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audio: remove legacy hal

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This commit is contained in:
Konsta
2025-10-22 15:49:39 +03:00
parent be817e8c9e
commit 3292156c2a
3 changed files with 0 additions and 1566 deletions
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43
audio/Android.bp
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@@ -1,43 +0,0 @@
// Copyright (C) 2015 The Android Open Source Project
// Copyright (C) 2021-2023 KonstaKANG
//
// SPDX-License-Identifier: Apache-2.0
cc_library_shared {
name: "audio.primary.rpi",
relative_install_path: "hw",
vendor: true,
srcs: ["audio_hw.c"],
include_dirs: [
"external/expat/lib",
"external/tinyalsa/include",
"system/media/audio_effects/include",
"system/media/audio_utils/include",
],
header_libs: ["libhardware_headers"],
shared_libs: [
"libcutils",
"liblog",
"libtinyalsa",
],
cflags: ["-Wno-unused-parameter"],
}
cc_library_shared {
name: "audio.primary.rpi_hdmi",
relative_install_path: "hw",
vendor: true,
srcs: ["audio_hw_hdmi.c"],
include_dirs: [
"external/expat/lib",
"system/media/audio_effects/include",
"system/media/audio_utils/include",
],
header_libs: ["libhardware_headers"],
shared_libs: [
"libcutils",
"liblog",
"libasound",
],
cflags: ["-Wno-unused-parameter"],
}

750
audio/audio_hw.c
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@@ -1,750 +0,0 @@
/*
* Copyright (C) 2016 The Android Open Source Project
* Copyright (C) 2021-2022 KonstaKANG
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#define LOG_TAG "audio_hw_rpi"
//#define LOG_NDEBUG 0
#include <errno.h>
#include <malloc.h>
#include <pthread.h>
#include <stdint.h>
#include <sys/time.h>
#include <stdlib.h>
#include <unistd.h>
#include <log/log.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include <hardware/audio.h>
#include <sound/asound.h>
#include <tinyalsa/asoundlib.h>
#include <audio_utils/resampler.h>
#include <audio_utils/echo_reference.h>
#include <hardware/audio_effect.h>
#include <hardware/audio_alsaops.h>
#include <audio_effects/effect_aec.h>
/* Minimum granularity - Arbitrary but small value */
#define CODEC_BASE_FRAME_COUNT 32
/* number of base blocks in a short period (low latency) */
#define PERIOD_MULTIPLIER 32 /* 21 ms */
/* number of frames per short period (low latency) */
#define PERIOD_SIZE (CODEC_BASE_FRAME_COUNT * PERIOD_MULTIPLIER)
/* number of pseudo periods for low latency playback */
#define PLAYBACK_PERIOD_COUNT 4
#define PLAYBACK_PERIOD_START_THRESHOLD 2
#define CODEC_SAMPLING_RATE 48000
#define CHANNEL_STEREO 2
#define MIN_WRITE_SLEEP_US 5000
int pcm_card;
int pcm_device;
struct stub_stream_in {
struct audio_stream_in stream;
};
struct alsa_audio_device {
struct audio_hw_device hw_device;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
int devices;
struct alsa_stream_in *active_input;
struct alsa_stream_out *active_output;
bool mic_mute;
};
struct alsa_stream_out {
struct audio_stream_out stream;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
struct pcm_config config;
struct pcm *pcm;
bool unavailable;
int standby;
struct alsa_audio_device *dev;
int write_threshold;
unsigned int written;
};
static int probe_pcm_out_card() {
FILE *fp;
char card_node[32];
char card_id[16];
char card_prop[PROPERTY_VALUE_MAX];
property_get("persist.vendor.audio.device", card_prop, "");
for (int i = 0; i < 5; i++) {
snprintf(card_node, sizeof(card_node), "/proc/asound/card%d/id", i);
if ((fp = fopen(card_node, "r")) != NULL) {
fgets(card_id, sizeof(card_id), fp);
ALOGV("%s: %s", card_node, card_id);
if (!strcmp(card_prop, "jack") && !strncmp(card_id, "Headphones", 10)) {
fclose(fp);
ALOGI("Using PCM card %d for 3.5mm audio jack", i);
return i;
} else if (!strcmp(card_prop, "dac") && strncmp(card_id, "Headphones", 10)
&& strncmp(card_id, "vc4hdmi", 7)) {
fclose(fp);
ALOGI("Using PCM card %d for audio DAC %s", i, card_id);
return i;
}
fclose(fp);
}
}
ALOGE("Could not probe PCM card for %s, using PCM card 0", card_prop);
return 0;
}
static int get_pcm_card()
{
char card[PROPERTY_VALUE_MAX];
property_get("persist.vendor.audio.pcm.card.auto", card, "false");
if (!strcmp(card, "true"))
return probe_pcm_out_card();
property_get("persist.vendor.audio.pcm.card", card, "0");
return atoi(card);
}
static int get_pcm_device()
{
char device[PROPERTY_VALUE_MAX];
property_get("persist.vendor.audio.pcm.device", device, "0");
return atoi(device);
}
/* must be called with hw device and output stream mutexes locked */
static int start_output_stream(struct alsa_stream_out *out)
{
struct alsa_audio_device *adev = out->dev;
if (out->unavailable)
return -ENODEV;
/* default to low power: will be corrected in out_write if necessary before first write to
* tinyalsa.
