Google Git
Sign in
android / platform / system / bt / 1e24bc0 / . / audio_a2dp_hw / src / audio_a2dp_hw.cc
blob: 1fcc4084f94e221874047b87e456855dd648780d [file] [log] [blame]
/******************************************************************************
*
* Copyright (C) 2009-2012 Broadcom Corporation
*
* 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.
*
******************************************************************************/
/*****************************************************************************
*
* Filename: audio_a2dp_hw.c
*
* Description: Implements hal for bluedroid a2dp audio device
*
*****************************************************************************/
#define LOG_TAG "bt_a2dp_hw"
#include <errno.h>
#include <fcntl.h>
#include <inttypes.h>
#include <stdint.h>
#include <sys/errno.h>
#include <sys/socket.h>
#include <sys/stat.h>
#include <sys/time.h>
#include <sys/un.h>
#include <unistd.h>
#include <mutex>
#include <hardware/audio.h>
#include <hardware/hardware.h>
#include <system/audio.h>
#include "osi/include/hash_map_utils.h"
#include "osi/include/log.h"
#include "osi/include/osi.h"
#include "osi/include/socket_utils/sockets.h"
#include "audio_a2dp_hw.h"
/*****************************************************************************
* Constants & Macros
*****************************************************************************/
#define CTRL_CHAN_RETRY_COUNT 3
#define USEC_PER_SEC 1000000L
#define SOCK_SEND_TIMEOUT_MS 2000 /* Timeout for sending */
#define SOCK_RECV_TIMEOUT_MS 5000 /* Timeout for receiving */
// set WRITE_POLL_MS to 0 for blocking sockets, nonzero for polled non-blocking
// sockets
#define WRITE_POLL_MS 20
#define FNLOG() LOG_VERBOSE(LOG_TAG, "%s", __func__);
#define DEBUG(fmt, ...) \
LOG_VERBOSE(LOG_TAG, "%s: " fmt, __func__, ##__VA_ARGS__)
#define INFO(fmt, ...) LOG_INFO(LOG_TAG, "%s: " fmt, __func__, ##__VA_ARGS__)
#define WARN(fmt, ...) LOG_WARN(LOG_TAG, "%s: " fmt, __func__, ##__VA_ARGS__)
#define ERROR(fmt, ...) LOG_ERROR(LOG_TAG, "%s: " fmt, __func__, ##__VA_ARGS__)
#define ASSERTC(cond, msg, val) \
if (!(cond)) { \
ERROR("### ASSERT : %s line %d %s (%d) ###", __FILE__, __LINE__, msg, \
val); \
}
/*****************************************************************************
* Local type definitions
*****************************************************************************/
typedef enum {
AUDIO_A2DP_STATE_STARTING,
AUDIO_A2DP_STATE_STARTED,
AUDIO_A2DP_STATE_STOPPING,
AUDIO_A2DP_STATE_STOPPED,
/* need explicit set param call to resume (suspend=false) */
AUDIO_A2DP_STATE_SUSPENDED,
AUDIO_A2DP_STATE_STANDBY /* allows write to autoresume */
} a2dp_state_t;
struct a2dp_stream_in;
struct a2dp_stream_out;
struct a2dp_audio_device {
// Important: device must be first as an audio_hw_device* may be cast to
// a2dp_audio_device* when the type is implicitly known.
struct audio_hw_device device;
std::recursive_mutex* mutex; // See note below on mutex acquisition order.
struct a2dp_stream_in* input;
struct a2dp_stream_out* output;
};
struct a2dp_config {
uint32_t rate;
uint32_t channel_mask;
int format;
};
/* move ctrl_fd outside output stream and keep open until HAL unloaded ? */
struct a2dp_stream_common {
std::recursive_mutex* mutex; // See note below on mutex acquisition order.
int ctrl_fd;
int audio_fd;
size_t buffer_sz;
struct a2dp_config cfg;
a2dp_state_t state;
};
struct a2dp_stream_out {
struct audio_stream_out stream;
struct a2dp_stream_common common;
uint64_t frames_presented; // frames written, never reset
uint64_t frames_rendered; // frames written, reset on standby
};
struct a2dp_stream_in {
struct audio_stream_in stream;
struct a2dp_stream_common common;
};
/*
* Mutex acquisition order:
*
* The a2dp_audio_device (adev) mutex must be acquired before
* the a2dp_stream_common (out or in) mutex.
*
* This may differ from other audio HALs.
