Merge "hal: Fix policy conf parser for FLAC playback on multiple sample rate" (79123b00) · Commits · e / devices / Fairphone 4 / android_hardware_qcom_audio

hal/audio_extn/audio_extn.h

+2 −0

Original line number	Diff line number	Diff line
		@@ -283,6 +283,8 @@ void audio_extn_utils_update_stream_app_type_cfg(void *platform,
		struct listnode *streams_output_cfg_list,
		audio_output_flags_t flags,
		audio_format_t format,
		uint32_t sample_rate,
		uint32_t bit_width,
		struct stream_app_type_cfg *app_type_cfg);
		int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase);
		void audio_extn_utils_send_audio_calibration(struct audio_device *adev,

hal/audio_extn/utils.c

+80 −17

Original line number	Diff line number	Diff line
		@@ -93,6 +93,7 @@ const struct string_to_enum s_format_name_to_enum_table[] = {
		STRING_TO_ENUM(AUDIO_FORMAT_EVRCNW),
		STRING_TO_ENUM(AUDIO_FORMAT_PCM_16_BIT_OFFLOAD),
		STRING_TO_ENUM(AUDIO_FORMAT_PCM_24_BIT_OFFLOAD),
		STRING_TO_ENUM(AUDIO_FORMAT_FLAC),
		#endif
		};

		@@ -151,16 +152,26 @@ static void parse_format_names(char name, struct streams_output_cfg so_info)
		}
		}

		static int parse_sample_rate_names(char *name)
		static void parse_sample_rate_names(char name, struct streams_output_cfg so_info)
		{
		int sample_rate = 48000;
		struct stream_sample_rate *ss_info = NULL;
		uint32_t sample_rate = 48000;
		char *str = strtok(name, "\|");

		if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG))
		sample_rate = (int)strtol(str, (char **)NULL, 10);
		if (str != NULL && 0 == strcmp(str, DYNAMIC_VALUE_TAG))
		return;

		list_init(&so_info->sample_rate_list);
		while (str != NULL) {
		sample_rate = (uint32_t)strtol(str, (char **)NULL, 10);
		ALOGV("%s: sample_rate - %d", __func__, sample_rate);
		return sample_rate;
		if (0 != sample_rate) {
		ss_info = (struct stream_sample_rate *)calloc(1, sizeof(struct stream_sample_rate));
		ss_info->sample_rate = sample_rate;
		list_add_tail(&so_info->sample_rate_list, &ss_info->list);
		}
		str = strtok(NULL, "\|");
		}
		}

		static int parse_bit_width_names(char *name)
		@@ -207,7 +218,8 @@ static void update_streams_output_cfg_list(cnode root, void platform,
		} else if (strcmp(node->name, FORMATS_TAG) == 0) {
		parse_format_names((char *)node->value, so_info);
		} else if (strcmp(node->name, SAMPLING_RATES_TAG) == 0) {
		so_info->app_type_cfg.sample_rate = parse_sample_rate_names((char *)node->value);
		so_info->app_type_cfg.sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
		parse_sample_rate_names((char *)node->value, so_info);
		} else if (strcmp(node->name, BIT_WIDTH_TAG) == 0) {
		so_info->app_type_cfg.bit_width = parse_bit_width_names((char *)node->value);
		} else if (strcmp(node->name, APP_TYPE_TAG) == 0) {
		@@ -325,16 +337,21 @@ void audio_extn_utils_dump_streams_output_cfg_list(
		struct listnode node_i, node_j;
		struct streams_output_cfg *so_info;
		struct stream_format *sf_info;
		struct stream_sample_rate *ss_info;
		ALOGV("%s", __func__);
		list_for_each(node_i, streams_output_cfg_list) {
		so_info = node_to_item(node_i, struct streams_output_cfg, list);
		ALOGV("%d: flags-%d, output_sample_rate-%d, output_bit_width-%d, app_type-%d",
		i++, so_info->flags, so_info->app_type_cfg.sample_rate,
		ALOGV("%s: flags-%d, output_sample_rate-%d, output_bit_width-%d, app_type-%d",
		__func__, so_info->flags, so_info->app_type_cfg.sample_rate,
		so_info->app_type_cfg.bit_width, so_info->app_type_cfg.app_type);
		list_for_each(node_j, &so_info->format_list) {
		sf_info = node_to_item(node_j, struct stream_format, list);
		ALOGV("format-%x", sf_info->format);
		}
		list_for_each(node_j, &so_info->sample_rate_list) {
		ss_info = node_to_item(node_j, struct stream_sample_rate, list);
		ALOGV("sample rate-%d", ss_info->sample_rate);
		}
		}
		}

		@@ -354,20 +371,67 @@ void audio_extn_utils_release_streams_output_cfg_list(
		list_remove(node_j);
		free(node_to_item(node_j, struct stream_format, list));
		}
		while (!list_empty(&so_info->sample_rate_list)) {
		node_j = list_head(&so_info->sample_rate_list);
		list_remove(node_j);
		free(node_to_item(node_j, struct stream_sample_rate, list));
		}
		list_remove(node_i);
		free(node_to_item(node_i, struct streams_output_cfg, list));
		}
		}

