Audio: update hal, add mixer pcmdai parser (74a314b7) · Commits · e / devices / samsung / android_hardware_samsung

audio/Android.mk

+7 −5

Original line number	Diff line number	Diff line
		@@ -25,7 +25,8 @@ LOCAL_SRC_FILES := \
		audio_hw.c \
		compress_offload.c \
		ril_interface.c \
		voice.c
		voice.c \
		mixer_util.c

		# TODO: remove resampler if possible when AudioFlinger supports downsampling from 48 to 8
		LOCAL_SHARED_LIBRARIES := \
		@@ -37,7 +38,8 @@ LOCAL_SHARED_LIBRARIES := \
		libtinycompress \
		libaudioroute \
		libdl \
		libsecril-client
		libsecril-client \
		libexpat


		LOCAL_C_INCLUDES += \
		@@ -48,11 +50,11 @@ LOCAL_C_INCLUDES += \
		hardware/samsung/ril/libsecril-client \
		$(call include-path-for, audio-utils) \
		$(call include-path-for, audio-route) \
		$(call include-path-for, audio-effects)
		$(call include-path-for, audio-effects) \
		external/expat/lib

		LOCAL_CFLAGS := -Werror -Wall
		LOCAL_CFLAGS := -Wall
		#LOCAL_CFLAGS += -DPREPROCESSING_ENABLED
		#LOCAL_CFLAGS += -DHW_AEC_LOOPBACK

		LOCAL_MODULE := audio.primary.$(TARGET_BOOTLOADER_BOARD_NAME)

audio/audio_hw.c

+757 −328

File changed.

Preview size limit exceeded, changes collapsed.

audio/audio_hw.h

+36 −14

Original line number	Diff line number	Diff line
		/*
		* Copyright (C) 2013 The Android Open Source Project
		* Copyright (C) 2017 Christopher N. Hesse <raymanfx@gmail.com>
		* Copyright (C) 2018 The LineageOS Project
		*
		* Licensed under the Apache License, Version 2.0 (the "License");
		* you may not use this file except in compliance with the License.
		@@ -186,6 +187,9 @@ typedef enum {
		USECASE_AUDIO_CAPTURE,

		USECASE_VOICE_CALL,

		USECASE_BT_SCO,
		USECASE_FM_RADIO,
		AUDIO_USECASE_MAX
		} audio_usecase_t;

		@@ -227,14 +231,28 @@ struct offload_cmd {
		int data[];
		};

		struct pcm_device_profile {
		struct pcm_config config;
		int card;
		int id;
		usecase_type_t type;
		audio_devices_t devices;
		};

		struct pcm_device {
		struct listnode stream_list_node;
		struct pcm_device_profile* pcm_profile;
		struct pcm* pcm;
		int status;
		};

		struct stream_out {
		struct audio_stream_out stream;
		pthread_mutex_t lock; /* see note below on mutex acquisition order */
		pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by playback thread */
		pthread_cond_t cond;
		struct pcm_config config;
		int pcm_device_id;
		struct pcm *pcm;
		struct listnode pcm_dev_list;
		struct compr_config compr_config;
		struct compress* compr;
		int standby;
		@@ -285,8 +303,7 @@ struct stream_in {
		pthread_mutex_t pre_lock; /* acquire before lock to avoid DOS by
		capture thread */
		struct pcm_config config;
		struct pcm *pcm;
		int pcm_device_id;
		struct listnode pcm_dev_list;
		int standby;
		audio_source_t source;
		audio_devices_t devices;
		@@ -315,13 +332,6 @@ struct stream_in {
		int16_t *ref_buf;
		size_t ref_buf_size;
		size_t ref_buf_frames;

		#ifdef HW_AEC_LOOPBACK
		bool hw_echo_reference;
		int16_t* hw_ref_buf;
		size_t hw_ref_buf_size;
		#endif

		int num_preprocessors;
		struct effect_info_s preprocessors[MAX_PREPROCESSORS];