*/
out->write_threshold = PLAYBACK_PERIOD_COUNT * PERIOD_SIZE;
out->config.start_threshold = PLAYBACK_PERIOD_START_THRESHOLD * PERIOD_SIZE;
out->config.avail_min = PERIOD_SIZE;
out->pcm = pcm_open(pcm_card, pcm_device, PCM_OUT | PCM_MMAP | PCM_NOIRQ | PCM_MONOTONIC, &out->config);
if (!pcm_is_ready(out->pcm)) {
ALOGE("cannot open pcm_out driver: %s", pcm_get_error(out->pcm));
pcm_close(out->pcm);
adev->active_output = NULL;
out->unavailable = true;
return -ENODEV;
}
adev->active_output = out;
return 0;
}
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
return out->config.rate;
}
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
ALOGV("out_set_sample_rate: %d", 0);
return -ENOSYS;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
ALOGV("out_get_buffer_size: %d", 4096);
/* return the closest majoring multiple of 16 frames, as
* audioflinger expects audio buffers to be a multiple of 16 frames */
size_t size = PERIOD_SIZE;
size = ((size + 15) / 16) * 16;
return size * audio_stream_out_frame_size((struct audio_stream_out *)stream);
}
static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
{
ALOGV("out_get_channels");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
return audio_channel_out_mask_from_count(out->config.channels);
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
ALOGV("out_get_format");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
return audio_format_from_pcm_format(out->config.format);
}
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
ALOGV("out_set_format: %d",format);
return -ENOSYS;
}
static int do_output_standby(struct alsa_stream_out *out)
{
struct alsa_audio_device *adev = out->dev;
if (!out->standby) {
pcm_close(out->pcm);
out->pcm = NULL;
adev->active_output = NULL;
out->standby = 1;
}
return 0;
}
static int out_standby(struct audio_stream *stream)
{
ALOGV("out_standby");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
int status;
pthread_mutex_lock(&out->dev->lock);
pthread_mutex_lock(&out->lock);
status = do_output_standby(out);
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&out->dev->lock);
return status;
}
static int out_dump(const struct audio_stream *stream, int fd)
{
ALOGV("out_dump");
return 0;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
ALOGV("out_set_parameters");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
struct alsa_audio_device *adev = out->dev;
struct str_parms *parms;
char value[32];
int val = 0;
int ret = -EINVAL;
if (kvpairs == NULL || kvpairs[0] == 0) {
return 0;
}
parms = str_parms_create_str(kvpairs);
if (str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)) >= 0) {
val = atoi(value);
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
if (((adev->devices & AUDIO_DEVICE_OUT_ALL) != val) && (val != 0)) {
adev->devices &= ~AUDIO_DEVICE_OUT_ALL;
adev->devices |= val;
}
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&adev->lock);
ret = 0;
}
str_parms_destroy(parms);
return ret;
}
static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
{
ALOGV("out_get_parameters");
return strdup("");
}
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
ALOGV("out_get_latency");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
return (PERIOD_SIZE * PLAYBACK_PERIOD_COUNT * 1000) / out->config.rate;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
ALOGV("out_set_volume: Left:%f Right:%f", left, right);
return 0;
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
int ret;
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
struct alsa_audio_device *adev = out->dev;
size_t frame_size = audio_stream_out_frame_size(stream);
size_t out_frames = bytes / frame_size;
/* acquiring hw device mutex systematically is useful if a low priority thread is waiting
* on the output stream mutex - e.g. executing select_mode() while holding the hw device
* mutex
*/
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
if (out->standby) {
ret = start_output_stream(out);
if (ret != 0) {
pthread_mutex_unlock(&adev->lock);
goto exit;
}
out->standby = 0;
}
pthread_mutex_unlock(&adev->lock);
ret = pcm_mmap_write(out->pcm, buffer, out_frames * frame_size);
if (ret == 0) {
out->written += out_frames;
}
exit:
pthread_mutex_unlock(&out->lock);
if (ret != 0) {
usleep((int64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
out_get_sample_rate(&stream->common));
}
return bytes;
}
static int out_get_render_position(const struct audio_stream_out *stream,
uint32_t *dsp_frames)
{
*dsp_frames = 0;
ALOGV("out_get_render_position: dsp_frames: %p", dsp_frames);
return -EINVAL;
}
static int out_get_presentation_position(const struct audio_stream_out *stream,
