Merge 9c4ba54b3ecb1ee854763970b3057220994dad8a on remote branch (61716b1b) · Commits · e / devices / Fairphone 4 / android_hardware_qcom_audio

audiod/AudioDaemon.cpp

+1 −1

Original line number	Diff line number	Diff line
		/* AudioDaemon.cpp
		Copyright (c) 2012-2014, The Linux Foundation. All rights reserved.
		Copyright (c) 2012-2015, The Linux Foundation. All rights reserved.

		Redistribution and use in source and binary forms, with or without
		modification, are permitted provided that the following conditions are

hal/audio_extn/audio_extn.h

+2 −0

Original line number	Diff line number	Diff line
		@@ -384,6 +384,7 @@ void audio_extn_dolby_send_ddp_endp_params(struct audio_device *adev);
		#define audio_extn_dolby_get_passt_buffer_size(info) (0)
		#define audio_extn_dolby_set_passt_volume(out, mute) (0)
		#define audio_extn_dolby_set_passt_latency(out, latency) (0)
		#define AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH 0x4000
		#else
		int audio_extn_dolby_update_passt_formats(struct audio_device *adev,
		struct stream_out *out);
		@@ -445,6 +446,7 @@ void audio_extn_utils_update_stream_app_type_cfg(void *platform,
		audio_format_t format,
		uint32_t sample_rate,
		uint32_t bit_width,
		audio_channel_mask_t channel_mask,
		struct stream_app_type_cfg *app_type_cfg);
		int audio_extn_utils_send_app_type_cfg(struct audio_device *adev,
		struct audio_usecase *usecase);

hal/audio_extn/dolby.c

100755 → 100644

+3 −1

Original line number	Diff line number	Diff line
		/*
		* Copyright (c) 2011-2014, The Linux Foundation. All rights reserved.
		* Copyright (c) 2011-2015, The Linux Foundation. All rights reserved.
		* Not a Contribution.
		*
		* Copyright (C) 2010 The Android Open Source Project
		@@ -609,6 +609,8 @@ void audio_extn_dolby_set_license(struct audio_device *adev)
		property_get("audio.ds1.metainfo.key",value,"0");
		#ifdef DOLBY_ACDB_LICENSE
		key = atoi(value);
		#else
		key = 0;
		#endif
		ALOGV("%s Setting DS1 License, key:0x%x",__func__, key);
		ret = mixer_ctl_set_value(ctl, 0, key);

hal/audio_extn/spkr_protection.c

+12 −9

Original line number	Diff line number	Diff line
		@@ -197,8 +197,6 @@ int get_tzn(const char *sensor_name)
		ALOGE("Unable to open %s\n", THERMAL_SYSFS);
		return found;
		}
		if (!sensor_name)
		return found;

		while ((tdirent = readdir(tdir))) {
		char buf[50];
		@@ -212,12 +210,11 @@ int get_tzn(const char *sensor_name)
		if (strcmp(tzdirent->d_name, "type"))
		continue;
		snprintf(name, MAX_PATH, TZ_TYPE, tzn);
		ALOGV("Opening %s\n", name);
		ALOGD("Opening %s\n", name);
		read_line_from_file(name, buf, sizeof(buf));
		if (strlen(buf) > 0)
		buf[strlen(buf) - 1] = '\0';
		if (!strcmp(buf, sensor_name)) {
		ALOGD(" spkr tz name found, %s\n", name);
		found = 1;
		break;
		}
		@@ -1070,17 +1067,21 @@ int audio_extn_spkr_prot_get_acdb_id(snd_device_t snd_device)

		switch(snd_device) {
		case SND_DEVICE_OUT_SPEAKER:
		case SND_DEVICE_OUT_SPEAKER_WSA:
		acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_PROTECTED);
		break;
		case SND_DEVICE_OUT_VOICE_SPEAKER:
		acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED);
		break;
		case SND_DEVICE_OUT_SPEAKER_VBAT:
		acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT);
		break;
		case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
		acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT);
		break;
		case SND_DEVICE_OUT_VOICE_SPEAKER:

		case SND_DEVICE_OUT_VOICE_SPEAKER_WSA:

		acdb_id = platform_get_snd_device_acdb_id(SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED);
		break;
		default:
		acdb_id = -EINVAL;
		break;
		@@ -1095,13 +1096,15 @@ int audio_extn_get_spkr_prot_snd_device(snd_device_t snd_device)

		switch(snd_device) {
		case SND_DEVICE_OUT_SPEAKER:
		case SND_DEVICE_OUT_SPEAKER_WSA:
		return SND_DEVICE_OUT_SPEAKER_PROTECTED;
		case SND_DEVICE_OUT_VOICE_SPEAKER:
		return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
		case SND_DEVICE_OUT_SPEAKER_VBAT:
		return SND_DEVICE_OUT_SPEAKER_PROTECTED_VBAT;
		case SND_DEVICE_OUT_VOICE_SPEAKER_VBAT:
		return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED_VBAT;
		case SND_DEVICE_OUT_VOICE_SPEAKER:
		case SND_DEVICE_OUT_VOICE_SPEAKER_WSA:
		return SND_DEVICE_OUT_VOICE_SPEAKER_PROTECTED;
		default:
		return snd_device;
		}