		@@ -337,6 +347,15 @@ struct stream_in {
		entering standby */
		};

		struct mixer_card {
		struct listnode adev_list_node;
		struct listnode uc_list_node[AUDIO_USECASE_MAX];
		int card;
		struct mixer* mixer;
		struct audio_route* audio_route;
		struct timespec dsp_poweroff_time;
		};

		struct audio_usecase {
		struct listnode adev_list_node;
		audio_usecase_t id;
		@@ -345,6 +364,7 @@ struct audio_usecase {
		snd_device_t out_snd_device;
		snd_device_t in_snd_device;
		struct audio_stream* stream;
		struct listnode mixer_list;
		};

		struct voice_data {
		@@ -358,12 +378,10 @@ struct voice_data {
		struct audio_device {
		struct audio_hw_device device;
		pthread_mutex_t lock; /* see note below on mutex acquisition order */
		struct mixer *mixer;
		struct audio_route *audio_route;
		struct listnode mixer_list;
		audio_mode_t mode;
		struct stream_in* active_input;
		struct stream_out* primary_output;
		struct stream_out* current_call_output;
		bool mic_mute;
		bool screen_off;

		@@ -393,6 +411,10 @@ struct audio_device {
		amplifier_device_t *amp;
		};

		/* From platform_info.c */
		int init_pcm_ids();
		int get_pcm_device_id(audio_usecase_t usecase, usecase_type_t type);

		/*
		* NOTE: when multiple mutexes have to be acquired, always take the
		* lock_inputs, stream_in, stream_out, then audio_device mutex.

audio/include/samsung_audio.h

+0 −3

Original line number	Diff line number	Diff line
		@@ -54,9 +54,6 @@
		#endif
		#endif

		#define MIXER_PATH_MAX_LENGTH 100
		#define MIXER_XML_PATH "mixer_paths_0.xml"

		/* Unusupported
		#define SOUND_CAPTURE_LOOPBACK_AEC_DEVICE 1
		#define SOUND_CAPTURE_HOTWORD_DEVICE 0

audio/mixer_util.c

0 → 100644

+174 −0

Original line number	Diff line number	Diff line
		/*
		* Copyright (C) 2018 The LineageOS Project
		*
		* Licensed under the Apache License, Version 2.0 (the "License");
		* you may not use this file except in compliance with the License.
		* You may obtain a copy of the License at
		*
		* http://www.apache.org/licenses/LICENSE-2.0
		*
		* Unless required by applicable law or agreed to in writing, software
		* distributed under the License is distributed on an "AS IS" BASIS,
		* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
		* See the License for the specific language governing permissions and
		* limitations under the License.
		*/

		#define LOG_TAG "mixer_util"
		#define LOG_NDEBUG 0

		#include <errno.h>
		#include <stdio.h>
		#include <expat.h>
		#include <cutils/log.h>
		#include <audio_hw.h>
		#include <math.h>
		#include <mixer_util.h>
		#include <samsung_audio.h>


		/* init with default ids, but updated while parsing mixer xml */
		int pcm_device_table[AUDIO_USECASE_MAX][2] = {
		[USECASE_AUDIO_PLAYBACK] = {SOUND_PLAYBACK_DEVICE, SOUND_PLAYBACK_DEVICE},
		[USECASE_AUDIO_PLAYBACK_OFFLOAD] = {-1, -1},
		[USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = {SOUND_DEEP_BUFFER_DEVICE, SOUND_DEEP_BUFFER_DEVICE},
		[USECASE_AUDIO_CAPTURE] = {SOUND_CAPTURE_DEVICE, SOUND_CAPTURE_DEVICE},
		[USECASE_BT_SCO] = {SOUND_PLAYBACK_SCO_DEVICE, SOUND_PLAYBACK_SCO_DEVICE},
		[USECASE_FM_RADIO] = {-1, -1},
		[USECASE_VOICE_CALL] = {SOUND_PLAYBACK_VOICE_DEVICE, SOUND_CAPTURE_VOICE_DEVICE},
		};

		static const char * const uc_pcmdai_table[AUDIO_USECASE_MAX] = {

		[USECASE_AUDIO_PLAYBACK] = "playback_link",
		[USECASE_AUDIO_PLAYBACK_MULTI_CH] = "none",
		[USECASE_AUDIO_PLAYBACK_OFFLOAD] = "playback_offload_link",
		[USECASE_AUDIO_PLAYBACK_DEEP_BUFFER] = "playback_deep_link",
		[USECASE_AUDIO_CAPTURE] = "capture_link",
		[USECASE_BT_SCO] = "bluetooth_link",
		[USECASE_FM_RADIO] = "fmradio_link",
		[USECASE_VOICE_CALL] = "baseband_link",
		};