uint64_t *frames, struct timespec *timestamp)
{
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
int ret = -1;
if (out->pcm) {
unsigned int avail;
if (pcm_get_htimestamp(out->pcm, &avail, timestamp) == 0) {
size_t kernel_buffer_size = out->config.period_size * out->config.period_count;
int64_t signed_frames = out->written - kernel_buffer_size + avail;
if (signed_frames >= 0) {
*frames = signed_frames;
ret = 0;
}
}
}
return ret;
}
static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
ALOGV("out_add_audio_effect: %p", effect);
return 0;
}
static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
ALOGV("out_remove_audio_effect: %p", effect);
return 0;
}
static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
int64_t *timestamp)
{
*timestamp = 0;
ALOGV("out_get_next_write_timestamp: %ld", (long int)(*timestamp));
return -EINVAL;
}
/** audio_stream_in implementation **/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
ALOGV("in_get_sample_rate");
return 8000;
}
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
ALOGV("in_set_sample_rate: %d", rate);
return -ENOSYS;
}
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
ALOGV("in_get_buffer_size: %d", 320);
return 320;
}
static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
{
ALOGV("in_get_channels: %d", AUDIO_CHANNEL_IN_MONO);
return AUDIO_CHANNEL_IN_MONO;
}
static audio_format_t in_get_format(const struct audio_stream *stream)
{
return AUDIO_FORMAT_PCM_16_BIT;
}
static int in_set_format(struct audio_stream *stream, audio_format_t format)
{
return -ENOSYS;
}
static int in_standby(struct audio_stream *stream)
{
return 0;
}
static int in_dump(const struct audio_stream *stream, int fd)
{
return 0;
}
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
return 0;
}
static char * in_get_parameters(const struct audio_stream *stream,
const char *keys)
{
return strdup("");
}
static int in_set_gain(struct audio_stream_in *stream, float gain)
{
return 0;
}
static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
size_t bytes)
{
ALOGV("in_read: bytes %zu", bytes);
/* XXX: fake timing for audio input */
usleep((int64_t)bytes * 1000000 / audio_stream_in_frame_size(stream) /
in_get_sample_rate(&stream->common));
memset(buffer, 0, bytes);
return bytes;
}
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
{
return 0;
}
static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int adev_open_output_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
struct audio_config *config,
struct audio_stream_out **stream_out,
const char *address __unused)
{
ALOGV("adev_open_output_stream...");
struct alsa_audio_device *ladev = (struct alsa_audio_device *)dev;
struct alsa_stream_out *out;
struct pcm_params *params;
int ret = 0;
params = pcm_params_get(pcm_card, pcm_device, PCM_OUT);
if (!params)
return -ENOSYS;
out = (struct alsa_stream_out *)calloc(1, sizeof(struct alsa_stream_out));
if (!out)
return -ENOMEM;
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->config.channels = CHANNEL_STEREO;
out->config.rate = CODEC_SAMPLING_RATE;
out->config.format = PCM_FORMAT_S16_LE;
out->config.period_size = PERIOD_SIZE;
out->config.period_count = PLAYBACK_PERIOD_COUNT;
if (out->config.rate != config->sample_rate ||
audio_channel_count_from_out_mask(config->channel_mask) != CHANNEL_STEREO ||
out->config.format != pcm_format_from_audio_format(config->format) ) {
config->sample_rate = out->config.rate;
config->format = audio_format_from_pcm_format(out->config.format);
config->channel_mask = audio_channel_out_mask_from_count(CHANNEL_STEREO);
ret = -EINVAL;
}
ALOGI("adev_open_output_stream selects channels=%d rate=%d format=%d",
out->config.channels, out->config.rate, out->config.format);
out->dev = ladev;
out->standby = 1;
out->unavailable = false;
config->format = out_get_format(&out->stream.common);
config->channel_mask = out_get_channels(&out->stream.common);
config->sample_rate = out_get_sample_rate(&out->stream.common);
*stream_out = &out->stream;
/* TODO The retry mechanism isn't implemented in AudioPolicyManager/AudioFlinger. */
ret = 0;
return ret;
}
static void adev_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
ALOGV("adev_close_output_stream...");
free(stream);
}
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
ALOGV("adev_set_parameters");
return -ENOSYS;
}
static char * adev_get_parameters(const struct audio_hw_device *dev,
const char *keys)
{
ALOGV("adev_get_parameters");
return strdup("");
}
static int adev_init_check(const struct audio_hw_device *dev)
{
ALOGV("adev_init_check");
return 0;
}
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
