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platform_hardware_libhardware/modules/usbaudio/audio_hal.c
Paul McLean 64345f8f2e Simplify and clarify channel count/mask calculation for stream open routines. 
Bug: 29154372
Change-Id: I849c02da0cc500e906cbda97b7258a3acf1e38f2
2016-06-13 13:59:59 -06:00

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/*
* Copyright (C) 2012 The Android Open Source 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 "modules.usbaudio.audio_hal"
/*#define LOG_NDEBUG 0*/
#include <errno.h>
#include <inttypes.h>
#include <pthread.h>
#include <stdint.h>
#include <stdlib.h>
#include <sys/time.h>
#include <log/log.h>
#include <cutils/list.h>
#include <cutils/str_parms.h>
#include <cutils/properties.h>
#include <hardware/audio.h>
#include <hardware/audio_alsaops.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include <tinyalsa/asoundlib.h>
#include <audio_utils/channels.h>
#include "alsa_device_profile.h"
#include "alsa_device_proxy.h"
#include "alsa_logging.h"
#define DEFAULT_INPUT_BUFFER_SIZE_MS 20
/* Lock play & record samples rates at or above this threshold */
#define RATELOCK_THRESHOLD 96000
struct audio_device {
struct audio_hw_device hw_device;
pthread_mutex_t lock; /* see note below on mutex acquisition order */
/* output */
alsa_device_profile out_profile;
struct listnode output_stream_list;
/* input */
alsa_device_profile in_profile;
struct listnode input_stream_list;
/* lock input & output sample rates */
/*FIXME - How do we address multiple output streams? */
uint32_t device_sample_rate;
bool mic_muted;
bool standby;
};
struct stream_lock {
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 */
};
struct stream_out {
struct audio_stream_out stream;
struct stream_lock lock;
bool standby;
struct audio_device *adev; /* hardware information - only using this for the lock */
alsa_device_profile * profile; /* Points to the alsa_device_profile in the audio_device */
alsa_device_proxy proxy; /* state of the stream */
unsigned hal_channel_count; /* channel count exposed to AudioFlinger.
* This may differ from the device channel count when
* the device is not compatible with AudioFlinger
* capabilities, e.g. exposes too many channels or
* too few channels. */
audio_channel_mask_t hal_channel_mask; /* USB devices deal in channel counts, not masks
* so the proxy doesn't have a channel_mask, but
* audio HALs need to talk about channel masks
* so expose the one calculated by
* adev_open_output_stream */
struct listnode list_node;
void * conversion_buffer; /* any conversions are put into here
* they could come from here too if
* there was a previous conversion */
size_t conversion_buffer_size; /* in bytes */
};
struct stream_in {
struct audio_stream_in stream;
struct stream_lock lock;
bool standby;
struct audio_device *adev; /* hardware information - only using this for the lock */
alsa_device_profile * profile; /* Points to the alsa_device_profile in the audio_device */
alsa_device_proxy proxy; /* state of the stream */
unsigned hal_channel_count; /* channel count exposed to AudioFlinger.
* This may differ from the device channel count when
* the device is not compatible with AudioFlinger
* capabilities, e.g. exposes too many channels or
* too few channels. */
audio_channel_mask_t hal_channel_mask; /* USB devices deal in channel counts, not masks
* so the proxy doesn't have a channel_mask, but
* audio HALs need to talk about channel masks
* so expose the one calculated by
* adev_open_input_stream */
struct listnode list_node;
/* We may need to read more data from the device in order to data reduce to 16bit, 4chan */
void * conversion_buffer; /* any conversions are put into here
* they could come from here too if
* there was a previous conversion */
size_t conversion_buffer_size; /* in bytes */
};
/*
* Locking Helpers
*/
/*
* NOTE: when multiple mutexes have to be acquired, always take the
* stream_in or stream_out mutex first, followed by the audio_device mutex.
* stream pre_lock is always acquired before stream lock to prevent starvation of control thread by
* higher priority playback or capture thread.