*/
/*****************************************************************************
* Static variables
*****************************************************************************/
/*****************************************************************************
* Static functions
*****************************************************************************/
static size_t out_get_buffer_size(const struct audio_stream* stream);
/*****************************************************************************
* Externs
*****************************************************************************/
/*****************************************************************************
* Functions
*****************************************************************************/
static void a2dp_open_ctrl_path(struct a2dp_stream_common* common);
/*****************************************************************************
* Miscellaneous helper functions
*****************************************************************************/
/* logs timestamp with microsec precision
pprev is optional in case a dedicated diff is required */
static void ts_log(UNUSED_ATTR const char* tag, UNUSED_ATTR int val,
struct timespec* pprev_opt) {
struct timespec now;
static struct timespec prev = {0, 0};
unsigned long long now_us;
unsigned long long diff_us;
clock_gettime(CLOCK_MONOTONIC, &now);
now_us = now.tv_sec * USEC_PER_SEC + now.tv_nsec / 1000;
if (pprev_opt) {
diff_us = (now.tv_sec - prev.tv_sec) * USEC_PER_SEC +
(now.tv_nsec - prev.tv_nsec) / 1000;
*pprev_opt = now;
DEBUG("[%s] ts %08lld, *diff %08lld, val %d", tag, now_us, diff_us, val);
} else {
diff_us = (now.tv_sec - prev.tv_sec) * USEC_PER_SEC +
(now.tv_nsec - prev.tv_nsec) / 1000;
prev = now;
DEBUG("[%s] ts %08lld, diff %08lld, val %d", tag, now_us, diff_us, val);
}
}
static int calc_audiotime_usec(struct a2dp_config cfg, int bytes) {
int chan_count = audio_channel_count_from_out_mask(cfg.channel_mask);
int bytes_per_sample;
switch (cfg.format) {
case AUDIO_FORMAT_PCM_8_BIT:
bytes_per_sample = 1;
break;
case AUDIO_FORMAT_PCM_16_BIT:
bytes_per_sample = 2;
break;
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
bytes_per_sample = 3;
break;
case AUDIO_FORMAT_PCM_8_24_BIT:
bytes_per_sample = 4;
break;
case AUDIO_FORMAT_PCM_32_BIT:
bytes_per_sample = 4;
break;
default:
ASSERTC(false, "unsupported sample format", cfg.format);
bytes_per_sample = 2;
break;
}
return (
int)(((int64_t)bytes * (USEC_PER_SEC / (chan_count * bytes_per_sample))) /
cfg.rate);
}
/*****************************************************************************
*
* bluedroid stack adaptation
*
****************************************************************************/
static int skt_connect(const char* path, size_t buffer_sz) {
int ret;
int skt_fd;
int len;
INFO("connect to %s (sz %zu)", path, buffer_sz);
skt_fd = socket(AF_LOCAL, SOCK_STREAM, 0);
if (osi_socket_local_client_connect(
skt_fd, path, ANDROID_SOCKET_NAMESPACE_ABSTRACT, SOCK_STREAM) < 0) {
ERROR("failed to connect (%s)", strerror(errno));
close(skt_fd);
return -1;
}
len = buffer_sz;
ret =
setsockopt(skt_fd, SOL_SOCKET, SO_SNDBUF, (char*)&len, (int)sizeof(len));
if (ret < 0) ERROR("setsockopt failed (%s)", strerror(errno));
ret =
setsockopt(skt_fd, SOL_SOCKET, SO_RCVBUF, (char*)&len, (int)sizeof(len));
if (ret < 0) ERROR("setsockopt failed (%s)", strerror(errno));
/* Socket send/receive timeout value */
struct timeval tv;
tv.tv_sec = SOCK_SEND_TIMEOUT_MS / 1000;
tv.tv_usec = (SOCK_SEND_TIMEOUT_MS % 1000) * 1000;
ret = setsockopt(skt_fd, SOL_SOCKET, SO_SNDTIMEO, &tv, sizeof(tv));
if (ret < 0) ERROR("setsockopt failed (%s)", strerror(errno));
tv.tv_sec = SOCK_RECV_TIMEOUT_MS / 1000;
tv.tv_usec = (SOCK_RECV_TIMEOUT_MS % 1000) * 1000;
ret = setsockopt(skt_fd, SOL_SOCKET, SO_RCVTIMEO, &tv, sizeof(tv));
if (ret < 0) ERROR("setsockopt failed (%s)", strerror(errno));
INFO("connected to stack fd = %d", skt_fd);
return skt_fd;
}
static int skt_read(int fd, void* p, size_t len) {
ssize_t read;
FNLOG();
ts_log("skt_read recv", len, NULL);
OSI_NO_INTR(read = recv(fd, p, len, MSG_NOSIGNAL));
if (read == -1) ERROR("read failed with errno=%d\n", errno);
return (int)read;
}
static int skt_write(int fd, const void* p, size_t len) {
ssize_t sent;
FNLOG();
ts_log("skt_write", len, NULL);
if (WRITE_POLL_MS == 0) {
// do not poll, use blocking send
OSI_NO_INTR(sent = send(fd, p, len, MSG_NOSIGNAL));
if (sent == -1) ERROR("write failed with error(%s)", strerror(errno));
return (int)sent;
}
// use non-blocking send, poll
int ms_timeout = SOCK_SEND_TIMEOUT_MS;
size_t count = 0;
while (count < len) {
OSI_NO_INTR(sent = send(fd, p, len - count, MSG_NOSIGNAL | MSG_DONTWAIT));
if (sent == -1) {
if (errno != EAGAIN && errno != EWOULDBLOCK) {
ERROR("write failed with error(%s)", strerror(errno));
return -1;
}
if (ms_timeout >= WRITE_POLL_MS) {
usleep(WRITE_POLL_MS * 1000);
ms_timeout -= WRITE_POLL_MS;
continue;
}
WARN("write timeout exceeded, sent %zu bytes", count);
return -1;
}
count += sent;
p = (const uint8_t*)p + sent;
}
return (int)count;
}
static int skt_disconnect(int fd) {
INFO("fd %d", fd);
if (fd != AUDIO_SKT_DISCONNECTED) {
shutdown(fd, SHUT_RDWR);
close(fd);
}
return 0;
}
/*****************************************************************************
*
* AUDIO CONTROL PATH
*
****************************************************************************/
static int a2dp_ctrl_receive(struct a2dp_stream_common* common, void* buffer,
size_t length) {
ssize_t ret;
int i;
for (i = 0;; i++) {
OSI_NO_INTR(ret = recv(common->ctrl_fd, buffer, length, MSG_NOSIGNAL));
if (ret > 0) {
break;
}
if (ret == 0) {
ERROR("receive control data failed: peer closed");
break;
}
if (errno != EWOULDBLOCK && errno != EAGAIN) {
ERROR("receive control data failed: error(%s)", strerror(errno));
break;
}
if (i == (CTRL_CHAN_RETRY_COUNT - 1)) {
ERROR("receive control data failed: max retry count");
break;
}
INFO("receive control data failed (%s), retrying", strerror(errno));
}
if (ret <= 0) {
skt_disconnect(common->ctrl_fd);
common->ctrl_fd = AUDIO_SKT_DISCONNECTED;
}
return ret;
}
// Sends control info for stream |common|. The data to send is stored in
// |buffer| and has size |length|.