hal/audio_extn/utils.c

+57 −52

Original line number	Diff line number	Diff line
		@@ -158,14 +158,15 @@ static uint32_t string_to_enum(const struct string_to_enum *table, size_t size,
		static audio_output_flags_t parse_flag_names(char *name)
		{
		uint32_t flag = 0;
		char *flag_name = strtok(name, "\|");
		char *last_r;
		char *flag_name = strtok_r(name, "\|", &last_r);
		while (flag_name != NULL) {
		if (strlen(flag_name) != 0) {
		flag \|= string_to_enum(s_flag_name_to_enum_table,
		ARRAY_SIZE(s_flag_name_to_enum_table),
		flag_name);
		}
		flag_name = strtok(NULL, "\|");
		flag_name = strtok_r(NULL, "\|", &last_r);
		}

		ALOGV("parse_flag_names: flag - %d", flag);
		@@ -175,7 +176,8 @@ static audio_output_flags_t parse_flag_names(char *name)
		static void parse_format_names(char name, struct streams_output_cfg so_info)
		{
		struct stream_format *sf_info = NULL;
		char *str = strtok(name, "\|");
		char *last_r;
		char *str = strtok_r(name, "\|", &last_r);

		if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG) == 0)
		return;
		@@ -193,7 +195,7 @@ static void parse_format_names(char name, struct streams_output_cfg so_info)
		sf_info->format = format;
		list_add_tail(&so_info->format_list, &sf_info->list);
		}
		str = strtok(NULL, "\|");
		str = strtok_r(NULL, "\|", &last_r);
		}
		}

		@@ -201,7 +203,8 @@ static void parse_sample_rate_names(char name, struct streams_output_cfg so_in
		{
		struct stream_sample_rate *ss_info = NULL;
		uint32_t sample_rate = 48000;
		char *str = strtok(name, "\|");
		char *last_r;
		char *str = strtok_r(name, "\|", &last_r);

		if (str != NULL && 0 == strcmp(str, DYNAMIC_VALUE_TAG))
		return;
		@@ -212,20 +215,22 @@ static void parse_sample_rate_names(char name, struct streams_output_cfg so_in
		ALOGV("%s: sample_rate - %d", __func__, sample_rate);
		if (0 != sample_rate) {
		ss_info = (struct stream_sample_rate *)calloc(1, sizeof(struct stream_sample_rate));
		if (ss_info == NULL)
		break; /* return whatever was parsed */

		if (!ss_info) {
		ALOGE("%s: memory allocation failure", __func__);
		return;
		}
		ss_info->sample_rate = sample_rate;
		list_add_tail(&so_info->sample_rate_list, &ss_info->list);
		}
		str = strtok(NULL, "\|");
		str = strtok_r(NULL, "\|", &last_r);
		}
		}

		static int parse_bit_width_names(char *name)
		{
		int bit_width = 16;
		char *str = strtok(name, "\|");
		char *last_r;
		char *str = strtok_r(name, "\|", &last_r);

		if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG))
		bit_width = (int)strtol(str, (char **)NULL, 10);
		@@ -237,7 +242,8 @@ static int parse_bit_width_names(char *name)
		static int parse_app_type_names(void platform, char name)
		{
		int app_type = platform_get_default_app_type(platform);
		char *str = strtok(name, "\|");
		char *last_r;
		char *str = strtok_r(name, "\|", &last_r);

		if (str != NULL && strcmp(str, DYNAMIC_VALUE_TAG))
		app_type = (int)strtol(str, (char **)NULL, 10);
		@@ -477,12 +483,14 @@ void audio_extn_utils_update_stream_app_type_cfg(void *platform,
		audio_format_t format,
		uint32_t sample_rate,
		uint32_t bit_width,
		audio_channel_mask_t channel_mask,
		struct stream_app_type_cfg *app_type_cfg)
		{
		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;
		char value[PROPERTY_VALUE_MAX] = {0};

		if ((24 == bit_width) &&
		(devices & AUDIO_DEVICE_OUT_SPEAKER)) {
		@@ -493,6 +501,16 @@ void audio_extn_utils_update_stream_app_type_cfg(void *platform,
		ALOGI("%s Allowing 24-bit playback on speaker ONLY at default sampling rate", __func__);
		}

		property_get("audio.playback.mch.downsample",value,"");
		if (!strncmp("true", value, sizeof("true"))) {
		if ((popcount(channel_mask) > 2) &&
		(sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
		!(flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)) {
		sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
		ALOGD("%s: MCH session defaulting sample rate to %d",
		__func__, sample_rate);
		}
		}
		ALOGV("%s: flags: %x, format: %x sample_rate %d",
		__func__, flags, format, sample_rate);
		list_for_each(node_i, streams_output_cfg_list) {
		@@ -533,6 +551,7 @@ int audio_extn_utils_send_app_type_cfg(struct audio_device *adev,
		struct mixer_ctl *ctl;
		int pcm_device_id, acdb_dev_id, snd_device = usecase->out_snd_device;
		int32_t sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
		char value[PROPERTY_VALUE_MAX] = {0};