*/
static void stream_lock_init(struct stream_lock *lock) {
pthread_mutex_init(&lock->lock, (const pthread_mutexattr_t *) NULL);
pthread_mutex_init(&lock->pre_lock, (const pthread_mutexattr_t *) NULL);
}
static void stream_lock(struct stream_lock *lock) {
pthread_mutex_lock(&lock->pre_lock);
pthread_mutex_lock(&lock->lock);
pthread_mutex_unlock(&lock->pre_lock);
}
static void stream_unlock(struct stream_lock *lock) {
pthread_mutex_unlock(&lock->lock);
}
static void device_lock(struct audio_device *adev) {
pthread_mutex_lock(&adev->lock);
}
static int device_try_lock(struct audio_device *adev) {
return pthread_mutex_trylock(&adev->lock);
}
static void device_unlock(struct audio_device *adev) {
pthread_mutex_unlock(&adev->lock);
}
/*
* streams list management
*/
static void adev_add_stream_to_list(
struct audio_device* adev, struct listnode* list, struct listnode* stream_node) {
device_lock(adev);
list_add_tail(list, stream_node);
device_unlock(adev);
}
static void adev_remove_stream_from_list(
struct audio_device* adev, struct listnode* stream_node) {
device_lock(adev);
list_remove(stream_node);
device_unlock(adev);
}
/*
* Extract the card and device numbers from the supplied key/value pairs.
* kvpairs A null-terminated string containing the key/value pairs or card and device.
* i.e. "card=1;device=42"
* card A pointer to a variable to receive the parsed-out card number.
* device A pointer to a variable to receive the parsed-out device number.
* NOTE: The variables pointed to by card and device return -1 (undefined) if the
* associated key/value pair is not found in the provided string.
* Return true if the kvpairs string contain a card/device spec, false otherwise.
*/
static bool parse_card_device_params(const char *kvpairs, int *card, int *device)
{
struct str_parms * parms = str_parms_create_str(kvpairs);
char value[32];
int param_val;
// initialize to "undefined" state.
*card = -1;
*device = -1;
param_val = str_parms_get_str(parms, "card", value, sizeof(value));
if (param_val >= 0) {
*card = atoi(value);
}
param_val = str_parms_get_str(parms, "device", value, sizeof(value));
if (param_val >= 0) {
*device = atoi(value);
}
str_parms_destroy(parms);
return *card >= 0 && *device >= 0;
}
static char * device_get_parameters(alsa_device_profile * profile, const char * keys)
{
if (profile->card < 0 || profile->device < 0) {
return strdup("");
}
struct str_parms *query = str_parms_create_str(keys);
struct str_parms *result = str_parms_create();
/* These keys are from hardware/libhardware/include/audio.h */
/* supported sample rates */
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES)) {
char* rates_list = profile_get_sample_rate_strs(profile);
str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES,
rates_list);
free(rates_list);
}
/* supported channel counts */
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_CHANNELS)) {
char* channels_list = profile_get_channel_count_strs(profile);
str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_CHANNELS,
channels_list);
free(channels_list);
}
/* supported sample formats */
if (str_parms_has_key(query, AUDIO_PARAMETER_STREAM_SUP_FORMATS)) {
char * format_params = profile_get_format_strs(profile);
str_parms_add_str(result, AUDIO_PARAMETER_STREAM_SUP_FORMATS,
format_params);
free(format_params);
}
str_parms_destroy(query);
char* result_str = str_parms_to_str(result);
str_parms_destroy(result);
ALOGV("device_get_parameters = %s", result_str);
return result_str;
}
/*
* HAl Functions
*/
/**
* NOTE: when multiple mutexes have to be acquired, always respect the
* following order: hw device > out stream
*/
/*
* OUT functions
*/
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
uint32_t rate = proxy_get_sample_rate(&((struct stream_out*)stream)->proxy);
ALOGV("out_get_sample_rate() = %d", rate);
return rate;
}
static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
return 0;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
const struct stream_out* out = (const struct stream_out*)stream;
size_t buffer_size =
proxy_get_period_size(&out->proxy) * audio_stream_out_frame_size(&(out->stream));
return buffer_size;
}
static uint32_t out_get_channels(const struct audio_stream *stream)
{
const struct stream_out *out = (const struct stream_out*)stream;
return out->hal_channel_mask;
}
static audio_format_t out_get_format(const struct audio_stream *stream)
{
/* Note: The HAL doesn't do any FORMAT conversion at this time. It
* Relies on the framework to provide data in the specified format.