// On success, returns the number of octets sent, otherwise -1.
static int a2dp_ctrl_send(struct a2dp_stream_common* common, const void* buffer,
size_t length) {
ssize_t sent;
size_t remaining = length;
int i;
if (length == 0) return 0; // Nothing to do
for (i = 0;; i++) {
OSI_NO_INTR(sent = send(common->ctrl_fd, buffer, remaining, MSG_NOSIGNAL));
if (sent == static_cast<ssize_t>(remaining)) {
remaining = 0;
break;
}
if (sent > 0) {
buffer = (static_cast<const char*>(buffer) + sent);
remaining -= sent;
continue;
}
if (sent < 0) {
if (errno != EWOULDBLOCK && errno != EAGAIN) {
ERROR("send control data failed: error(%s)", strerror(errno));
break;
}
INFO("send control data failed (%s), retrying", strerror(errno));
}
if (i >= (CTRL_CHAN_RETRY_COUNT - 1)) {
ERROR("send control data failed: max retry count");
break;
}
}
if (remaining > 0) {
skt_disconnect(common->ctrl_fd);
common->ctrl_fd = AUDIO_SKT_DISCONNECTED;
return -1;
}
return length;
}
static int a2dp_command(struct a2dp_stream_common* common, tA2DP_CTRL_CMD cmd) {
char ack;
DEBUG("A2DP COMMAND %s", audio_a2dp_hw_dump_ctrl_event(cmd));
if (common->ctrl_fd == AUDIO_SKT_DISCONNECTED) {
INFO("starting up or recovering from previous error");
a2dp_open_ctrl_path(common);
if (common->ctrl_fd == AUDIO_SKT_DISCONNECTED) {
ERROR("failure to open ctrl path");
return -1;
}
}
/* send command */
ssize_t sent;
OSI_NO_INTR(sent = send(common->ctrl_fd, &cmd, 1, MSG_NOSIGNAL));
if (sent == -1) {
ERROR("cmd failed (%s)", strerror(errno));
skt_disconnect(common->ctrl_fd);
common->ctrl_fd = AUDIO_SKT_DISCONNECTED;
return -1;
}
/* wait for ack byte */
if (a2dp_ctrl_receive(common, &ack, 1) < 0) {
ERROR("A2DP COMMAND %s: no ACK", audio_a2dp_hw_dump_ctrl_event(cmd));
return -1;
}
DEBUG("A2DP COMMAND %s DONE STATUS %d", audio_a2dp_hw_dump_ctrl_event(cmd),
ack);
if (ack == A2DP_CTRL_ACK_INCALL_FAILURE) return ack;
if (ack != A2DP_CTRL_ACK_SUCCESS) {
ERROR("A2DP COMMAND %s error %d", audio_a2dp_hw_dump_ctrl_event(cmd), ack);
return -1;
}
return 0;
}
static int check_a2dp_ready(struct a2dp_stream_common* common) {
if (a2dp_command(common, A2DP_CTRL_CMD_CHECK_READY) < 0) {
ERROR("check a2dp ready failed");
return -1;
}
return 0;
}
static int a2dp_read_input_audio_config(struct a2dp_stream_common* common) {
tA2DP_SAMPLE_RATE sample_rate;
tA2DP_CHANNEL_COUNT channel_count;
if (a2dp_command(common, A2DP_CTRL_GET_INPUT_AUDIO_CONFIG) < 0) {
ERROR("get a2dp input audio config failed");
return -1;
}
if (a2dp_ctrl_receive(common, &sample_rate, sizeof(tA2DP_SAMPLE_RATE)) < 0)
return -1;
if (a2dp_ctrl_receive(common, &channel_count, sizeof(tA2DP_CHANNEL_COUNT)) <
0) {
return -1;
}
switch (sample_rate) {
case 44100:
case 48000:
common->cfg.rate = sample_rate;
break;
default:
ERROR("Invalid sample rate: %" PRIu32, sample_rate);
return -1;
}
switch (channel_count) {
case 1:
common->cfg.channel_mask = AUDIO_CHANNEL_IN_MONO;
break;
case 2:
common->cfg.channel_mask = AUDIO_CHANNEL_IN_STEREO;
break;
default:
ERROR("Invalid channel count: %" PRIu32, channel_count);
return -1;
}
// TODO: For now input audio format is always hard-coded as PCM 16-bit
common->cfg.format = AUDIO_FORMAT_PCM_16_BIT;
INFO("got input audio config %d %d", common->cfg.format, common->cfg.rate);
return 0;
}
static int a2dp_read_output_audio_config(
struct a2dp_stream_common* common, btav_a2dp_codec_config_t* codec_config,
btav_a2dp_codec_config_t* codec_capability, bool update_stream_config) {
struct a2dp_config stream_config;
if (a2dp_command(common, A2DP_CTRL_GET_OUTPUT_AUDIO_CONFIG) < 0) {
ERROR("get a2dp output audio config failed");
return -1;
}
// Receive the current codec config
if (a2dp_ctrl_receive(common, &codec_config->sample_rate,
sizeof(btav_a2dp_codec_sample_rate_t)) < 0) {
return -1;
}
if (a2dp_ctrl_receive(common, &codec_config->bits_per_sample,
sizeof(btav_a2dp_codec_bits_per_sample_t)) < 0) {
return -1;
}
if (a2dp_ctrl_receive(common, &codec_config->channel_mode,
sizeof(btav_a2dp_codec_channel_mode_t)) < 0) {
return -1;
}
// Receive the current codec capability
if (a2dp_ctrl_receive(common, &codec_capability->sample_rate,
sizeof(btav_a2dp_codec_sample_rate_t)) < 0) {
return -1;
}
if (a2dp_ctrl_receive(common, &codec_capability->bits_per_sample,
sizeof(btav_a2dp_codec_bits_per_sample_t)) < 0) {
return -1;
}
if (a2dp_ctrl_receive(common, &codec_capability->channel_mode,
sizeof(btav_a2dp_codec_channel_mode_t)) < 0) {
return -1;
}
// Check the codec config sample rate
switch (codec_config->sample_rate) {
case BTAV_A2DP_CODEC_SAMPLE_RATE_44100:
stream_config.rate = 44100;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_48000:
stream_config.rate = 48000;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_88200:
stream_config.rate = 88200;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_96000:
stream_config.rate = 96000;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_176400:
stream_config.rate = 176400;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_192000:
stream_config.rate = 192000;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_NONE:
default:
ERROR("Invalid sample rate: 0x%x", codec_config->sample_rate);
return -1;
}
// Check the codec config bits per sample
switch (codec_config->bits_per_sample) {
case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_16:
stream_config.format = AUDIO_FORMAT_PCM_16_BIT;
break;
case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_24:
stream_config.format = AUDIO_FORMAT_PCM_24_BIT_PACKED;
break;
case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_32:
stream_config.format = AUDIO_FORMAT_PCM_32_BIT;
break;
case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_NONE:
default:
ERROR("Invalid bits per sample: 0x%x", codec_config->bits_per_sample);
return -1;
}
// Check the codec config channel mode
switch (codec_config->channel_mode) {
case BTAV_A2DP_CODEC_CHANNEL_MODE_MONO:
stream_config.channel_mask = AUDIO_CHANNEL_OUT_MONO;
break;
case BTAV_A2DP_CODEC_CHANNEL_MODE_STEREO:
stream_config.channel_mask = AUDIO_CHANNEL_OUT_STEREO;
break;
case BTAV_A2DP_CODEC_CHANNEL_MODE_NONE:
default:
ERROR("Invalid channel mode: 0x%x", codec_config->channel_mode);
return -1;
}
// Update the output stream configuration
if (update_stream_config) {
common->cfg.rate = stream_config.rate;
common->cfg.channel_mask = stream_config.channel_mask;
common->cfg.format = stream_config.format;
common->buffer_sz = audio_a2dp_hw_stream_compute_buffer_size(
codec_config->sample_rate, codec_config->bits_per_sample,
codec_config->channel_mode);
}
INFO(
"got output codec capability: sample_rate=0x%x bits_per_sample=0x%x "
"channel_mode=0x%x",
codec_capability->sample_rate, codec_capability->bits_per_sample,
codec_capability->channel_mode);
return 0;
}
static int a2dp_write_output_audio_config(struct a2dp_stream_common* common) {
btav_a2dp_codec_config_t codec_config;
if (a2dp_command(common, A2DP_CTRL_SET_OUTPUT_AUDIO_CONFIG) < 0) {
ERROR("set a2dp output audio config failed");
return -1;
}
codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_NONE;
codec_config.bits_per_sample = BTAV_A2DP_CODEC_BITS_PER_SAMPLE_NONE;
codec_config.channel_mode = BTAV_A2DP_CODEC_CHANNEL_MODE_NONE;
switch (common->cfg.rate) {
case 44100:
codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_44100;
break;
case 48000:
codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_48000;
break;
case 88200:
codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_88200;
break;
case 96000:
codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_96000;
break;
case 176400:
codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_176400;
break;
case 192000:
codec_config.sample_rate = BTAV_A2DP_CODEC_SAMPLE_RATE_192000;
break;
default:
ERROR("Invalid sample rate: %" PRIu32, common->cfg.rate);
return -1;
}
switch (common->cfg.format) {
case AUDIO_FORMAT_PCM_16_BIT:
codec_config.bits_per_sample = BTAV_A2DP_CODEC_BITS_PER_SAMPLE_16;
break;
case AUDIO_FORMAT_PCM_24_BIT_PACKED:
codec_config.bits_per_sample = BTAV_A2DP_CODEC_BITS_PER_SAMPLE_24;
break;
case AUDIO_FORMAT_PCM_32_BIT:
codec_config.bits_per_sample = BTAV_A2DP_CODEC_BITS_PER_SAMPLE_32;
break;
case AUDIO_FORMAT_PCM_8_24_BIT:
// FALLTHROUGH
// All 24-bit audio is expected in AUDIO_FORMAT_PCM_24_BIT_PACKED format
default:
ERROR("Invalid audio format: 0x%x", common->cfg.format);
return -1;
}
switch (common->cfg.channel_mask) {
case AUDIO_CHANNEL_OUT_MONO:
codec_config.channel_mode = BTAV_A2DP_CODEC_CHANNEL_MODE_MONO;
break;
case AUDIO_CHANNEL_OUT_STEREO:
codec_config.channel_mode = BTAV_A2DP_CODEC_CHANNEL_MODE_STEREO;
break;
default:
ERROR("Invalid channel mask: 0x%x", common->cfg.channel_mask);
return -1;
}
// Send the current codec config that has been selected by us
if (a2dp_ctrl_send(common, &codec_config.sample_rate,
sizeof(btav_a2dp_codec_sample_rate_t)) < 0)
return -1;
if (a2dp_ctrl_send(common, &codec_config.bits_per_sample,
sizeof(btav_a2dp_codec_bits_per_sample_t)) < 0) {
return -1;
}
if (a2dp_ctrl_send(common, &codec_config.channel_mode,
sizeof(btav_a2dp_codec_channel_mode_t)) < 0) {
return -1;
}
INFO(
"sent output codec config: sample_rate=0x%x bits_per_sample=0x%x "
"channel_mode=0x%x",
codec_config.sample_rate, codec_config.bits_per_sample,
codec_config.channel_mode);
return 0;
}
static void a2dp_open_ctrl_path(struct a2dp_stream_common* common) {
int i;
if (common->ctrl_fd != AUDIO_SKT_DISCONNECTED) return; // already connected
/* retry logic to catch any timing variations on control channel */
for (i = 0; i < CTRL_CHAN_RETRY_COUNT; i++) {
/* connect control channel if not already connected */
if ((common->ctrl_fd = skt_connect(
A2DP_CTRL_PATH, AUDIO_STREAM_CONTROL_OUTPUT_BUFFER_SZ)) >= 0) {
/* success, now check if stack is ready */
if (check_a2dp_ready(common) == 0) break;
ERROR("error : a2dp not ready, wait 250 ms and retry");
usleep(250000);
skt_disconnect(common->ctrl_fd);
common->ctrl_fd = AUDIO_SKT_DISCONNECTED;
}
/* ctrl channel not ready, wait a bit */