ALOGV("adev_set_voice_volume: %f", volume);
return -ENOSYS;
}
static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
{
ALOGV("adev_set_master_volume: %f", volume);
return -ENOSYS;
}
static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
{
ALOGV("adev_get_master_volume: %f", *volume);
return -ENOSYS;
}
static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
{
ALOGV("adev_set_master_mute: %d", muted);
return -ENOSYS;
}
static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
{
ALOGV("adev_get_master_mute: %d", *muted);
return -ENOSYS;
}
static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
ALOGV("adev_set_mode: %d", mode);
return 0;
}
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
ALOGV("adev_set_mic_mute: %d",state);
return -ENOSYS;
}
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
ALOGV("adev_get_mic_mute");
return -ENOSYS;
}
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
const struct audio_config *config)
{
ALOGV("adev_get_input_buffer_size: %d", 320);
return 320;
}
static int adev_open_input_stream(struct audio_hw_device __unused *dev,
audio_io_handle_t handle,
audio_devices_t devices,
struct audio_config *config,
struct audio_stream_in **stream_in,
audio_input_flags_t flags __unused,
const char *address __unused,
audio_source_t source __unused)
{
struct stub_stream_in *in;
ALOGV("adev_open_input_stream...");
in = (struct stub_stream_in *)calloc(1, sizeof(struct stub_stream_in));
if (!in)
return -ENOMEM;
in->stream.common.get_sample_rate = in_get_sample_rate;
in->stream.common.set_sample_rate = in_set_sample_rate;
in->stream.common.get_buffer_size = in_get_buffer_size;
in->stream.common.get_channels = in_get_channels;
in->stream.common.get_format = in_get_format;
in->stream.common.set_format = in_set_format;
in->stream.common.standby = in_standby;
in->stream.common.dump = in_dump;
in->stream.common.set_parameters = in_set_parameters;
in->stream.common.get_parameters = in_get_parameters;
in->stream.common.add_audio_effect = in_add_audio_effect;
in->stream.common.remove_audio_effect = in_remove_audio_effect;
in->stream.set_gain = in_set_gain;
in->stream.read = in_read;
in->stream.get_input_frames_lost = in_get_input_frames_lost;
*stream_in = &in->stream;
return 0;
}
static void adev_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *in)
{
ALOGV("adev_close_input_stream...");
return;
}
static int adev_dump(const audio_hw_device_t *device, int fd)
{
ALOGV("adev_dump");
return 0;
}
static int adev_close(hw_device_t *device)
{
ALOGV("adev_close");
free(device);
return 0;
}
static int adev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
struct alsa_audio_device *adev;
ALOGV("adev_open: %s", name);
pcm_card = get_pcm_card();
pcm_device = get_pcm_device();
ALOGI("adev_open: pcm_card %d, pcm_device %d", pcm_card, pcm_device);
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
adev = calloc(1, sizeof(struct alsa_audio_device));
if (!adev)
return -ENOMEM;
adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
adev->hw_device.common.module = (struct hw_module_t *) module;
adev->hw_device.common.close = adev_close;
adev->hw_device.init_check = adev_init_check;
adev->hw_device.set_voice_volume = adev_set_voice_volume;
adev->hw_device.set_master_volume = adev_set_master_volume;
adev->hw_device.get_master_volume = adev_get_master_volume;
adev->hw_device.set_master_mute = adev_set_master_mute;
adev->hw_device.get_master_mute = adev_get_master_mute;
adev->hw_device.set_mode = adev_set_mode;
adev->hw_device.set_mic_mute = adev_set_mic_mute;
adev->hw_device.get_mic_mute = adev_get_mic_mute;
adev->hw_device.set_parameters = adev_set_parameters;
adev->hw_device.get_parameters = adev_get_parameters;
adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
adev->hw_device.open_output_stream = adev_open_output_stream;
adev->hw_device.close_output_stream = adev_close_output_stream;
adev->hw_device.open_input_stream = adev_open_input_stream;
adev->hw_device.close_input_stream = adev_close_input_stream;
adev->hw_device.dump = adev_dump;
adev->devices = AUDIO_DEVICE_NONE;
*device = &adev->hw_device.common;
return 0;
}
static struct hw_module_methods_t hal_module_methods = {
.open = adev_open,
};
struct audio_module HAL_MODULE_INFO_SYM = {
.common = {
.tag = HARDWARE_MODULE_TAG,
.module_api_version = AUDIO_MODULE_API_VERSION_0_1,
.hal_api_version = HARDWARE_HAL_API_VERSION,
.id = AUDIO_HARDWARE_MODULE_ID,
.name = "Raspberry Pi audio HW HAL",
.author = "The Android Open Source Project",
.methods = &hal_module_methods,
},
};

773
audio/audio_hw_hdmi.c
View File