		static bool set_output_cfg(struct streams_output_cfg *so_info,
		struct stream_app_type_cfg *app_type_cfg,
		uint32_t sample_rate, uint32_t bit_width)
		{
		struct listnode *node_i;
		struct stream_sample_rate *ss_info;
		list_for_each(node_i, &so_info->sample_rate_list) {
		ss_info = node_to_item(node_i, struct stream_sample_rate, list);
		if ((sample_rate <= ss_info->sample_rate) &&
		(bit_width == so_info->app_type_cfg.bit_width)) {
		app_type_cfg->app_type = so_info->app_type_cfg.app_type;
		app_type_cfg->sample_rate = ss_info->sample_rate;
		app_type_cfg->bit_width = so_info->app_type_cfg.bit_width;
		ALOGV("%s app_type_cfg->app_type %d, app_type_cfg->sample_rate %d, app_type_cfg->bit_width %d",
		__func__, app_type_cfg->app_type, app_type_cfg->sample_rate, app_type_cfg->bit_width);
		return true;
		}
		}
		/*
		* Reiterate through the list assuming dafault sample rate.
		* Handles scenario where input sample rate is higher
		* than all sample rates in list for the input bit width.
		*/
		sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
		list_for_each(node_i, &so_info->sample_rate_list) {
		ss_info = node_to_item(node_i, struct stream_sample_rate, list);
		if ((sample_rate <= ss_info->sample_rate) &&
		(bit_width == so_info->app_type_cfg.bit_width)) {
		app_type_cfg->app_type = so_info->app_type_cfg.app_type;
		app_type_cfg->sample_rate = sample_rate;
		app_type_cfg->bit_width = so_info->app_type_cfg.bit_width;
		ALOGV("%s Assuming default sample rate. app_type_cfg->app_type %d, app_type_cfg->sample_rate %d, app_type_cfg->bit_width %d",
		__func__, app_type_cfg->app_type, app_type_cfg->sample_rate, app_type_cfg->bit_width);
		return true;
		}
		}
		return false;
		}

		void audio_extn_utils_update_stream_app_type_cfg(void *platform,
		struct listnode *streams_output_cfg_list,
		audio_output_flags_t flags,
		audio_format_t format,
		uint32_t sample_rate,
		uint32_t bit_width,
		struct stream_app_type_cfg *app_type_cfg)
		{
		struct listnode node_i, node_j;
		struct listnode node_i, node_j, *node_k;
		struct streams_output_cfg *so_info;
		struct stream_format *sf_info;
		struct stream_sample_rate *ss_info;

		ALOGV("%s: flags: %x, format: %x", __func__, flags, format);
		list_for_each(node_i, streams_output_cfg_list) {
		@@ -376,10 +440,7 @@ void audio_extn_utils_update_stream_app_type_cfg(void *platform,
		list_for_each(node_j, &so_info->format_list) {
		sf_info = node_to_item(node_j, struct stream_format, list);
		if (sf_info->format == format) {
		ALOGV("App type: %d", so_info->app_type_cfg.app_type);
		app_type_cfg->app_type = so_info->app_type_cfg.app_type;
		app_type_cfg->sample_rate = so_info->app_type_cfg.sample_rate;
		app_type_cfg->bit_width = so_info->app_type_cfg.bit_width;
		if (set_output_cfg(so_info, app_type_cfg, sample_rate, bit_width))
		return;
		}
		}
		@@ -389,16 +450,17 @@ void audio_extn_utils_update_stream_app_type_cfg(void *platform,
		so_info = node_to_item(node_i, struct streams_output_cfg, list);
		if (so_info->flags == AUDIO_OUTPUT_FLAG_PRIMARY) {
		ALOGV("Compatible output profile not found.");
		ALOGV("App type default to primary output: %d", so_info->app_type_cfg.app_type);
		app_type_cfg->app_type = so_info->app_type_cfg.app_type;
		app_type_cfg->sample_rate = so_info->app_type_cfg.sample_rate;
		app_type_cfg->bit_width = so_info->app_type_cfg.bit_width;
		ALOGV("%s Default to primary output: App type: %d sample_rate %d",
		__func__, so_info->app_type_cfg.app_type, app_type_cfg->sample_rate);
		return;
		}
		}
		ALOGW("%s: App type could not be selected. Falling back to default", __func__);
		app_type_cfg->app_type = platform_get_default_app_type(platform);
		app_type_cfg->sample_rate = 48000;
		app_type_cfg->sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
		app_type_cfg->bit_width = 16;
		}

		@@ -453,6 +515,7 @@ int audio_extn_utils_send_app_type_cfg(struct audio_usecase *usecase)
		}
		app_type_cfg[len++] = out->app_type_cfg.app_type;
		app_type_cfg[len++] = acdb_dev_id;
		app_type_cfg[len++] = out->app_type_cfg.sample_rate;

		mixer_ctl_set_array(ctl, app_type_cfg, len);

hal/audio_hw.c

+4 −2

Original line number	Diff line number	Diff line
		@@ -2504,7 +2504,8 @@ static int adev_open_output_stream(struct audio_hw_device *dev,
		out->compr_config.codec->format = SNDRV_PCM_FORMAT_S16_LE;
		}

		out->compr_config.codec->options.flac_dec.sample_size = out->bit_width;
		if (config->offload_info.format == AUDIO_FORMAT_FLAC)
		out->compr_config.codec->options.flac_dec.sample_size = config->offload_info.bit_width;

		if (flags & AUDIO_OUTPUT_FLAG_NON_BLOCKING)
		out->non_blocking = 1;
		@@ -2542,7 +2543,8 @@ static int adev_open_output_stream(struct audio_hw_device *dev,

		audio_extn_utils_update_stream_app_type_cfg(adev->platform,
		&adev->streams_output_cfg_list,
		flags, format, &out->app_type_cfg);
		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 */

hal/audio_hw.h

+6 −0

Original line number	Diff line number	Diff line
		@@ -246,10 +246,16 @@ struct stream_format {
		audio_format_t format;
		};

		struct stream_sample_rate {
		struct listnode list;
		uint32_t sample_rate;
		};

		struct streams_output_cfg {
		struct listnode list;
		audio_output_flags_t flags;
		struct listnode format_list;
		struct listnode sample_rate_list;
		struct stream_app_type_cfg app_type_cfg;
		};