usleep(250000);
}
}
/*****************************************************************************
*
* AUDIO DATA PATH
*
****************************************************************************/
static void a2dp_stream_common_init(struct a2dp_stream_common* common) {
FNLOG();
common->mutex = new std::recursive_mutex;
common->ctrl_fd = AUDIO_SKT_DISCONNECTED;
common->audio_fd = AUDIO_SKT_DISCONNECTED;
common->state = AUDIO_A2DP_STATE_STOPPED;
/* manages max capacity of socket pipe */
common->buffer_sz = AUDIO_STREAM_OUTPUT_BUFFER_SZ;
}
static void a2dp_stream_common_destroy(struct a2dp_stream_common* common) {
FNLOG();
delete common->mutex;
common->mutex = NULL;
}
static int start_audio_datapath(struct a2dp_stream_common* common) {
INFO("state %d", common->state);
int oldstate = common->state;
common->state = AUDIO_A2DP_STATE_STARTING;
int a2dp_status = a2dp_command(common, A2DP_CTRL_CMD_START);
if (a2dp_status < 0) {
ERROR("Audiopath start failed (status %d)", a2dp_status);
goto error;
} else if (a2dp_status == A2DP_CTRL_ACK_INCALL_FAILURE) {
ERROR("Audiopath start failed - in call, move to suspended");
goto error;
}
/* connect socket if not yet connected */
if (common->audio_fd == AUDIO_SKT_DISCONNECTED) {
common->audio_fd = skt_connect(A2DP_DATA_PATH, common->buffer_sz);
if (common->audio_fd < 0) {
ERROR("Audiopath start failed - error opening data socket");
goto error;
}
}
common->state = (a2dp_state_t)AUDIO_A2DP_STATE_STARTED;
return 0;
error:
common->state = (a2dp_state_t)oldstate;
return -1;
}
static int stop_audio_datapath(struct a2dp_stream_common* common) {
int oldstate = common->state;
INFO("state %d", common->state);
/* prevent any stray output writes from autostarting the stream
while stopping audiopath */
common->state = AUDIO_A2DP_STATE_STOPPING;
if (a2dp_command(common, A2DP_CTRL_CMD_STOP) < 0) {
ERROR("audiopath stop failed");
common->state = (a2dp_state_t)oldstate;
return -1;
}
common->state = (a2dp_state_t)AUDIO_A2DP_STATE_STOPPED;
/* disconnect audio path */
skt_disconnect(common->audio_fd);
common->audio_fd = AUDIO_SKT_DISCONNECTED;
return 0;
}
static int suspend_audio_datapath(struct a2dp_stream_common* common,
bool standby) {
INFO("state %d", common->state);
if (common->state == AUDIO_A2DP_STATE_STOPPING) return -1;
if (a2dp_command(common, A2DP_CTRL_CMD_SUSPEND) < 0) return -1;
if (standby)
common->state = AUDIO_A2DP_STATE_STANDBY;
else
common->state = AUDIO_A2DP_STATE_SUSPENDED;
/* disconnect audio path */
skt_disconnect(common->audio_fd);
common->audio_fd = AUDIO_SKT_DISCONNECTED;
return 0;
}
/*****************************************************************************
*
* audio output callbacks
*
****************************************************************************/
static ssize_t out_write(struct audio_stream_out* stream, const void* buffer,
size_t bytes) {
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
int sent = -1;
DEBUG("write %zu bytes (fd %d)", bytes, out->common.audio_fd);
std::unique_lock<std::recursive_mutex> lock(*out->common.mutex);
if (out->common.state == AUDIO_A2DP_STATE_SUSPENDED ||
out->common.state == AUDIO_A2DP_STATE_STOPPING) {
DEBUG("stream suspended or closing");
goto finish;
}
/* only allow autostarting if we are in stopped or standby */
if ((out->common.state == AUDIO_A2DP_STATE_STOPPED) ||
(out->common.state == AUDIO_A2DP_STATE_STANDBY)) {
if (start_audio_datapath(&out->common) < 0) {
goto finish;
}
} else if (out->common.state != AUDIO_A2DP_STATE_STARTED) {
ERROR("stream not in stopped or standby");
goto finish;
}
lock.unlock();
sent = skt_write(out->common.audio_fd, buffer, bytes);
lock.lock();
if (sent == -1) {
skt_disconnect(out->common.audio_fd);
out->common.audio_fd = AUDIO_SKT_DISCONNECTED;
if ((out->common.state != AUDIO_A2DP_STATE_SUSPENDED) &&
(out->common.state != AUDIO_A2DP_STATE_STOPPING)) {
out->common.state = AUDIO_A2DP_STATE_STOPPED;
} else {
ERROR("write failed : stream suspended, avoid resetting state");
}
goto finish;
}
finish:;
const size_t frames = bytes / audio_stream_out_frame_size(stream);
out->frames_rendered += frames;
out->frames_presented += frames;
lock.unlock();
// If send didn't work out, sleep to emulate write delay.
if (sent == -1) {
const int us_delay = calc_audiotime_usec(out->common.cfg, bytes);
DEBUG("emulate a2dp write delay (%d us)", us_delay);
usleep(us_delay);
}
return bytes;
}
static uint32_t out_get_sample_rate(const struct audio_stream* stream) {
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
DEBUG("rate %" PRIu32, out->common.cfg.rate);
return out->common.cfg.rate;
}
static int out_set_sample_rate(struct audio_stream* stream, uint32_t rate) {
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
DEBUG("out_set_sample_rate : %" PRIu32, rate);
out->common.cfg.rate = rate;
return 0;
}
static size_t out_get_buffer_size(const struct audio_stream* stream) {
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
// period_size is the AudioFlinger mixer buffer size.