		ALOGV("%s", __func__);

		@@ -576,16 +595,6 @@ int audio_extn_utils_send_app_type_cfg(struct audio_device *adev,
		rc = -EINVAL;
		goto exit_send_app_type_cfg;
		}

		if ((usecase->type == PCM_PLAYBACK) && (usecase->stream.out == NULL)) {
		sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
		app_type_cfg[len++] = platform_get_default_app_type_v2(adev->platform, usecase->type);
		app_type_cfg[len++] = acdb_dev_id;
		app_type_cfg[len++] = sample_rate;
		ALOGI("%s PLAYBACK app_type %d, acdb_dev_id %d, sample_rate %d",
		__func__, platform_get_default_app_type_v2(adev->platform, usecase->type),
		acdb_dev_id, sample_rate);
		} else if (usecase->type == PCM_PLAYBACK) {
		if ((24 == usecase->stream.out->bit_width) &&
		(usecase->stream.out->devices & AUDIO_DEVICE_OUT_SPEAKER)) {
		usecase->stream.out->app_type_cfg.sample_rate = DEFAULT_OUTPUT_SAMPLING_RATE;
		@@ -596,27 +605,22 @@ int audio_extn_utils_send_app_type_cfg(struct audio_device *adev,
		}
		sample_rate = usecase->stream.out->app_type_cfg.sample_rate;

		property_get("audio.playback.mch.downsample",value,"");
		if (!strncmp("true", value, sizeof("true"))) {
		if ((popcount(usecase->stream.out->channel_mask) > 2) &&
		(usecase->stream.out->app_type_cfg.sample_rate > CODEC_BACKEND_DEFAULT_SAMPLE_RATE) &&
		!(usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))
		sample_rate = CODEC_BACKEND_DEFAULT_SAMPLE_RATE;
		}

		app_type_cfg[len++] = usecase->stream.out->app_type_cfg.app_type;
		app_type_cfg[len++] = acdb_dev_id;
		if (((usecase->stream.out->format == AUDIO_FORMAT_E_AC3) \|\|
		(usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC))
		#ifdef HDMI_PASSTHROUGH_ENABLED
		&& (out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH)
		#endif
		)
		(usecase->stream.out->format == AUDIO_FORMAT_E_AC3_JOC)) &&
		(usecase->stream.out->flags & AUDIO_OUTPUT_FLAG_COMPRESS_PASSTHROUGH))
		app_type_cfg[len++] = sample_rate * 4;
		else
		app_type_cfg[len++] = sample_rate;
		ALOGI("%s PLAYBACK app_type %d, acdb_dev_id %d, sample_rate %d",
		__func__, usecase->stream.out->app_type_cfg.app_type, acdb_dev_id, sample_rate);
		} else if (usecase->type == PCM_CAPTURE) {
		app_type_cfg[len++] = platform_get_default_app_type_v2(adev->platform, usecase->type);
		app_type_cfg[len++] = acdb_dev_id;
		app_type_cfg[len++] = sample_rate;
		ALOGI("%s CAPTURE app_type %d, acdb_dev_id %d, sample_rate %d",
		__func__, platform_get_default_app_type_v2(adev->platform, usecase->type),
		acdb_dev_id, sample_rate);
		}
		mixer_ctl_set_array(ctl, app_type_cfg, len);
		rc = 0;
		exit_send_app_type_cfg:
		@@ -656,7 +660,7 @@ void audio_extn_utils_send_audio_calibration(struct audio_device *adev,
		audio_extn_get_spkr_prot_snd_device(snd_device) : snd_device;
		platform_send_audio_calibration(adev->platform, usecase,
		out->app_type_cfg.app_type,
		out->app_type_cfg.sample_rate);
		usecase->stream.out->app_type_cfg.sample_rate);
		}
		if ((type == PCM_HFP_CALL) \|\| (type == PCM_CAPTURE)) {
		/* when app type is default. the sample rate is not used to send cal */
		@@ -875,6 +879,7 @@ void get_lpcm_channel_status(uint32_t sampleRate,
		case 16000:
		case 22050:
		channel_status[3] \|= SR_NOTID;
		break;
		case 24000:
		channel_status[3] \|= SR_24000;
		break;