* This could change in the future.
*/
alsa_device_proxy * proxy = &((struct stream_out*)stream)->proxy;
audio_format_t format = audio_format_from_pcm_format(proxy_get_format(proxy));
return format;
}
static int out_set_format(struct audio_stream *stream, audio_format_t format)
{
return 0;
}
static int out_standby(struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
stream_lock(&out->lock);
if (!out->standby) {
device_lock(out->adev);
proxy_close(&out->proxy);
device_unlock(out->adev);
out->standby = true;
}
stream_unlock(&out->lock);
return 0;
}
static int out_dump(const struct audio_stream *stream, int fd) {
const struct stream_out* out_stream = (const struct stream_out*) stream;
if (out_stream != NULL) {
dprintf(fd, "Output Profile:\n");
profile_dump(out_stream->profile, fd);
dprintf(fd, "Output Proxy:\n");
proxy_dump(&out_stream->proxy, fd);
}
return 0;
}
static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
ALOGV("out_set_parameters() keys:%s", kvpairs);
struct stream_out *out = (struct stream_out *)stream;
int routing = 0;
int ret_value = 0;
int card = -1;
int device = -1;
if (!parse_card_device_params(kvpairs, &card, &device)) {
// nothing to do
return ret_value;
}
stream_lock(&out->lock);
/* Lock the device because that is where the profile lives */
device_lock(out->adev);
if (!profile_is_cached_for(out->profile, card, device)) {
/* cannot read pcm device info if playback is active */
if (!out->standby)
ret_value = -ENOSYS;
else {
int saved_card = out->profile->card;
int saved_device = out->profile->device;
out->profile->card = card;
out->profile->device = device;
ret_value = profile_read_device_info(out->profile) ? 0 : -EINVAL;
if (ret_value != 0) {
out->profile->card = saved_card;
out->profile->device = saved_device;
}
}
}
device_unlock(out->adev);
stream_unlock(&out->lock);
return ret_value;
}
static char * out_get_parameters(const struct audio_stream *stream, const char *keys)
{
struct stream_out *out = (struct stream_out *)stream;
stream_lock(&out->lock);
device_lock(out->adev);
char * params_str = device_get_parameters(out->profile, keys);
device_unlock(out->adev);
stream_unlock(&out->lock);
return params_str;
}
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
alsa_device_proxy * proxy = &((struct stream_out*)stream)->proxy;
return proxy_get_latency(proxy);
}
static int out_set_volume(struct audio_stream_out *stream, float left, float right)
{
return -ENOSYS;
}
/* must be called with hw device and output stream mutexes locked */
static int start_output_stream(struct stream_out *out)
{
ALOGV("start_output_stream(card:%d device:%d)", out->profile->card, out->profile->device);
return proxy_open(&out->proxy);
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer, size_t bytes)
{
int ret;
struct stream_out *out = (struct stream_out *)stream;
stream_lock(&out->lock);
if (out->standby) {
device_lock(out->adev);
ret = start_output_stream(out);
device_unlock(out->adev);
if (ret != 0) {
goto err;
}
out->standby = false;
}
alsa_device_proxy* proxy = &out->proxy;
const void * write_buff = buffer;
int num_write_buff_bytes = bytes;
const int num_device_channels = proxy_get_channel_count(proxy); /* what we told alsa */
const int num_req_channels = out->hal_channel_count; /* what we told AudioFlinger */
if (num_device_channels != num_req_channels) {
/* allocate buffer */
const size_t required_conversion_buffer_size =
bytes * num_device_channels / num_req_channels;
if (required_conversion_buffer_size > out->conversion_buffer_size) {
out->conversion_buffer_size = required_conversion_buffer_size;
out->conversion_buffer = realloc(out->conversion_buffer,
out->conversion_buffer_size);
}
/* convert data */
const audio_format_t audio_format = out_get_format(&(out->stream.common));
const unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
num_write_buff_bytes =
adjust_channels(write_buff, num_req_channels,