const size_t period_size =
out->common.buffer_sz / AUDIO_STREAM_OUTPUT_BUFFER_PERIODS;
DEBUG("socket buffer size: %zu period size: %zu", out->common.buffer_sz,
period_size);
return period_size;
}
size_t audio_a2dp_hw_stream_compute_buffer_size(
btav_a2dp_codec_sample_rate_t codec_sample_rate,
btav_a2dp_codec_bits_per_sample_t codec_bits_per_sample,
btav_a2dp_codec_channel_mode_t codec_channel_mode) {
size_t buffer_sz = AUDIO_STREAM_OUTPUT_BUFFER_SZ; // Default value
const uint64_t time_period_ms = 20; // Conservative 20ms
uint32_t sample_rate;
uint32_t bits_per_sample;
uint32_t number_of_channels;
// Check the codec config sample rate
switch (codec_sample_rate) {
case BTAV_A2DP_CODEC_SAMPLE_RATE_44100:
sample_rate = 44100;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_48000:
sample_rate = 48000;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_88200:
sample_rate = 88200;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_96000:
sample_rate = 96000;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_176400:
sample_rate = 176400;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_192000:
sample_rate = 192000;
break;
case BTAV_A2DP_CODEC_SAMPLE_RATE_NONE:
default:
ERROR("Invalid sample rate: 0x%x", codec_sample_rate);
return buffer_sz;
}
// Check the codec config bits per sample
switch (codec_bits_per_sample) {
case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_16:
bits_per_sample = 16;
break;
case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_24:
bits_per_sample = 24;
break;
case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_32:
bits_per_sample = 32;
break;
case BTAV_A2DP_CODEC_BITS_PER_SAMPLE_NONE:
default:
ERROR("Invalid bits per sample: 0x%x", codec_bits_per_sample);
return buffer_sz;
}
// Check the codec config channel mode
switch (codec_channel_mode) {
case BTAV_A2DP_CODEC_CHANNEL_MODE_MONO:
number_of_channels = 1;
break;
case BTAV_A2DP_CODEC_CHANNEL_MODE_STEREO:
number_of_channels = 2;
break;
case BTAV_A2DP_CODEC_CHANNEL_MODE_NONE:
default:
ERROR("Invalid channel mode: 0x%x", codec_channel_mode);
return buffer_sz;
}
//
// The buffer size is computed by using the following formula:
//
// AUDIO_STREAM_OUTPUT_BUFFER_SIZE =
// (TIME_PERIOD_MS * AUDIO_STREAM_OUTPUT_BUFFER_PERIODS *
// SAMPLE_RATE_HZ * NUMBER_OF_CHANNELS * (BITS_PER_SAMPLE / 8)) / 1000
//
// AUDIO_STREAM_OUTPUT_BUFFER_PERIODS controls how the socket buffer is
// divided for AudioFlinger data delivery. The AudioFlinger mixer delivers
// data in chunks of
// (AUDIO_STREAM_OUTPUT_BUFFER_SIZE / AUDIO_STREAM_OUTPUT_BUFFER_PERIODS) .
// If the number of periods is 2, the socket buffer represents "double
// buffering" of the AudioFlinger mixer buffer.
//
// Furthermore, the AudioFlinger expects the buffer size to be a multiple
// of 16 frames.
const size_t divisor = (AUDIO_STREAM_OUTPUT_BUFFER_PERIODS * 16 *
number_of_channels * bits_per_sample) /
8;
buffer_sz = (time_period_ms * AUDIO_STREAM_OUTPUT_BUFFER_PERIODS *
sample_rate * number_of_channels * (bits_per_sample / 8)) /
1000;
// Adjust the buffer size so it can be divided by the divisor
const size_t remainder = buffer_sz % divisor;
if (remainder != 0) {
buffer_sz += divisor - remainder;
}
return buffer_sz;
}
static uint32_t out_get_channels(const struct audio_stream* stream) {
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
DEBUG("channels 0x%" PRIx32, out->common.cfg.channel_mask);
return out->common.cfg.channel_mask;
}
static audio_format_t out_get_format(const struct audio_stream* stream) {
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
DEBUG("format 0x%x", out->common.cfg.format);
return (audio_format_t)out->common.cfg.format;
}
static int out_set_format(UNUSED_ATTR struct audio_stream* stream,
UNUSED_ATTR audio_format_t format) {
DEBUG("setting format not yet supported (0x%x)", format);
return -ENOSYS;
}
static int out_standby(struct audio_stream* stream) {
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
int retVal = 0;
FNLOG();
std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);
// Do nothing in SUSPENDED state.
if (out->common.state != AUDIO_A2DP_STATE_SUSPENDED)
retVal = suspend_audio_datapath(&out->common, true);
out->frames_rendered = 0; // rendered is reset, presented is not
return retVal;
}
static int out_dump(UNUSED_ATTR const struct audio_stream* stream,
UNUSED_ATTR int fd) {
FNLOG();
return 0;
}
static int out_set_parameters(struct audio_stream* stream,
const char* kvpairs) {
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
INFO("state %d kvpairs %s", out->common.state, kvpairs);
std::unordered_map<std::string, std::string> params =
hash_map_utils_new_from_string_params(kvpairs);
int status = 0;
if (params.empty()) return status;
std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);
/* dump params */
hash_map_utils_dump_string_keys_string_values(params);
if (params["closing"].compare("true") == 0) {
DEBUG("stream closing, disallow any writes");
out->common.state = AUDIO_A2DP_STATE_STOPPING;
}
if (params["A2dpSuspended"].compare("true") == 0) {
if (out->common.state == AUDIO_A2DP_STATE_STARTED)
status = suspend_audio_datapath(&out->common, false);
} else {
/* Do not start the streaming automatically. If the phone was streaming
* prior to being suspended, the next out_write shall trigger the
* AVDTP start procedure */
if (out->common.state == AUDIO_A2DP_STATE_SUSPENDED)
out->common.state = AUDIO_A2DP_STATE_STANDBY;
/* Irrespective of the state, return 0 */
}
return status;
}
static char* out_get_parameters(const struct audio_stream* stream,
const char* keys) {
FNLOG();
btav_a2dp_codec_config_t codec_config;
btav_a2dp_codec_config_t codec_capability;