@@ -1,773 +0,0 @@
/*
* Copyright (C) 2016 The Android Open Source Project
* Copyright (C) 2021-2023 KonstaKANG
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#define LOG_TAG "audio_hw_rpi_hdmi"
//#define LOG_NDEBUG 0
#include <errno.h>
#include <malloc.h>
#include <pthread.h>
#include <stdint.h>
#include <sys/time.h>
#include <stdlib.h>
#include <unistd.h>
#include <log/log.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include <hardware/audio.h>
#include <alsa/asoundlib.h>
#include <audio_utils/resampler.h>
#include <audio_utils/echo_reference.h>
#include <hardware/audio_effect.h>
#include <audio_effects/effect_aec.h>
/* Minimum granularity - Arbitrary but small value */
#define CODEC_BASE_FRAME_COUNT 32
/* number of base blocks in a short period (low latency) */
#define PERIOD_MULTIPLIER 32 /* 21 ms */
/* number of frames per short period (low latency) */
#define PERIOD_SIZE (CODEC_BASE_FRAME_COUNT * PERIOD_MULTIPLIER)
/* number of pseudo periods for low latency playback */
#define PLAYBACK_PERIOD_COUNT 4
#define PLAYBACK_PERIOD_START_THRESHOLD 2
#define CODEC_SAMPLING_RATE 48000
#define CHANNEL_STEREO 2
#define MIN_WRITE_SLEEP_US 5000
char device_name[PROPERTY_VALUE_MAX];
struct stub_stream_in {
struct audio_stream_in stream;
};
struct alsa_audio_device {
struct audio_hw_device hw_device;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
int devices;
struct alsa_stream_in *active_input;
struct alsa_stream_out *active_output;
bool mic_mute;
};
struct alsa_stream_out {
struct audio_stream_out stream;
struct alsa_audio_device *dev;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
snd_pcm_t *pcm;
snd_pcm_uframes_t period_size;
unsigned int periods;
snd_pcm_uframes_t buffer_size;
bool unavailable;
int standby;
snd_pcm_uframes_t written;
};
static void get_alsa_device_name(char *name) {
char hdmi_device[PROPERTY_VALUE_MAX];
property_get("persist.vendor.audio.hdmi.device", hdmi_device, "vc4hdmi0");
// use card configured in vc4-hdmi.conf to get IEC958 subframe conversion
sprintf(name, "default:CARD=%s", hdmi_device);
}
/* must be called with hw device and output stream mutexes locked */
static int start_output_stream(struct alsa_stream_out *out)
{
struct alsa_audio_device *adev = out->dev;
if (out->unavailable)
return -ENODEV;
ALOGI("start_output_stream: %s", device_name);
int r;
snd_pcm_t *pcm;
if ((r = snd_pcm_open(&pcm, device_name, SND_PCM_STREAM_PLAYBACK, 0) < 0)) {
ALOGE("cannot open pcm_out driver: %s", snd_strerror(r));
adev->active_output = NULL;
out->unavailable = true;
return -ENODEV;
}
out->pcm = pcm;
snd_pcm_hw_params_t *hwp;
snd_pcm_hw_params_alloca(&hwp);
snd_pcm_hw_params_any(pcm, hwp);
snd_pcm_hw_params_set_access(pcm, hwp, SND_PCM_ACCESS_RW_INTERLEAVED);
snd_pcm_hw_params_set_format(pcm, hwp, SND_PCM_FORMAT_S16_LE);
snd_pcm_hw_params_set_rate(pcm, hwp, CODEC_SAMPLING_RATE, 0);
snd_pcm_hw_params_set_channels(pcm, hwp, CHANNEL_STEREO);
// Configurue period_size, periods and buffer_size
int dir = 0;
out->period_size = PERIOD_SIZE;
if ((r = snd_pcm_hw_params_set_period_size_near(pcm, hwp, &out->period_size, &dir)) < 0) {
ALOGE("cannot snd_pcm_hw_params_set_period_size_near: %s", snd_strerror(r));
adev->active_output = NULL;
out->unavailable = true;
return -ENODEV;
}
dir = 0;
out->periods = PLAYBACK_PERIOD_COUNT;
if ((r = snd_pcm_hw_params_set_periods_near(pcm, hwp, &out->periods, &dir)) < 0) {
ALOGE("cannot snd_pcm_hw_params_set_periods_near: %s", snd_strerror(r));
adev->active_output = NULL;
out->unavailable = true;
return -ENODEV;
}
out->buffer_size = out->period_size * out->periods;
if ((r = snd_pcm_hw_params_set_buffer_size_near(pcm, hwp, &out->buffer_size)) < 0) {
ALOGE("cannot snd_pcm_hw_params_set_buffer_size_near: %s", snd_strerror(r));
adev->active_output = NULL;
out->unavailable = true;
return -ENODEV;
}
//write the hw params
if ((r = snd_pcm_hw_params(pcm, hwp)) < 0) {
ALOGE("cannot snd_pcm_hw_params: %s", snd_strerror(r));
adev->active_output = NULL;
out->unavailable = true;
return -ENODEV;
}
//Software parameters
snd_pcm_sw_params_t *swp;
snd_pcm_sw_params_alloca(&swp);
snd_pcm_sw_params_current(pcm, swp);
// set avail_min to period_size
if ((r = snd_pcm_sw_params_set_avail_min(pcm, swp, out->period_size)) < 0) {
ALOGE("cannot snd_pcm_sw_params_set_avail_min: %s", snd_strerror(r));
adev->active_output = NULL;
out->unavailable = true;
return -ENODEV;
}
// set start_threshold to period_size * PLAYBACK_PERIOD_START_THRESHOLD
if ((r = snd_pcm_sw_params_set_start_threshold(pcm, swp, out->period_size * PLAYBACK_PERIOD_START_THRESHOLD)) < 0) {
ALOGE("cannot snd_pcm_sw_params_set_start_threshold: %s", snd_strerror(r));