out->conversion_buffer, num_device_channels,
sample_size_in_bytes, num_write_buff_bytes);
write_buff = out->conversion_buffer;
}
if (write_buff != NULL && num_write_buff_bytes != 0) {
proxy_write(&out->proxy, write_buff, num_write_buff_bytes);
}
stream_unlock(&out->lock);
return bytes;
err:
stream_unlock(&out->lock);
if (ret != 0) {
usleep(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)
{
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; // discard const qualifier
stream_lock(&out->lock);
const alsa_device_proxy *proxy = &out->proxy;
const int ret = proxy_get_presentation_position(proxy, frames, timestamp);
stream_unlock(&out->lock);
return ret;
}
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 adev_open_output_stream(struct audio_hw_device *hw_dev,
audio_io_handle_t handle,
audio_devices_t devicesSpec __unused,
audio_output_flags_t flags,
struct audio_config *config,
struct audio_stream_out **stream_out,
const char *address /*__unused*/)
{
ALOGV("adev_open_output_stream() handle:0x%X, devicesSpec:0x%X, flags:0x%X, addr:%s",
handle, devicesSpec, flags, address);
struct stream_out *out;
out = (struct stream_out *)calloc(1, sizeof(struct stream_out));
if (out == NULL) {
return -ENOMEM;
}
/* setup function pointers */
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_presentation_position = out_get_presentation_position;
out->stream.get_next_write_timestamp = out_get_next_write_timestamp;
stream_lock_init(&out->lock);
out->adev = (struct audio_device *)hw_dev;
device_lock(out->adev);
out->profile = &out->adev->out_profile;
// build this to hand to the alsa_device_proxy
struct pcm_config proxy_config;
memset(&proxy_config, 0, sizeof(proxy_config));
/* Pull out the card/device pair */
parse_card_device_params(address, &(out->profile->card), &(out->profile->device));
profile_read_device_info(out->profile);
int ret = 0;
/* Rate */
if (config->sample_rate == 0) {
proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(out->profile);
} else if (profile_is_sample_rate_valid(out->profile, config->sample_rate)) {
proxy_config.rate = config->sample_rate;
} else {
proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(out->profile);
ret = -EINVAL;
}
out->adev->device_sample_rate = config->sample_rate;
device_unlock(out->adev);
/* Format */
if (config->format == AUDIO_FORMAT_DEFAULT) {
proxy_config.format = profile_get_default_format(out->profile);
config->format = audio_format_from_pcm_format(proxy_config.format);
} else {
enum pcm_format fmt = pcm_format_from_audio_format(config->format);
if (profile_is_format_valid(out->profile, fmt)) {
proxy_config.format = fmt;
} else {
proxy_config.format = profile_get_default_format(out->profile);
config->format = audio_format_from_pcm_format(proxy_config.format);
ret = -EINVAL;
}
}
/* Channels */
bool calc_mask = false;
if (config->channel_mask == AUDIO_CHANNEL_NONE) {
/* query case */
out->hal_channel_count = profile_get_default_channel_count(out->profile);
calc_mask = true;
} else {
/* explicit case */
out->hal_channel_count = audio_channel_count_from_out_mask(config->channel_mask);
}
/* The Framework is currently limited to no more than this number of channels */
if (out->hal_channel_count > FCC_8) {
out->hal_channel_count = FCC_8;
calc_mask = true;
}
if (calc_mask) {
/* need to calculate the mask from channel count either because this is the query case
* or the specified mask isn't valid for this device, or is more then the FW can handle */
config->channel_mask = out->hal_channel_count <= FCC_2
/* position mask for mono and stereo*/
? audio_channel_out_mask_from_count(out->hal_channel_count)
/* otherwise indexed */
: audio_channel_mask_for_index_assignment_from_count(out->hal_channel_count);
}
out->hal_channel_mask = config->channel_mask;
// Validate the "logical" channel count against support in the "actual" profile.