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
std::unordered_map<std::string, std::string> params =
hash_map_utils_new_from_string_params(keys);
std::unordered_map<std::string, std::string> return_params;
if (params.empty()) return strdup("");
std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);
if (a2dp_read_output_audio_config(&out->common, &codec_config,
&codec_capability,
false /* update_stream_config */) < 0) {
ERROR("a2dp_read_output_audio_config failed");
goto done;
}
// Add the format
if (params.find(AUDIO_PARAMETER_STREAM_SUP_FORMATS) != params.end()) {
std::string param;
if (codec_capability.bits_per_sample & BTAV_A2DP_CODEC_BITS_PER_SAMPLE_16) {
if (!param.empty()) param += "|";
param += "AUDIO_FORMAT_PCM_16_BIT";
}
if (codec_capability.bits_per_sample & BTAV_A2DP_CODEC_BITS_PER_SAMPLE_24) {
if (!param.empty()) param += "|";
param += "AUDIO_FORMAT_PCM_24_BIT_PACKED";
}
if (codec_capability.bits_per_sample & BTAV_A2DP_CODEC_BITS_PER_SAMPLE_32) {
if (!param.empty()) param += "|";
param += "AUDIO_FORMAT_PCM_32_BIT";
}
if (param.empty()) {
ERROR("Invalid codec capability bits_per_sample=0x%x",
codec_capability.bits_per_sample);
goto done;
} else {
return_params[AUDIO_PARAMETER_STREAM_SUP_FORMATS] = param;
}
}
// Add the sample rate
if (params.find(AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES) != params.end()) {
std::string param;
if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_44100) {
if (!param.empty()) param += "|";
param += "44100";
}
if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_48000) {
if (!param.empty()) param += "|";
param += "48000";
}
if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_88200) {
if (!param.empty()) param += "|";
param += "88200";
}
if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_96000) {
if (!param.empty()) param += "|";
param += "96000";
}
if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_176400) {
if (!param.empty()) param += "|";
param += "176400";
}
if (codec_capability.sample_rate & BTAV_A2DP_CODEC_SAMPLE_RATE_192000) {
if (!param.empty()) param += "|";
param += "192000";
}
if (param.empty()) {
ERROR("Invalid codec capability sample_rate=0x%x",
codec_capability.sample_rate);
goto done;
} else {
return_params[AUDIO_PARAMETER_STREAM_SUP_SAMPLING_RATES] = param;
}
}
// Add the channel mask
if (params.find(AUDIO_PARAMETER_STREAM_SUP_CHANNELS) != params.end()) {
std::string param;
if (codec_capability.channel_mode & BTAV_A2DP_CODEC_CHANNEL_MODE_MONO) {
if (!param.empty()) param += "|";
param += "AUDIO_CHANNEL_OUT_MONO";
}
if (codec_capability.channel_mode & BTAV_A2DP_CODEC_CHANNEL_MODE_STEREO) {
if (!param.empty()) param += "|";
param += "AUDIO_CHANNEL_OUT_STEREO";
}
if (param.empty()) {
ERROR("Invalid codec capability channel_mode=0x%x",
codec_capability.channel_mode);
goto done;
} else {
return_params[AUDIO_PARAMETER_STREAM_SUP_CHANNELS] = param;
}
}
done:
std::string result;
for (const auto& ptr : return_params) {
result += ptr.first + "=" + ptr.second + ";";
}
INFO("get parameters result = %s", result.c_str());
return strdup(result.c_str());
}
static uint32_t out_get_latency(const struct audio_stream_out* stream) {
int latency_us;
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
FNLOG();
latency_us =
((out->common.buffer_sz * 1000) /
audio_stream_out_frame_size(&out->stream) / out->common.cfg.rate) *
1000;
return (latency_us / 1000) + 200;
}
static int out_set_volume(UNUSED_ATTR struct audio_stream_out* stream,
UNUSED_ATTR float left, UNUSED_ATTR float right) {
FNLOG();
/* volume controlled in audioflinger mixer (digital) */
return -ENOSYS;
}
static int out_get_presentation_position(const struct audio_stream_out* stream,
uint64_t* frames,
struct timespec* timestamp) {
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
FNLOG();
if (stream == NULL || frames == NULL || timestamp == NULL) return -EINVAL;
int ret = -EWOULDBLOCK;
std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);
uint64_t latency_frames =
(uint64_t)out_get_latency(stream) * out->common.cfg.rate / 1000;
if (out->frames_presented >= latency_frames) {
*frames = out->frames_presented - latency_frames;
clock_gettime(CLOCK_MONOTONIC,
timestamp); // could also be associated with out_write().
ret = 0;
}
return ret;
}
static int out_get_render_position(const struct audio_stream_out* stream,
uint32_t* dsp_frames) {
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
FNLOG();
if (stream == NULL || dsp_frames == NULL) return -EINVAL;
std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);
uint64_t latency_frames =
(uint64_t)out_get_latency(stream) * out->common.cfg.rate / 1000;
if (out->frames_rendered >= latency_frames) {
*dsp_frames = (uint32_t)(out->frames_rendered - latency_frames);
} else {
*dsp_frames = 0;
}
return 0;
}
static int out_add_audio_effect(UNUSED_ATTR const struct audio_stream* stream,
UNUSED_ATTR effect_handle_t effect) {
FNLOG();
return 0;
}
static int out_remove_audio_effect(
UNUSED_ATTR const struct audio_stream* stream,
UNUSED_ATTR effect_handle_t effect) {
FNLOG();
return 0;
}
/*
* AUDIO INPUT STREAM
*/
static uint32_t in_get_sample_rate(const struct audio_stream* stream) {
struct a2dp_stream_in* in = (struct a2dp_stream_in*)stream;
FNLOG();
return in->common.cfg.rate;
}
static int in_set_sample_rate(struct audio_stream* stream, uint32_t rate) {
struct a2dp_stream_in* in = (struct a2dp_stream_in*)stream;
FNLOG();
if (in->common.cfg.rate > 0 && in->common.cfg.rate == rate)
return 0;
else
return -1;
}
static size_t in_get_buffer_size(
UNUSED_ATTR const struct audio_stream* stream) {
FNLOG();
return 320;
}
static uint32_t in_get_channels(const struct audio_stream* stream) {
