 doesn't mean the value is guaranteed to work,
+ * just that it _might_ work.
+ */
+static int period_size_is_plausible_for_low_latency(int period_size)
+{
+ switch (period_size) {
+ case 160:
+ case 240:
+ case 320:
+ case 480:
+ return 1;
+ default:
+ return 0;
+ }
+}
+
+static int adev_open(const hw_module_t *module, const char *name,
+ hw_device_t **device)
+{
+ int i, ret;
+
+ ALOGD("%s: enter", __func__);
+ if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) return -EINVAL;
+
+ pthread_mutex_lock(&adev_init_lock);
+ if (audio_device_ref_count != 0){
+ *device = &adev->device.common;
+ audio_device_ref_count++;
+ ALOGD("%s: returning existing instance of adev", __func__);
+ ALOGD("%s: exit", __func__);
+ pthread_mutex_unlock(&adev_init_lock);
+ return 0;
+ }
+
+ adev = calloc(1, sizeof(struct audio_device));
+
+ if (!adev) {
+ pthread_mutex_unlock(&adev_init_lock);
+ return -ENOMEM;
+ }
+
+ pthread_mutex_init(&adev->lock, (const pthread_mutexattr_t *) NULL);
+
+ adev->device.common.tag = HARDWARE_DEVICE_TAG;
+ adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
+ adev->device.common.module = (struct hw_module_t *)module;
+ adev->device.common.close = adev_close;
+
+ adev->device.init_check = adev_init_check;
+ adev->device.set_voice_volume = adev_set_voice_volume;
+ adev->device.set_master_volume = adev_set_master_volume;
+ adev->device.get_master_volume = adev_get_master_volume;
+ adev->device.set_master_mute = adev_set_master_mute;
+ adev->device.get_master_mute = adev_get_master_mute;
+ adev->device.set_mode = adev_set_mode;
+ adev->device.set_mic_mute = adev_set_mic_mute;
+ adev->device.get_mic_mute = adev_get_mic_mute;
+ adev->device.set_parameters = adev_set_parameters;
+ adev->device.get_parameters = adev_get_parameters;
+ adev->device.get_input_buffer_size = adev_get_input_buffer_size;
+ adev->device.open_output_stream = adev_open_output_stream;
+ adev->device.close_output_stream = adev_close_output_stream;
+ adev->device.open_input_stream = adev_open_input_stream;
+ adev->device.close_input_stream = adev_close_input_stream;
+ adev->device.dump = adev_dump;
+
+ /* Set the default route before the PCM stream is opened */
+ adev->mode = AUDIO_MODE_NORMAL;
+ adev->active_input = NULL;
+ adev->primary_output = NULL;
+ adev->out_device = AUDIO_DEVICE_NONE;
+ adev->bluetooth_nrec = true;
+ adev->acdb_settings = TTY_MODE_OFF;
+ /* adev->cur_hdmi_channels = 0; by calloc() */
+ adev->cur_codec_backend_samplerate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
+ adev->cur_codec_backend_bit_width = CODEC_BACKEND_DEFAULT_BIT_WIDTH;
+ adev->snd_dev_ref_cnt = calloc(SND_DEVICE_MAX, sizeof(int));
+ voice_init(adev);
+ list_init(&adev->usecase_list);
+ adev->cur_wfd_channels = 2;
+ adev->offload_usecases_state = 0;
+
+ pthread_mutex_init(&adev->snd_card_status.lock, (const pthread_mutexattr_t *) NULL);
+ adev->snd_card_status.state = SND_CARD_STATE_OFFLINE;
+ /* Loads platform specific libraries dynamically */
+ adev->platform = platform_init(adev);
+ if (!adev->platform) {
+ free(adev->snd_dev_ref_cnt);
+ free(adev);
+ ALOGE("%s: Failed to init platform data, aborting.", __func__);
+ *device = NULL;
+ pthread_mutex_unlock(&adev_init_lock);
+ return -EINVAL;
+ }
+
+ adev->snd_card_status.state = SND_CARD_STATE_ONLINE;
+
+ if (access(VISUALIZER_LIBRARY_PATH, R_OK) == 0) {
+ adev->visualizer_lib = dlopen(VISUALIZER_LIBRARY_PATH, RTLD_NOW);
+ if (adev->visualizer_lib == NULL) {
+ ALOGE("%s: DLOPEN failed for %s", __func__, VISUALIZER_LIBRARY_PATH);
+ } else {
+ ALOGV("%s: DLOPEN successful for %s", __func__, VISUALIZER_LIBRARY_PATH);
+ adev->visualizer_start_output =
+ (int (*)(audio_io_handle_t, int))dlsym(adev->visualizer_lib,
+ "visualizer_hal_start_output");
+ adev->visualizer_stop_output =
+ (int (*)(audio_io_handle_t, int))dlsym(adev->visualizer_lib,
+ "visualizer_hal_stop_output");
+ }
+ }
+ audio_extn_listen_init(adev, adev->snd_card);
+ audio_extn_sound_trigger_init(adev);
+
+ if (access(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, R_OK) == 0) {
+ adev->offload_effects_lib = dlopen(OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH, RTLD_NOW);
+ if (adev->offload_effects_lib == NULL) {
+ ALOGE("%s: DLOPEN failed for %s", __func__,
+ OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
+ } else {
+ ALOGV("%s: DLOPEN successful for %s", __func__,
+ OFFLOAD_EFFECTS_BUNDLE_LIBRARY_PATH);
+ adev->offload_effects_start_output =
+ (int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
+ "offload_effects_bundle_hal_start_output");
+ adev->offload_effects_stop_output =
+ (int (*)(audio_io_handle_t, int))dlsym(adev->offload_effects_lib,
+ "offload_effects_bundle_hal_stop_output");
+ }
+ }
+
+ adev->bt_wb_speech_enabled = false;
+
+ audio_extn_ds2_enable(adev);
+ *device = &adev->device.common;
+
+ audio_extn_utils_update_streams_output_cfg_list(adev->platform, adev->mixer,
+ &adev->streams_output_cfg_list);
+
+ audio_device_ref_count++;
+
+ char value[PROPERTY_VALUE_MAX];
+ int trial;
+ if (property_get("audio_hal.period_size", value, NULL) > 0) {
+ trial = atoi(value);
+ if (period_size_is_plausible_for_low_latency(trial)) {
+ pcm_config_low_latency.period_size = trial;
+ pcm_config_low_latency.start_threshold = trial / 4;
+ pcm_config_low_latency.avail_min = trial / 4;
+ configured_low_latency_capture_period_size = trial;
+ }
+ }
+ if (property_get("audio_hal.in_period_size", value, NULL) > 0) {
+ trial = atoi(value);
+ if (period_size_is_plausible_for_low_latency(trial)) {
+ configured_low_latency_capture_period_size = trial;
+ }
+ }
+
+ pthread_mutex_unlock(&adev_init_lock);
+
+ ALOGV("%s: exit", __func__);
+ return 0;
+}
+
+static struct hw_module_methods_t hal_module_methods = {
+ .open = adev_open,
+};
+
+struct audio_module HAL_MODULE_INFO_SYM = {
+ .common = {
+ .tag = HARDWARE_MODULE_TAG,
+ .module_api_version = AUDIO_MODULE_API_VERSION_0_1,
+ .hal_api_version = HARDWARE_HAL_API_VERSION,
+ .id = AUDIO_HARDWARE_MODULE_ID,
+ .name = "QCOM Audio HAL",
+ .author = "The Linux Foundation",
+ .methods = &hal_module_methods,
+ },
+};