// if they differ, choose the "actual" number of channels *closest* to the "logical".
// and store THAT in proxy_config.channels
proxy_config.channels = profile_get_closest_channel_count(out->profile, out->hal_channel_count);
proxy_prepare(&out->proxy, out->profile, &proxy_config);
/* TODO The retry mechanism isn't implemented in AudioPolicyManager/AudioFlinger. */
ret = 0;
out->conversion_buffer = NULL;
out->conversion_buffer_size = 0;
out->standby = true;
/* Save the stream for adev_dump() */
adev_add_stream_to_list(out->adev, &out->adev->output_stream_list, &out->list_node);
*stream_out = &out->stream;
return ret;
err_open:
free(out);
*stream_out = NULL;
return -ENOSYS;
}
static void adev_close_output_stream(struct audio_hw_device *hw_dev,
struct audio_stream_out *stream)
{
struct stream_out *out = (struct stream_out *)stream;
ALOGV("adev_close_output_stream(c:%d d:%d)", out->profile->card, out->profile->device);
adev_remove_stream_from_list(out->adev, &out->list_node);
/* Close the pcm device */
out_standby(&stream->common);
free(out->conversion_buffer);
out->conversion_buffer = NULL;
out->conversion_buffer_size = 0;
device_lock(out->adev);
out->adev->device_sample_rate = 0;
device_unlock(out->adev);
free(stream);
}
static size_t adev_get_input_buffer_size(const struct audio_hw_device *hw_dev,
const struct audio_config *config)
{
/* TODO This needs to be calculated based on format/channels/rate */
return 320;
}
/*
* IN functions
*/
static uint32_t in_get_sample_rate(const struct audio_stream *stream)
{
uint32_t rate = proxy_get_sample_rate(&((const struct stream_in *)stream)->proxy);
ALOGV("in_get_sample_rate() = %d", rate);
return rate;
}
static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
{
ALOGV("in_set_sample_rate(%d) - NOPE", rate);
return -ENOSYS;
}
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
const struct stream_in * in = ((const struct stream_in*)stream);
return proxy_get_period_size(&in->proxy) * audio_stream_in_frame_size(&(in->stream));
}
static uint32_t in_get_channels(const struct audio_stream *stream)
{
const struct stream_in *in = (const struct stream_in*)stream;
return in->hal_channel_mask;
}
static audio_format_t in_get_format(const struct audio_stream *stream)
{
alsa_device_proxy *proxy = &((struct stream_in*)stream)->proxy;
audio_format_t format = audio_format_from_pcm_format(proxy_get_format(proxy));
return format;
}
static int in_set_format(struct audio_stream *stream, audio_format_t format)
{
ALOGV("in_set_format(%d) - NOPE", format);
return -ENOSYS;
}
static int in_standby(struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
stream_lock(&in->lock);
if (!in->standby) {
device_lock(in->adev);
proxy_close(&in->proxy);
device_unlock(in->adev);
in->standby = true;
}
stream_unlock(&in->lock);
return 0;
}
static int in_dump(const struct audio_stream *stream, int fd)
{
const struct stream_in* in_stream = (const struct stream_in*)stream;
if (in_stream != NULL) {
dprintf(fd, "Input Profile:\n");
profile_dump(in_stream->profile, fd);
dprintf(fd, "Input Proxy:\n");
proxy_dump(&in_stream->proxy, fd);
}
return 0;
}
static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
{
ALOGV("in_set_parameters() keys:%s", kvpairs);
struct stream_in *in = (struct stream_in *)stream;
char value[32];
int param_val;
int routing = 0;
int ret_value = 0;
int card = -1;
int device = -1;
if (!parse_card_device_params(kvpairs, &card, &device)) {
// nothing to do
return ret_value;
}
stream_lock(&in->lock);
device_lock(in->adev);