adev->active_output = NULL;
out->unavailable = true;
return -ENODEV;
}
//write the sw params
if ((r = snd_pcm_sw_params(pcm, swp)) < 0) {
ALOGE("cannot snd_pcm_sw_params: %s", snd_strerror(r));
adev->active_output = NULL;
out->unavailable = true;
return -ENODEV;
}
// prepare
if ((r = snd_pcm_prepare(pcm)) < 0) {
ALOGE("cannot snd_pcm_prepare: %s", snd_strerror(r));
adev->active_output = NULL;
out->unavailable = true;
return -ENODEV;
}
adev->active_output = out;
return 0;
}
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
ALOGV("out_get_sample_rate: %d", CODEC_SAMPLING_RATE);
return CODEC_SAMPLING_RATE;
}
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
ALOGV("out_set_sample_rate: %d", 0);
return -ENOSYS;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
/* return the closest majoring multiple of 16 frames, as
* audioflinger expects audio buffers to be a multiple of 16 frames */
size_t size = out->period_size;
size = ((size + 15) / 16) * 16;
ALOGV("out_get_buffer_size: %ld", (long int)size);
return size * audio_stream_out_frame_size((struct audio_stream_out *)stream);
}
static audio_channel_mask_t out_get_channels(const struct audio_stream *stream)
{
ALOGV("out_get_channels: %d", CHANNEL_STEREO);
return audio_channel_out_mask_from_count(CHANNEL_STEREO);
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
ALOGV("out_get_format: %d", AUDIO_FORMAT_PCM_16_BIT);
return AUDIO_FORMAT_PCM_16_BIT;
}
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
ALOGV("out_set_format: %d",format);
return -ENOSYS;
}
static int do_output_standby(struct alsa_stream_out *out)
{
struct alsa_audio_device *adev = out->dev;
if (!out->standby) {
snd_pcm_close(out->pcm);
out->pcm = NULL;
adev->active_output = NULL;
out->standby = 1;
}
return 0;
}
static int out_standby(struct audio_stream *stream)
{
ALOGV("out_standby");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
int status;
pthread_mutex_lock(&out->dev->lock);
pthread_mutex_lock(&out->lock);
status = do_output_standby(out);
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&out->dev->lock);
return status;
}
static int out_dump(const struct audio_stream *stream, int fd)
{
ALOGV("out_dump");
return 0;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
ALOGV("out_set_parameters");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
struct alsa_audio_device *adev = out->dev;
struct str_parms *parms;
char value[32];
int val = 0;
int ret = -EINVAL;
if (kvpairs == NULL || kvpairs[0] == 0) {
return 0;
}
parms = str_parms_create_str(kvpairs);
if (str_parms_get_str(parms, AUDIO_PARAMETER_STREAM_ROUTING, value, sizeof(value)) >= 0) {
val = atoi(value);
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
if (((adev->devices & AUDIO_DEVICE_OUT_ALL) != val) && (val != 0)) {
adev->devices &= ~AUDIO_DEVICE_OUT_ALL;
adev->devices |= val;
}
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&adev->lock);
ret = 0;
}
str_parms_destroy(parms);
return ret;
}
static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
{
ALOGV("out_get_parameters");
return strdup("");
}
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
ALOGV("out_get_latency");
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
// latency = buffer_size / rate
return (out->buffer_size * 1000) / CODEC_SAMPLING_RATE;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
float right)
{
ALOGV("out_set_volume: Left:%f Right:%f", left, right);
return 0;
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
int ret;
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
struct alsa_audio_device *adev = out->dev;
size_t frame_size = audio_stream_out_frame_size(stream);
snd_pcm_uframes_t out_frames = bytes / frame_size;
/* acquiring hw device mutex systematically is useful if a low priority thread is waiting
* on the output stream mutex - e.g. executing select_mode() while holding the hw device
* mutex
*/
pthread_mutex_lock(&adev->lock);
pthread_mutex_lock(&out->lock);
if (out->standby) {
ret = start_output_stream(out);
if (ret != 0) {
pthread_mutex_unlock(&adev->lock);
goto exit;
}
out->standby = 0;
}
pthread_mutex_unlock(&adev->lock);
ALOGV("out_write: out_frames:%ld", (long int)out_frames);
ret = snd_pcm_writei(out->pcm, buffer, out_frames);
if (ret == out_frames) {
out->written += out_frames;
}
exit:
pthread_mutex_unlock(&out->lock);
if (ret != out_frames) {
if (ret == -EPIPE) {
ALOGE("underrun deteced -> redo snd_pcm_prepare");
snd_pcm_prepare(out->pcm);
} else {
ALOGE("out_write err: %s", snd_strerror(ret));
usleep((int64_t)bytes * 1000000 / audio_stream_out_frame_size(stream) /
