diff --git a/hal_mpq/Android.mk b/hal_mpq/Android.mk index 338ee401..24fca4ea 100644 --- a/hal_mpq/Android.mk +++ b/hal_mpq/Android.mk @@ -10,6 +10,7 @@ AUDIO_PLATFORM := $(TARGET_BOARD_PLATFORM) LOCAL_SRC_FILES := \ audio_hw.c \ + audio_stream_out.c \ $(AUDIO_PLATFORM)/hw_info.c \ $(AUDIO_PLATFORM)/platform.c diff --git a/hal_mpq/audio_hw.c b/hal_mpq/audio_hw.c index 8df46be8..fb83245e 100644 --- a/hal_mpq/audio_hw.c +++ b/hal_mpq/audio_hw.c @@ -40,135 +40,26 @@ #include #include #include -#include #include -#include #include #include #include "audio_hw.h" #include "platform_api.h" #include -#include "sound/compress_params.h" - -#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024) -#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 - -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, }; -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", - [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_VOICE_CALL] = "voice-call", - - [USECASE_VOICE2_CALL] = "voice2-call", - [USECASE_VOLTE_CALL] = "volte-call", - [USECASE_QCHAT_CALL] = "qchat-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_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", -}; - - -#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 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 bool is_supported_format(audio_format_t format) -{ - if (format == AUDIO_FORMAT_MP3 || - format == AUDIO_FORMAT_AAC) - return true; - - return false; -} - -static int get_snd_codec_id(audio_format_t format) -{ - int id = 0; - - switch (format) { - case AUDIO_FORMAT_MP3: - id = SND_AUDIOCODEC_MP3; - break; - case AUDIO_FORMAT_AAC: - id = SND_AUDIOCODEC_AAC; - break; - default: - ALOGE("%s: Unsupported audio format", __func__); - } - - return id; -} - int enable_audio_route(struct audio_device *adev, struct audio_usecase *usecase, bool update_mixer) @@ -483,34 +374,6 @@ static int enable_all_usecases_of_type(struct audio_device *adev, return ret; } -/* 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; @@ -751,327 +614,6 @@ error_config: 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)); - - 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); - } - } -} - -static void *offload_thread_loop(void *context) -{ - struct stream_out *out = (struct stream_out *) context; - struct listnode *item; - - out->offload_state = OFFLOAD_STATE_IDLE; - out->playback_started = 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: - compress_wait(out->compr, -1); - send_callback = true; - event = STREAM_CBK_EVENT_WRITE_READY; - break; - case OFFLOAD_CMD_PARTIAL_DRAIN: - compress_next_track(out->compr); - compress_partial_drain(out->compr); - send_callback = true; - event = STREAM_CBK_EVENT_DRAIN_READY; - break; - case OFFLOAD_CMD_DRAIN: - compress_drain(out->compr); - 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; - } - } - } - 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", __func__); - 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, true); - } - } - - /* - * 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, true); - } - } - - 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD && - adev->visualizer_stop_output != NULL) - adev->visualizer_stop_output(out->handle); - - /* 1. Get and set stream specific mixer controls */ - disable_audio_route(adev, uc_info, true); - - /* 2. Disable the rx device */ - disable_snd_device(adev, uc_info->out_snd_device, true); - - 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; - struct audio_usecase *uc_info; - struct audio_device *adev = out->dev; - - ALOGV("%s: enter: usecase(%d: %s) devices(%#x)", - __func__, out->usecase, use_case_table[out->usecase], out->devices); - 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)); - 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) - 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)", - __func__, 0, out->pcm_device_id); - if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) { - out->pcm = pcm_open(SOUND_CARD, out->pcm_device_id, - PCM_OUT | PCM_MONOTONIC, &out->config); - if (out->pcm && !pcm_is_ready(out->pcm)) { - ALOGE("%s: %s", __func__, pcm_get_error(out->pcm)); - pcm_close(out->pcm); - out->pcm = NULL; - ret = -EIO; - goto error_open; - } - } else { - out->pcm = NULL; - out->compr = compress_open(SOUND_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); - - if (adev->visualizer_start_output != NULL) - adev->visualizer_start_output(out->handle); - } - ALOGV("%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) @@ -1128,493 +670,6 @@ static size_t get_input_buffer_size(uint32_t sample_rate, 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, uint32_t rate) -{ - return -ENOSYS; -} - -static size_t out_get_buffer_size(const struct audio_stream *stream) -{ - struct stream_out *out = (struct stream_out *)stream; - - if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) - return out->compr_config.fragment_size; - - return out->config.period_size * audio_stream_frame_size(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, audio_format_t format) -{ - return -ENOSYS; -} - -static int out_standby(struct audio_stream *stream) -{ - struct stream_out *out = (struct stream_out *)stream; - struct audio_device *adev = out->dev; - - ALOGV("%s: enter: usecase(%d: %s)", __func__, - 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() - */ - ALOGV("%s: Ignore Standby in VOIP call", __func__); - return 0; - } - - pthread_mutex_lock(&out->lock); - pthread_mutex_lock(&adev->lock); - if (!out->standby) { - out->standby = true; - if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) { - if (out->pcm) { - pcm_close(out->pcm); - out->pcm = NULL; - } - } else { - 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, int fd) -{ - return 0; -} - -static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms) -{ - int ret = 0; - char value[32]; - struct compr_gapless_mdata tmp_mdata; - - if (!out || !parms) { - return -EINVAL; - } - - ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value)); - if (ret >= 0) { - tmp_mdata.encoder_delay = atoi(value); //whats a good limit check? - } else { - return -EINVAL; - } - - ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value)); - if (ret >= 0) { - tmp_mdata.encoder_padding = atoi(value); - } else { - return -EINVAL; - } - - out->gapless_mdata = tmp_mdata; - 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 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, val = 0; - 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); - ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); - if (ret >= 0) { - val = atoi(value); - pthread_mutex_lock(&out->lock); - pthread_mutex_lock(&adev->lock); - - /* - * When HDMI cable is unplugged the music playback is paused and - * the policy manager sends routing=0. But the audioflinger - * continues to write data until standby time (3sec). - * As the HDMI core is turned off, the write gets blocked. - * Avoid this by routing audio to speaker until standby. - */ - if (out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL && - val == AUDIO_DEVICE_NONE) { - 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); - } - - pthread_mutex_unlock(&adev->lock); - pthread_mutex_unlock(&out->lock); - } - - if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { - parse_compress_metadata(out, parms); - } - - str_parms_destroy(parms); - 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; - ALOGV("%s: enter: keys - %s", __func__, keys); - ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value)); - if (ret >= 0) { - value[0] = '\0'; - i = 0; - while (out->supported_channel_masks[i] != 0) { - for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) { - if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) { - if (!first) { - strlcat(value, "|", sizeof(value)); - } - strlcat(value, out_channels_name_to_enum_table[j].name, sizeof(value)); - first = false; - break; - } - } - i++; - } - str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value); - str = str_parms_to_str(reply); - } - 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) - 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { - const char *mixer_ctl_name = "Compress Playback Volume"; - struct audio_device *adev = out->dev; - struct mixer_ctl *ctl; - - ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name); - if (!ctl) { - ALOGE("%s: Could not get ctl for mixer cmd - %s", - __func__, mixer_ctl_name); - return -EINVAL; - } - volume[0] = (int)(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; - ssize_t ret = 0; - - pthread_mutex_lock(&out->lock); - if (out->standby) { - out->standby = false; - pthread_mutex_lock(&adev->lock); - 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { - ALOGVV("%s: writing buffer (%d bytes) to compress device", __func__, bytes); - if (out->send_new_metadata) { - ALOGVV("send new gapless metadata"); - compress_set_gapless_metadata(out->compr, &out->gapless_mdata); - out->send_new_metadata = 0; - } - - ret = compress_write(out->compr, buffer, bytes); - ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret); - if (ret >= 0 && ret < (ssize_t)bytes) { - send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER); - } - if (!out->playback_started) { - 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); - ret = pcm_write(out->pcm, (void *)buffer, bytes); - if (ret == 0) - out->written += bytes / (out->config.channels * sizeof(short)); - } - } - -exit: - pthread_mutex_unlock(&out->lock); - - if (ret != 0) { - if (out->pcm) - ALOGE("%s: error %d - %s", __func__, ret, pcm_get_error(out->pcm)); - out_standby(&out->stream.common); - usleep(bytes * 1000000 / audio_stream_frame_size(&out->stream.common) / - 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; - *dsp_frames = 0; - if ((out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) && (dsp_frames != NULL)) { - pthread_mutex_lock(&out->lock); - if (out->compr != NULL) { - compress_get_tstamp(out->compr, (unsigned long *)dsp_frames, - &out->sample_rate); - ALOGVV("%s rendered frames %d sample_rate %d", - __func__, *dsp_frames, out->sample_rate); - } - pthread_mutex_unlock(&out->lock); - return 0; - } else - return -EINVAL; -} - -static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) -{ - return 0; -} - -static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) -{ - return 0; -} - -static int out_get_next_write_timestamp(const struct audio_stream_out *stream, - int64_t *timestamp) -{ - 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { - if (out->compr != NULL) { - 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; - ret = 0; - /* this is the best we can do */ - clock_gettime(CLOCK_MONOTONIC, timestamp); - } - } else { - if (out->pcm) { - size_t 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { - pthread_mutex_lock(&out->lock); - if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) { - 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { - status = 0; - pthread_mutex_lock(&out->lock); - if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) { - 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { - 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { - pthread_mutex_lock(&out->lock); - stop_compressed_output_l(out); - pthread_mutex_unlock(&out->lock); - return 0; - } - return -ENOSYS; -} - /** audio_stream_in implementation **/ static uint32_t in_get_sample_rate(const struct audio_stream *stream) { @@ -1801,29 +856,6 @@ static int add_remove_audio_effect(const struct audio_stream *stream, { 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; } @@ -1842,198 +874,6 @@ static int in_remove_audio_effect(const struct audio_stream *stream, 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) -{ - struct audio_device *adev = (struct audio_device *)dev; - struct stream_out *out; - int i, ret; - - ALOGV("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)", - __func__, config->sample_rate, config->channel_mask, devices, flags); - *stream_out = NULL; - out = (struct stream_out *)calloc(1, sizeof(struct stream_out)); - - if (devices == AUDIO_DEVICE_NONE) - devices = AUDIO_DEVICE_OUT_SPEAKER; - - out->flags = flags; - out->devices = devices; - out->dev = adev; - 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; - - /* Init use case and pcm_config */ - if (out->flags == AUDIO_OUTPUT_FLAG_DIRECT && - out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { - pthread_mutex_lock(&adev->lock); - ret = read_hdmi_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 = popcount(out->channel_mask); - out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels * 2); - } 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)) { - ALOGE("%s: Unsupported audio format", __func__); - ret = -EINVAL; - goto error_open; - } - - out->compr_config.codec = (struct snd_codec *) - calloc(1, sizeof(struct snd_codec)); - - out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD; - 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; - 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->compr_config.codec->id = - get_snd_codec_id(config->offload_info.format); - out->compr_config.fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE; - 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 = - popcount(config->channel_mask); - out->compr_config.codec->ch_out = out->compr_config.codec->ch_in; - - if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) - out->non_blocking = 1; - - out->send_new_metadata = 1; - 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); - } else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) { - out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY; - out->config = pcm_config_low_latency; - out->sample_rate = out->config.rate; - } else { - out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER; - out->config = pcm_config_deep_buffer; - out->sample_rate = out->config.rate; - } - - if (flags & AUDIO_OUTPUT_FLAG_PRIMARY) { - 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) { - 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; - 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, - struct audio_stream_out *stream) -{ - struct stream_out *out = (struct stream_out *)stream; - struct audio_device *adev = out->dev; - int ret = 0; - - ALOGV("%s: enter", __func__); - out_standby(&stream->common); - - if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { - destroy_offload_callback_thread(out); - - if (out->compr_config.codec != NULL) - free(out->compr_config.codec); - } - 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; diff --git a/hal_mpq/audio_hw.h b/hal_mpq/audio_hw.h index 262fda84..e1319a5f 100644 --- a/hal_mpq/audio_hw.h +++ b/hal_mpq/audio_hw.h @@ -220,6 +220,41 @@ struct audio_device { int (*visualizer_stop_output)(audio_io_handle_t); }; +static 