if (card >= 0 && device >= 0 && !profile_is_cached_for(in->profile, card, device)) {
/* cannot read pcm device info if playback is active */
if (!in->standby)
ret_value = -ENOSYS;
else {
int saved_card = in->profile->card;
int saved_device = in->profile->device;
in->profile->card = card;
in->profile->device = device;
ret_value = profile_read_device_info(in->profile) ? 0 : -EINVAL;
if (ret_value != 0) {
in->profile->card = saved_card;
in->profile->device = saved_device;
}
}
}
device_unlock(in->adev);
stream_unlock(&in->lock);
return ret_value;
}
static char * in_get_parameters(const struct audio_stream *stream, const char *keys)
{
struct stream_in *in = (struct stream_in *)stream;
stream_lock(&in->lock);
device_lock(in->adev);
char * params_str = device_get_parameters(in->profile, keys);
device_unlock(in->adev);
stream_unlock(&in->lock);
return params_str;
}
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 in_set_gain(struct audio_stream_in *stream, float gain)
{
return 0;
}
/* must be called with hw device and output stream mutexes locked */
static int start_input_stream(struct stream_in *in)
{
ALOGV("start_input_stream(card:%d device:%d)", in->profile->card, in->profile->device);
return proxy_open(&in->proxy);
}
/* TODO mutex stuff here (see out_write) */
static ssize_t in_read(struct audio_stream_in *stream, void* buffer, size_t bytes)
{
size_t num_read_buff_bytes = 0;
void * read_buff = buffer;
void * out_buff = buffer;
int ret = 0;
struct stream_in * in = (struct stream_in *)stream;
stream_lock(&in->lock);
if (in->standby) {
device_lock(in->adev);
ret = start_input_stream(in);
device_unlock(in->adev);
if (ret != 0) {
goto err;
}
in->standby = false;
}
alsa_device_profile * profile = in->profile;
/*
* OK, we need to figure out how much data to read to be able to output the requested
* number of bytes in the HAL format (16-bit, stereo).
*/
num_read_buff_bytes = bytes;
int num_device_channels = proxy_get_channel_count(&in->proxy); /* what we told Alsa */
int num_req_channels = in->hal_channel_count; /* what we told AudioFlinger */
if (num_device_channels != num_req_channels) {
num_read_buff_bytes = (num_device_channels * num_read_buff_bytes) / num_req_channels;
}
/* Setup/Realloc the conversion buffer (if necessary). */
if (num_read_buff_bytes != bytes) {
if (num_read_buff_bytes > in->conversion_buffer_size) {
/*TODO Remove this when AudioPolicyManger/AudioFlinger support arbitrary formats
(and do these conversions themselves) */
in->conversion_buffer_size = num_read_buff_bytes;
in->conversion_buffer = realloc(in->conversion_buffer, in->conversion_buffer_size);
}
read_buff = in->conversion_buffer;
}
ret = proxy_read(&in->proxy, read_buff, num_read_buff_bytes);
if (ret == 0) {
if (num_device_channels != num_req_channels) {
// ALOGV("chans dev:%d req:%d", num_device_channels, num_req_channels);
out_buff = buffer;
/* Num Channels conversion */
if (num_device_channels != num_req_channels) {
audio_format_t audio_format = in_get_format(&(in->stream.common));
unsigned sample_size_in_bytes = audio_bytes_per_sample(audio_format);
num_read_buff_bytes =
adjust_channels(read_buff, num_device_channels,
out_buff, num_req_channels,
sample_size_in_bytes, num_read_buff_bytes);
}
}
/* no need to acquire in->adev->lock to read mic_muted here as we don't change its state */
if (num_read_buff_bytes > 0 && in->adev->mic_muted)
memset(buffer, 0, num_read_buff_bytes);
} else {
num_read_buff_bytes = 0; // reset the value after USB headset is unplugged
}
err:
stream_unlock(&in->lock);
return num_read_buff_bytes;
}