		int get_pcm_device_id(audio_usecase_t usecase, usecase_type_t type)
		{
		if (type == PCM_PLAYBACK)
		return pcm_device_table[usecase][0];
		else
		return pcm_device_table[usecase][1];
		}

		static void process_pcmdai(const XML_Char **attr)
		{
		int i;

		for (i=0; i < AUDIO_USECASE_MAX; i++) {
		if (strcmp(attr[0], uc_pcmdai_table[i])==0) {
		ALOGV("%s: found %s = %d", __func__,uc_pcmdai_table[i],atoi((char *)attr[1]));
		pcm_device_table[i][0] = atoi((char *)attr[1]);
		pcm_device_table[i][1] = atoi((char *)attr[1]);
		}
		}

		/* extended capture links, usually equal with same playback links*/

		if (strcmp(attr[0], "baseband_capture_link")==0) {
		ALOGV("%s: found %s id =%d", __func__,"baseband_capture_link",atoi((char *)attr[1]));
		pcm_device_table[USECASE_VOICE_CALL][1] = atoi((char *)attr[1]);
		}

		if (strcmp(attr[0], "bluetooth_capture_link")==0) {
		ALOGV("%s: found %s id =%d", __func__,"bluetooth_capture_link",atoi((char *)attr[1]));
		pcm_device_table[USECASE_BT_SCO][1] = atoi((char *)attr[1]);
		}


		return;
		}


		static void start_tag(void userdata __unused, const XML_Char tag_name,
		const XML_Char **attr)
		{

		if (strcmp(tag_name, "pcmdai") == 0) {
		process_pcmdai(attr);
		}
		return;
		}

		static void end_tag(void userdata __unused, const XML_Char tag_name __unused)
		{
		}

		int init_pcm_ids(const char *filename)
		{
		XML_Parser parser;
		FILE *file;
		int ret = 0;
		int bytes_read;
		void *buf;
		static const uint32_t kBufSize = 1024;
		char mixer_file_name[MIXER_PATH_MAX_LENGTH]= {0};

		if (filename == NULL) {
		strlcpy(mixer_file_name, MIXER_PATH_DEFAULT, MIXER_PATH_MAX_LENGTH);
		} else {
		strlcpy(mixer_file_name, filename, MIXER_PATH_MAX_LENGTH);
		}

		ALOGV("%s: mixer file name is %s", __func__, mixer_file_name);

		file = fopen(mixer_file_name, "r");

		if (!file) {
		ALOGD("%s: Failed to open %s, using defaults.",
		__func__, mixer_file_name);
		ret = -ENODEV;
		goto done;
		}


		parser = XML_ParserCreate(NULL);
		if (!parser) {
		ALOGE("%s: Failed to create XML parser!", __func__);
		ret = -ENODEV;
		goto err_close_file;
		}

		XML_SetElementHandler(parser, start_tag, end_tag);

		while (1) {
		buf = XML_GetBuffer(parser, kBufSize);
		if (buf == NULL) {
		ALOGE("%s: XML_GetBuffer failed", __func__);
		ret = -ENOMEM;
		goto err_free_parser;
		}

		bytes_read = fread(buf, 1, kBufSize, file);
		if (bytes_read < 0) {
		ALOGE("%s: fread failed, bytes read = %d", __func__, bytes_read);
		ret = bytes_read;
		goto err_free_parser;
		}

		if (XML_ParseBuffer(parser, bytes_read,
		bytes_read == 0) == XML_STATUS_ERROR) {
		ALOGE("%s: XML_ParseBuffer failed, for %s",
		__func__, mixer_file_name);
		ret = -EINVAL;
		goto err_free_parser;
		}

		if (bytes_read == 0)
		break;
		}

		err_free_parser:
		XML_ParserFree(parser);
		err_close_file:
		fclose(file);
		done:
		return ret;
		}