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", + [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_VOICE_CALL] = "voice-call", + + [USECASE_VOICE2_CALL] = "voice2-call", + [USECASE_VOLTE_CALL] = "volte-call", + [USECASE_QCHAT_CALL] = "qchat-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_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", +}; + +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); + +void adev_close_output_stream(struct audio_hw_device *dev, + struct audio_stream_out *stream); + int select_devices(struct audio_device *adev, audio_usecase_t uc_id); int disable_audio_route(struct audio_device *adev, diff --git a/hal_mpq/audio_stream_out.c b/hal_mpq/audio_stream_out.c new file mode 100644 index 00000000..71eb2f1d --- /dev/null +++ b/hal_mpq/audio_stream_out.c @@ -0,0 +1,1157 @@ +/* audio_stream_out.c + ** + ** Copyright 2008-2009 Wind River Systems + ** Copyright (c) 2011-2013, The Linux Foundation. All rights reserved + ** Not a Contribution, Apache license notifications and license are retained + ** for attribution purposes only. + ** + ** 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_stream_out" +/*#define LOG_NDEBUG 0*/ +/*#define VERY_VERY_VERBOSE_LOGGING*/ +#ifdef VERY_VERY_VERBOSE_LOGGING +#define ALOGVV ALOGV +#else +#define ALOGVV(a...) do { } while(0) +#endif + +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include +#include +#include +#include + +#include +#include "audio_hw.h" +#include "platform_api.h" +#include + +#include "sound/compress_params.h" + +#define COMPRESS_OFFLOAD_FRAGMENT_SIZE (32 * 1024) +#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 + +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, +}; + +#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 bool is_supported_format(audio_format_t format) +{ + if (format == AUDIO_FORMAT_MP3 || + format == AUDIO_FORMAT_AAC) + return true; + + return false; +} + +static int get_snd_codec_id(audio_format_t format) +{ + int id = 0; + + switch (format) { + case AUDIO_FORMAT_MP3: + id = SND_AUDIOCODEC_MP3; + break; + case AUDIO_FORMAT_AAC: + id = SND_AUDIOCODEC_AAC; + break; + default: + ALOGE("%s: Unsupported audio format", __func__); + } + + return id; +} + +/* 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; +} + +/* 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)); + + 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); + } + } +} + +static void *offload_thread_loop(void *context) +{ + struct stream_out *out = (struct stream_out *) context; + struct listnode *item; + + out->offload_state = OFFLOAD_STATE_IDLE; + out->playback_started = 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: + compress_wait(out->compr, -1); + send_callback = true; + event = STREAM_CBK_EVENT_WRITE_READY; + break; + case OFFLOAD_CMD_PARTIAL_DRAIN: + compress_next_track(out->compr); + compress_partial_drain(out->compr); + send_callback = true; + event = STREAM_CBK_EVENT_DRAIN_READY; + break; + case OFFLOAD_CMD_DRAIN: + compress_drain(out->compr); + 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; + } + } + } + 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", __func__); + 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, true); + } + } + + /* + * 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, true); + } + } + + 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD && + adev->visualizer_stop_output != NULL) + adev->visualizer_stop_output(out->handle); + + /* 1. Get and set stream specific mixer controls */ + disable_audio_route(adev, uc_info, true); + + /* 2. Disable the rx device */ + disable_snd_device(adev, uc_info->out_snd_device, true); + + 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; + struct audio_usecase *uc_info; + struct audio_device *adev = out->dev; + + ALOGV("%s: enter: usecase(%d: %s) devices(%#x)", + __func__, out->usecase, use_case_table[out->usecase], out->devices); + 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)); + 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) + 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)", + __func__, 0, out->pcm_device_id); + if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) { + out->pcm = pcm_open(SOUND_CARD, out->pcm_device_id, + PCM_OUT | PCM_MONOTONIC, &out->config); + if (out->pcm && !pcm_is_ready(out->pcm)) { + ALOGE("%s: %s", __func__, pcm_get_error(out->pcm)); + pcm_close(out->pcm); + out->pcm = NULL; + ret = -EIO; + goto error_open; + } + } else { + out->pcm = NULL; + out->compr = compress_open(SOUND_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); + + if (adev->visualizer_start_output != NULL) + adev->visualizer_start_output(out->handle); + } + ALOGV("%s: exit", __func__); + return 0; +error_open: + stop_output_stream(out); +error_config: + return ret; +} + +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, uint32_t rate) +{ + return -ENOSYS; +} + +static size_t out_get_buffer_size(const struct audio_stream *stream) +{ + struct stream_out *out = (struct stream_out *)stream; + + if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) + return