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
{
return 0;
}
static int adev_open_input_stream(struct audio_hw_device *hw_dev,
audio_io_handle_t handle,
audio_devices_t devicesSpec __unused,
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)
{
ALOGV("adev_open_input_stream() rate:%" PRIu32 ", chanMask:0x%" PRIX32 ", fmt:%" PRIu8,
config->sample_rate, config->channel_mask, config->format);
struct stream_in *in = (struct stream_in *)calloc(1, sizeof(struct stream_in));
int ret = 0;
if (in == NULL) {
return -ENOMEM;
}
/* setup function pointers */
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_lock_init(&in->lock);
in->adev = (struct audio_device *)hw_dev;
device_lock(in->adev);
in->profile = &in->adev->in_profile;
struct pcm_config proxy_config;
memset(&proxy_config, 0, sizeof(proxy_config));
/* Pull out the card/device pair */
parse_card_device_params(address, &(in->profile->card), &(in->profile->device));
profile_read_device_info(in->profile);
/* Rate */
if (config->sample_rate == 0) {
config->sample_rate = profile_get_default_sample_rate(in->profile);
}
if (in->adev->device_sample_rate != 0 && /* we are playing, so lock the rate */
in->adev->device_sample_rate >= RATELOCK_THRESHOLD) {/* but only for high sample rates */
ret = config->sample_rate != in->adev->device_sample_rate ? -EINVAL : 0;
proxy_config.rate = config->sample_rate = in->adev->device_sample_rate;
} else if (profile_is_sample_rate_valid(in->profile, config->sample_rate)) {
in->adev->device_sample_rate = proxy_config.rate = config->sample_rate;
} else {
proxy_config.rate = config->sample_rate = profile_get_default_sample_rate(in->profile);
ret = -EINVAL;
}
device_unlock(in->adev);
/* Format */
if (config->format == AUDIO_FORMAT_DEFAULT) {
proxy_config.format = profile_get_default_format(in->profile);
config->format = audio_format_from_pcm_format(proxy_config.format);
} else {
enum pcm_format fmt = pcm_format_from_audio_format(config->format);
if (profile_is_format_valid(in->profile, fmt)) {
proxy_config.format = fmt;
} else {
proxy_config.format = profile_get_default_format(in->profile);
config->format = audio_format_from_pcm_format(proxy_config.format);
ret = -EINVAL;
}
}
/* Channels */
bool calc_mask = false;
if (config->channel_mask == AUDIO_CHANNEL_NONE) {
/* query case */
in->hal_channel_count = profile_get_default_channel_count(in->profile);
calc_mask = true;
} else {
/* explicit case */
in->hal_channel_count = audio_channel_count_from_in_mask(config->channel_mask);
}
/* The Framework is currently limited to no more than this number of channels */
if (in->hal_channel_count > FCC_8) {
in->hal_channel_count = FCC_8;
calc_mask = true;
}
if (calc_mask) {
/* need to calculate the mask from channel count either because this is the query case
* or the specified mask isn't valid for this device, or is more then the FW can handle */
in->hal_channel_mask = in->hal_channel_count <= FCC_2
/* position mask for mono & stereo */
? audio_channel_in_mask_from_count(in->hal_channel_count)
/* otherwise indexed */
: audio_channel_mask_for_index_assignment_from_count(in->hal_channel_count);
// if we change the mask...
if (in->hal_channel_mask != config->channel_mask &&
config->channel_mask != AUDIO_CHANNEL_NONE) {
config->channel_mask = in->hal_channel_mask;
ret = -EINVAL;
}
} else {
in->hal_channel_mask = config->channel_mask;
}
if (ret == 0) {
// Validate the "logical" channel count against support in the "actual" profile.
// if they differ, choose the "actual" number of channels *closest* to the "logical".