out_get_sample_rate(&stream->common));
}
}
return bytes;
}
static int out_get_render_position(const struct audio_stream_out *stream,
uint32_t *dsp_frames)
{
*dsp_frames = 0;
ALOGV("out_get_render_position: dsp_frames: %p", dsp_frames);
return -EINVAL;
}
static int out_get_presentation_position(const struct audio_stream_out *stream,
uint64_t *frames, struct timespec *timestamp)
{
struct alsa_stream_out *out = (struct alsa_stream_out *)stream;
int ret = -1;
if (out->pcm) {
snd_pcm_uframes_t avail;
int r;
if ((r = snd_pcm_htimestamp(out->pcm, &avail, timestamp)) == 0) {
int64_t signed_frames = (int64_t)(out->written) - out->buffer_size + avail;
if (signed_frames >= 0) {
*frames = signed_frames;
ret = 0;
}
ALOGV("out_get_presentation_position: %ld", (long int)(*frames));
} else {
ALOGE("out_get_presentation_position: err: %s", snd_strerror(r));
}
} else {
ALOGV("out_get_presentation_position: stream in standby");
}
return ret;
}
static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
ALOGV("out_add_audio_effect: %p", effect);
return 0;
}
static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
ALOGV("out_remove_audio_effect: %p", effect);
return 0;
}
static int out_get_next_write_timestamp(const struct audio_stream_out *stream,
int64_t *timestamp)
{
*timestamp = 0;
ALOGV("out_get_next_write_timestamp: %ld", (long int)(*timestamp));
return -EINVAL;
}
/** audio_stream_in implementation **/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
ALOGV("in_get_sample_rate");
return 8000;
}
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
ALOGV("in_set_sample_rate: %d", rate);
return -ENOSYS;
}
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
ALOGV("in_get_buffer_size: %d", 320);
return 320;
}
static audio_channel_mask_t in_get_channels(const struct audio_stream *stream)
{
ALOGV("in_get_channels: %d", AUDIO_CHANNEL_IN_MONO);
return AUDIO_CHANNEL_IN_MONO;
}
static audio_format_t in_get_format(const struct audio_stream *stream)
{
return AUDIO_FORMAT_PCM_16_BIT;
}
static int in_set_format(struct audio_stream *stream, audio_format_t format)
{
return -ENOSYS;
}
static int in_standby(struct audio_stream *stream)
{
return 0;
}
static int in_dump(const struct audio_stream *stream, int fd)
{
return 0;
}
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
return 0;
}
static char * in_get_parameters(const struct audio_stream *stream,
const char *keys)
{
return strdup("");
}
static int in_set_gain(struct audio_stream_in *stream, float gain)
{
return 0;
}
static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
size_t bytes)
{
ALOGV("in_read: bytes %zu", bytes);
/* XXX: fake timing for audio input */
usleep((int64_t)bytes * 1000000 / audio_stream_in_frame_size(stream) /
in_get_sample_rate(&stream->common));
memset(buffer, 0, bytes);
return bytes;
}
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
{
return 0;
}
static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect)
{
return 0;
}
static int adev_open_output_stream(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
struct audio_config *config,
struct audio_stream_out **stream_out,
const char *address __unused)
{
ALOGV("adev_open_output_stream...");
struct alsa_audio_device *ladev = (struct alsa_audio_device *)dev;
struct alsa_stream_out *out;
int ret = 0;
out = (struct alsa_stream_out *)calloc(1, sizeof(struct alsa_stream_out));
if (!out)
return -ENOMEM;
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->period_size = PERIOD_SIZE;
out->periods = PLAYBACK_PERIOD_COUNT;
out->buffer_size = out->period_size * out->periods;
out->dev = ladev;
out->standby = 1;
out->unavailable = false;
config->format = out_get_format(&out->stream.common);
config->channel_mask = out_get_channels(&out->stream.common);
config->sample_rate = out_get_sample_rate(&out->stream.common);
*stream_out = &out->stream;
/* TODO The retry mechanism isn't implemented in AudioPolicyManager/AudioFlinger. */
ret = 0;
return ret;
}
static void adev_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
ALOGV("adev_close_output_stream...");
free(stream);
}
static int adev_set_parameters(struct audio_hw_device *dev, const char *kvpairs)
{
ALOGV("adev_set_parameters");
return -ENOSYS;
}
static char * adev_get_parameters(const struct audio_hw_device *dev,
const char *keys)
{
ALOGV("adev_get_parameters");
return strdup("");
}
static int adev_init_check(const struct audio_hw_device *dev)
{
ALOGV("adev_init_check");
return 0;
}
static int adev_set_voice_volume(struct audio_hw_device *dev, float volume)
{
ALOGV("adev_set_voice_volume: %f", volume);
return -ENOSYS;
}