out->compr_config.fragment_size; + + return out->config.period_size * audio_stream_frame_size(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, audio_format_t format) +{ + return -ENOSYS; +} + +static int out_standby(struct audio_stream *stream) +{ + struct stream_out *out = (struct stream_out *)stream; + struct audio_device *adev = out->dev; + + ALOGV("%s: enter: usecase(%d: %s)", __func__, + 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() + */ + ALOGV("%s: Ignore Standby in VOIP call", __func__); + return 0; + } + + pthread_mutex_lock(&out->lock); + pthread_mutex_lock(&adev->lock); + if (!out->standby) { + out->standby = true; + if (out->usecase != USECASE_AUDIO_PLAYBACK_OFFLOAD) { + if (out->pcm) { + pcm_close(out->pcm); + out->pcm = NULL; + } + } else { + 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, int fd) +{ + return 0; +} + +static int parse_compress_metadata(struct stream_out *out, struct str_parms *parms) +{ + int ret = 0; + char value[32]; + struct compr_gapless_mdata tmp_mdata; + + if (!out || !parms) { + return -EINVAL; + } + + ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_DELAY_SAMPLES, value, sizeof(value)); + if (ret >= 0) { + tmp_mdata.encoder_delay = atoi(value); //whats a good limit check? + } else { + return -EINVAL; + } + + ret = str_parms_get_str(parms, AUDIO_OFFLOAD_CODEC_PADDING_SAMPLES, value, sizeof(value)); + if (ret >= 0) { + tmp_mdata.encoder_padding = atoi(value); + } else { + return -EINVAL; + } + + out->gapless_mdata = tmp_mdata; + 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 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, val = 0; + 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); + ret = str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)); + if (ret >= 0) { + val = atoi(value); + pthread_mutex_lock(&out->lock); + pthread_mutex_lock(&adev->lock); + + /* + * When HDMI cable is unplugged the music playback is paused and + * the policy manager sends routing=0. But the audioflinger + * continues to write data until standby time (3sec). + * As the HDMI core is turned off, the write gets blocked. + * Avoid this by routing audio to speaker until standby. + */ + if (out->devices == AUDIO_DEVICE_OUT_AUX_DIGITAL && + val == AUDIO_DEVICE_NONE) { + 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); + } + + pthread_mutex_unlock(&adev->lock); + pthread_mutex_unlock(&out->lock); + } + + if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { + parse_compress_metadata(out, parms); + } + + str_parms_destroy(parms); + 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; + ALOGV("%s: enter: keys - %s", __func__, keys); + ret = str_parms_get_str(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value, sizeof(value)); + if (ret >= 0) { + value[0] = '\0'; + i = 0; + while (out->supported_channel_masks[i] != 0) { + for (j = 0; j < ARRAY_SIZE(out_channels_name_to_enum_table); j++) { + if (out_channels_name_to_enum_table[j].value == out->supported_channel_masks[i]) { + if (!first) { + strlcat(value, "|", sizeof(value)); + } + strlcat(value, out_channels_name_to_enum_table[j].name, sizeof(value)); + first = false; + break; + } + } + i++; + } + str_parms_add_str(reply, AUDIO_PARAMETER_STREAM_SUP_CHANNELS, value); + str = str_parms_to_str(reply); + } + 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) + 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { + const char *mixer_ctl_name = "Compress Playback Volume"; + struct audio_device *adev = out->dev; + struct mixer_ctl *ctl; + + ctl = mixer_get_ctl_by_name(adev->mixer, mixer_ctl_name); + if (!ctl) { + ALOGE("%s: Could not get ctl for mixer cmd - %s", + __func__, mixer_ctl_name); + return -EINVAL; + } + volume[0] = (int)(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; + ssize_t ret = 0; + + pthread_mutex_lock(&out->lock); + if (out->standby) { + out->standby = false; + pthread_mutex_lock(&adev->lock); + 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { + ALOGVV("%s: writing buffer (%d bytes) to compress device", __func__, bytes); + if (out->send_new_metadata) { + ALOGVV("send new gapless metadata"); + compress_set_gapless_metadata(out->compr, &out->gapless_mdata); + out->send_new_metadata = 0; + } + + ret = compress_write(out->compr, buffer, bytes); + ALOGVV("%s: writing buffer (%d bytes) to compress device returned %d", __func__, bytes, ret); + if (ret >= 0 && ret < (ssize_t)bytes) { + send_offload_cmd_l(out, OFFLOAD_CMD_WAIT_FOR_BUFFER); + } + if (!out->playback_started) { + 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); + ret = pcm_write(out->pcm, (void *)buffer, bytes); + if (ret == 0) + out->written += bytes / (out->config.channels * sizeof(short)); + } + } + +exit: + pthread_mutex_unlock(&out->lock); + + if (ret != 0) { + if (out->pcm) + ALOGE("%s: error %d - %s", __func__, ret, pcm_get_error(out->pcm)); + out_standby(&out->stream.common); + usleep(bytes * 1000000 / audio_stream_frame_size(&out->stream.common) / + 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; + *dsp_frames = 0; + if ((out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) && (dsp_frames != NULL)) { + pthread_mutex_lock(&out->lock); + if (out->compr != NULL) { + compress_get_tstamp(out->compr, (unsigned long *)dsp_frames, + &out->sample_rate); + ALOGVV("%s rendered frames %d sample_rate %d", + __func__, *dsp_frames, out->sample_rate); + } + pthread_mutex_unlock(&out->lock); + return 0; + } else + return -EINVAL; +} + +static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) +{ + return 0; +} + +static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) +{ + return 0; +} + +static int out_get_next_write_timestamp(const struct audio_stream_out *stream, + int64_t *timestamp) +{ + 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { + if (out->compr != NULL) { + 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; + ret = 0; + /* this is the best we can do */ + clock_gettime(CLOCK_MONOTONIC, timestamp); + } + } else { + if (out->pcm) { + size_t 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { + pthread_mutex_lock(&out->lock); + if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PLAYING) { + 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { + status = 0; + pthread_mutex_lock(&out->lock); + if (out->compr != NULL && out->offload_state == OFFLOAD_STATE_PAUSED) { + 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { + 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 (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { + pthread_mutex_lock(&out->lock); + stop_compressed_output_l(out); + pthread_mutex_unlock(&out->lock); + return 0; + } + return -ENOSYS; +} + +int adev_open_output_stream(struct audio_hw_device *dev, + audio_io_handle_t handle, + audio_devices_t devices, + audio_output_flags_t flags, + struct audio_config *config, + struct audio_stream_out **stream_out) +{ + struct audio_device *adev = (struct audio_device *)dev; + struct stream_out *out; + int i, ret; + + ALOGV("%s: enter: sample_rate(%d) channel_mask(%#x) devices(%#x) flags(%#x)", + __func__, config->sample_rate, config->channel_mask, devices, flags); + *stream_out = NULL; + out = (struct stream_out *)calloc(1, sizeof(struct stream_out)); + + if (devices == AUDIO_DEVICE_NONE) + devices = AUDIO_DEVICE_OUT_SPEAKER; + + out->flags = flags; + out->devices = devices; + out->dev = adev; + 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; + + /* Init use case and pcm_config */ + if (out->flags == AUDIO_OUTPUT_FLAG_DIRECT && + out->devices & AUDIO_DEVICE_OUT_AUX_DIGITAL) { + pthread_mutex_lock(&adev->lock); + ret = read_hdmi_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 = popcount(out->channel_mask); + out->config.period_size = HDMI_MULTI_PERIOD_BYTES / (out->config.channels * 2); + } 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)) { + ALOGE("%s: Unsupported audio format", __func__); + ret = -EINVAL; + goto error_open; + } + + out->compr_config.codec = (struct snd_codec *) + calloc(1, sizeof(struct snd_codec)); + + out->usecase = USECASE_AUDIO_PLAYBACK_OFFLOAD; + 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; + 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->compr_config.codec->id = + get_snd_codec_id(config->offload_info.format); + out->compr_config.fragment_size = COMPRESS_OFFLOAD_FRAGMENT_SIZE; + 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 = + popcount(config->channel_mask); + out->compr_config.codec->ch_out = out->compr_config.codec->ch_in; + + if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING) + out->non_blocking = 1; + + out->send_new_metadata = 1; + 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); + } else if (out->flags & AUDIO_OUTPUT_FLAG_FAST) { + out->usecase = USECASE_AUDIO_PLAYBACK_LOW_LATENCY; + out->config = pcm_config_low_latency; + out->sample_rate = out->config.rate; + } else { + out->usecase = USECASE_AUDIO_PLAYBACK_DEEP_BUFFER; + out->config = pcm_config_deep_buffer; + out->sample_rate = out->config.rate; + } + + if (flags & AUDIO_OUTPUT_FLAG_PRIMARY) { + 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) { + 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; + ALOGV("%s: exit", __func__); + return 0; + +error_open: + free(out); + *stream_out = NULL; + ALOGD("%s: exit: ret %d", __func__, ret); + return ret; +} + +void adev_close_output_stream(struct audio_hw_device *dev, + struct audio_stream_out *stream) +{ + struct stream_out *out = (struct stream_out *)stream; + struct audio_device *adev = out->dev; + int ret = 0; + + ALOGV("%s: enter", __func__); + out_standby(&stream->common); + + if (out->usecase == USECASE_AUDIO_PLAYBACK_OFFLOAD) { + destroy_offload_callback_thread(out); + + if (out->compr_config.codec != NULL) + free(out->compr_config.codec); + } + pthread_cond_destroy(&out->cond); + pthread_mutex_destroy(&out->lock); + free(stream); + ALOGV("%s: exit", __func__); +}