// and store THAT in proxy_config.channels
proxy_config.channels =
profile_get_closest_channel_count(in->profile, in->hal_channel_count);
proxy_prepare(&in->proxy, in->profile, &proxy_config);
in->standby = true;
in->conversion_buffer = NULL;
in->conversion_buffer_size = 0;
*stream_in = &in->stream;
/* Save this for adev_dump() */
adev_add_stream_to_list(in->adev, &in->adev->input_stream_list, &in->list_node);
} else {
// Deallocate this stream on error, because AudioFlinger won't call
// adev_close_input_stream() in this case.
*stream_in = NULL;
free(in);
}
return ret;
}
static void adev_close_input_stream(struct audio_hw_device *hw_dev,
struct audio_stream_in *stream)
{
struct stream_in *in = (struct stream_in *)stream;
ALOGV("adev_close_input_stream(c:%d d:%d)", in->profile->card, in->profile->device);
adev_remove_stream_from_list(in->adev, &in->list_node);
/* Close the pcm device */
in_standby(&stream->common);
free(in->conversion_buffer);
free(stream);
}
/*
* ADEV Functions
*/
static int adev_set_parameters(struct audio_hw_device *hw_dev, const char *kvpairs)
{
return 0;
}
static char * adev_get_parameters(const struct audio_hw_device *hw_dev, const char *keys)
{
return strdup("");
}
static int adev_init_check(const struct audio_hw_device *hw_dev)
{
return 0;
}
static int adev_set_voice_volume(struct audio_hw_device *hw_dev, float volume)
{
return -ENOSYS;
}
static int adev_set_master_volume(struct audio_hw_device *hw_dev, float volume)
{
return -ENOSYS;
}
static int adev_set_mode(struct audio_hw_device *hw_dev, audio_mode_t mode)
{
return 0;
}
static int adev_set_mic_mute(struct audio_hw_device *hw_dev, bool state)
{
struct audio_device * adev = (struct audio_device *)hw_dev;
device_lock(adev);
adev->mic_muted = state;
device_unlock(adev);
return -ENOSYS;
}
static int adev_get_mic_mute(const struct audio_hw_device *hw_dev, bool *state)
{
return -ENOSYS;
}
static int adev_dump(const struct audio_hw_device *device, int fd)
{
dprintf(fd, "\nUSB audio module:\n");
struct audio_device* adev = (struct audio_device*)device;
const int kNumRetries = 3;
const int kSleepTimeMS = 500;
// use device_try_lock() in case we dumpsys during a deadlock
int retry = kNumRetries;
while (retry > 0 && device_try_lock(adev) != 0) {
sleep(kSleepTimeMS);
retry--;
}
if (retry > 0) {
if (list_empty(&adev->output_stream_list)) {
dprintf(fd, " No output streams.\n");
} else {
struct listnode* node;
list_for_each(node, &adev->output_stream_list) {
struct audio_stream* stream =
(struct audio_stream *)node_to_item(node, struct stream_out, list_node);
out_dump(stream, fd);
}
}
if (list_empty(&adev->input_stream_list)) {
dprintf(fd, "\n No input streams.\n");
} else {
struct listnode* node;
list_for_each(node, &adev->input_stream_list) {
struct audio_stream* stream =
(struct audio_stream *)node_to_item(node, struct stream_in, list_node);
in_dump(stream, fd);
}
}
device_unlock(adev);
} else {
// Couldn't lock
dprintf(fd, " Could not obtain device lock.\n");
}
return 0;
}
static int adev_close(hw_device_t *device)
{
struct audio_device *adev = (struct audio_device *)device;
free(device);
return 0;
}
static int adev_open(const hw_module_t* module, const char* name, hw_device_t** device)
{
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0)
return -EINVAL;
struct audio_device *adev = calloc(1, sizeof(struct audio_device));
if (!adev)
return -ENOMEM;
profile_init(&adev->out_profile, PCM_OUT);
profile_init(&adev->in_profile, PCM_IN);
list_init(&adev->output_stream_list);
list_init(&adev->input_stream_list);
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.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;
*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 = "USB audio HW HAL",
.author = "The Android Open Source Project",
.methods = &hal_module_methods,
},
};
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