static int adev_set_master_volume(struct audio_hw_device *dev, float volume)
{
ALOGV("adev_set_master_volume: %f", volume);
return -ENOSYS;
}
static int adev_get_master_volume(struct audio_hw_device *dev, float *volume)
{
ALOGV("adev_get_master_volume: %f", *volume);
return -ENOSYS;
}
static int adev_set_master_mute(struct audio_hw_device *dev, bool muted)
{
ALOGV("adev_set_master_mute: %d", muted);
return -ENOSYS;
}
static int adev_get_master_mute(struct audio_hw_device *dev, bool *muted)
{
ALOGV("adev_get_master_mute: %d", *muted);
return -ENOSYS;
}
static int adev_set_mode(struct audio_hw_device *dev, audio_mode_t mode)
{
ALOGV("adev_set_mode: %d", mode);
return 0;
}
static int adev_set_mic_mute(struct audio_hw_device *dev, bool state)
{
ALOGV("adev_set_mic_mute: %d",state);
return -ENOSYS;
}
static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state)
{
ALOGV("adev_get_mic_mute");
return -ENOSYS;
}
static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev,
const struct audio_config *config)
{
ALOGV("adev_get_input_buffer_size: %d", 320);
return 320;
}
static int adev_open_input_stream(struct audio_hw_device __unused *dev,
audio_io_handle_t handle,
audio_devices_t devices,
struct audio_config *config,
struct audio_stream_in **stream_in,
audio_input_flags_t flags __unused,
const char *address __unused,
audio_source_t source __unused)
{
struct stub_stream_in *in;
ALOGV("adev_open_input_stream...");
in = (struct stub_stream_in *)calloc(1, sizeof(struct stub_stream_in));
if (!in)
return -ENOMEM;
in->stream.common.get_sample_rate = in_get_sample_rate;
in->stream.common.set_sample_rate = in_set_sample_rate;
in->stream.common.get_buffer_size = in_get_buffer_size;
in->stream.common.get_channels = in_get_channels;
in->stream.common.get_format = in_get_format;
in->stream.common.set_format = in_set_format;
in->stream.common.standby = in_standby;
in->stream.common.dump = in_dump;
in->stream.common.set_parameters = in_set_parameters;
in->stream.common.get_parameters = in_get_parameters;
in->stream.common.add_audio_effect = in_add_audio_effect;
in->stream.common.remove_audio_effect = in_remove_audio_effect;
in->stream.set_gain = in_set_gain;
in->stream.read = in_read;
in->stream.get_input_frames_lost = in_get_input_frames_lost;
*stream_in = &in->stream;
return 0;
}
static void adev_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *in)
{
ALOGV("adev_close_input_stream...");
return;
}
static int adev_dump(const audio_hw_device_t *device, int fd)
{
ALOGV("adev_dump");
return 0;
}
static int adev_close(hw_device_t *device)
{
ALOGV("adev_close");
free(device);
return 0;
}
static int adev_open(const hw_module_t* module, const char* name,
hw_device_t** device)
{
struct alsa_audio_device *adev;
ALOGV("adev_open: %s", name);
get_alsa_device_name(device_name);
ALOGI("adev_open: %s", device_name);
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
adev = calloc(1, sizeof(struct alsa_audio_device));
if (!adev)
return -ENOMEM;
adev->hw_device.common.tag = HARDWARE_DEVICE_TAG;
adev->hw_device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
adev->hw_device.common.module = (struct hw_module_t *) module;
adev->hw_device.common.close = adev_close;
adev->hw_device.init_check = adev_init_check;
adev->hw_device.set_voice_volume = adev_set_voice_volume;
adev->hw_device.set_master_volume = adev_set_master_volume;
adev->hw_device.get_master_volume = adev_get_master_volume;
adev->hw_device.set_master_mute = adev_set_master_mute;
adev->hw_device.get_master_mute = adev_get_master_mute;
adev->hw_device.set_mode = adev_set_mode;
adev->hw_device.set_mic_mute = adev_set_mic_mute;
adev->hw_device.get_mic_mute = adev_get_mic_mute;
adev->hw_device.set_parameters = adev_set_parameters;
adev->hw_device.get_parameters = adev_get_parameters;
adev->hw_device.get_input_buffer_size = adev_get_input_buffer_size;
adev->hw_device.open_output_stream = adev_open_output_stream;
adev->hw_device.close_output_stream = adev_close_output_stream;
adev->hw_device.open_input_stream = adev_open_input_stream;
adev->hw_device.close_input_stream = adev_close_input_stream;
adev->hw_device.dump = adev_dump;
adev->devices = AUDIO_DEVICE_NONE;
*device = &adev->hw_device.common;
return 0;
}
static struct hw_module_methods_t hal_module_methods = {
.open = adev_open,
};
struct audio_module HAL_MODULE_INFO_SYM = {
.common = {
.tag = HARDWARE_MODULE_TAG,
.module_api_version = AUDIO_MODULE_API_VERSION_0_1,
.hal_api_version = HARDWARE_HAL_API_VERSION,
.id = AUDIO_HARDWARE_MODULE_ID,
.name = "Raspberry Pi audio hdmi HW HAL",
.author = "The Android Open Source Project",
.methods = &hal_module_methods,
},
};