struct a2dp_stream_in* in = (struct a2dp_stream_in*)stream;
FNLOG();
return in->common.cfg.channel_mask;
}
static audio_format_t in_get_format(
UNUSED_ATTR const struct audio_stream* stream) {
FNLOG();
return AUDIO_FORMAT_PCM_16_BIT;
}
static int in_set_format(UNUSED_ATTR struct audio_stream* stream,
UNUSED_ATTR audio_format_t format) {
FNLOG();
if (format == AUDIO_FORMAT_PCM_16_BIT)
return 0;
else
return -1;
}
static int in_standby(UNUSED_ATTR struct audio_stream* stream) {
FNLOG();
return 0;
}
static int in_dump(UNUSED_ATTR const struct audio_stream* stream,
UNUSED_ATTR int fd) {
FNLOG();
return 0;
}
static int in_set_parameters(UNUSED_ATTR struct audio_stream* stream,
UNUSED_ATTR const char* kvpairs) {
FNLOG();
return 0;
}
static char* in_get_parameters(UNUSED_ATTR const struct audio_stream* stream,
UNUSED_ATTR const char* keys) {
FNLOG();
return strdup("");
}
static int in_set_gain(UNUSED_ATTR struct audio_stream_in* stream,
UNUSED_ATTR float gain) {
FNLOG();
return 0;
}
static ssize_t in_read(struct audio_stream_in* stream, void* buffer,
size_t bytes) {
struct a2dp_stream_in* in = (struct a2dp_stream_in*)stream;
int read;
int us_delay;
DEBUG("read %zu bytes, state: %d", bytes, in->common.state);
std::unique_lock<std::recursive_mutex> lock(*in->common.mutex);
if (in->common.state == AUDIO_A2DP_STATE_SUSPENDED ||
in->common.state == AUDIO_A2DP_STATE_STOPPING) {
DEBUG("stream suspended");
goto error;
}
/* only allow autostarting if we are in stopped or standby */
if ((in->common.state == AUDIO_A2DP_STATE_STOPPED) ||
(in->common.state == AUDIO_A2DP_STATE_STANDBY)) {
if (start_audio_datapath(&in->common) < 0) {
goto error;
}
} else if (in->common.state != AUDIO_A2DP_STATE_STARTED) {
ERROR("stream not in stopped or standby");
goto error;
}
lock.unlock();
read = skt_read(in->common.audio_fd, buffer, bytes);
lock.lock();
if (read == -1) {
skt_disconnect(in->common.audio_fd);
in->common.audio_fd = AUDIO_SKT_DISCONNECTED;
if ((in->common.state != AUDIO_A2DP_STATE_SUSPENDED) &&
(in->common.state != AUDIO_A2DP_STATE_STOPPING)) {
in->common.state = AUDIO_A2DP_STATE_STOPPED;
} else {
ERROR("read failed : stream suspended, avoid resetting state");
}
goto error;
} else if (read == 0) {
DEBUG("read time out - return zeros");
memset(buffer, 0, bytes);
read = bytes;
}
lock.unlock();
DEBUG("read %d bytes out of %zu bytes", read, bytes);
return read;
error:
memset(buffer, 0, bytes);
us_delay = calc_audiotime_usec(in->common.cfg, bytes);
DEBUG("emulate a2dp read delay (%d us)", us_delay);
usleep(us_delay);
return bytes;
}
static uint32_t in_get_input_frames_lost(
UNUSED_ATTR struct audio_stream_in* stream) {
FNLOG();
return 0;
}
static int in_add_audio_effect(UNUSED_ATTR const struct audio_stream* stream,
UNUSED_ATTR effect_handle_t effect) {
FNLOG();
return 0;
}
static int in_remove_audio_effect(UNUSED_ATTR const struct audio_stream* stream,
UNUSED_ATTR effect_handle_t effect) {
FNLOG();
return 0;
}
static int adev_open_output_stream(struct audio_hw_device* dev,
UNUSED_ATTR audio_io_handle_t handle,
UNUSED_ATTR audio_devices_t devices,
UNUSED_ATTR audio_output_flags_t flags,
struct audio_config* config,
struct audio_stream_out** stream_out,
UNUSED_ATTR const char* address)
{
struct a2dp_audio_device* a2dp_dev = (struct a2dp_audio_device*)dev;
struct a2dp_stream_out* out;
int ret = 0;
INFO("opening output");
// protect against adev->output and stream_out from being inconsistent
std::lock_guard<std::recursive_mutex> lock(*a2dp_dev->mutex);
out = (struct a2dp_stream_out*)calloc(1, sizeof(struct a2dp_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_presentation_position = out_get_presentation_position;
/* initialize a2dp specifics */
a2dp_stream_common_init(&out->common);
// Make sure we always have the feeding parameters configured
btav_a2dp_codec_config_t codec_config;
btav_a2dp_codec_config_t codec_capability;
if (a2dp_read_output_audio_config(&out->common, &codec_config,
&codec_capability,
true /* update_stream_config */) < 0) {
ERROR("a2dp_read_output_audio_config failed");
ret = -1;
goto err_open;
}
// a2dp_read_output_audio_config() opens the socket control path (or fails)
/* set output config values */
if (config != nullptr) {
// Try to use the config parameters and send it to the remote side
// TODO: Shall we use out_set_format() and similar?
if (config->format != 0) out->common.cfg.format = config->format;
if (config->sample_rate != 0) out->common.cfg.rate = config->sample_rate;
if (config->channel_mask != 0)
out->common.cfg.channel_mask = config->channel_mask;
if ((out->common.cfg.format != 0) || (out->common.cfg.rate != 0) ||
(out->common.cfg.channel_mask != 0)) {
if (a2dp_write_output_audio_config(&out->common) < 0) {
ERROR("a2dp_write_output_audio_config failed");
ret = -1;
goto err_open;
}
// Read again and make sure we use the same parameters as the remote side
if (a2dp_read_output_audio_config(&out->common, &codec_config,
&codec_capability,
true /* update_stream_config */) < 0) {
ERROR("a2dp_read_output_audio_config failed");
ret = -1;
goto err_open;
}
}
config->format = out_get_format((const struct audio_stream*)&out->stream);
config->sample_rate =
out_get_sample_rate((const struct audio_stream*)&out->stream);
config->channel_mask =
out_get_channels((const struct audio_stream*)&out->stream);
INFO(
"Output stream config: format=0x%x sample_rate=%d channel_mask=0x%x "
"buffer_sz=%zu",
config->format, config->sample_rate, config->channel_mask,
out->common.buffer_sz);
}
*stream_out = &out->stream;
a2dp_dev->output = out;
DEBUG("success");
/* Delay to ensure Headset is in proper state when START is initiated from
* DUT immediately after the connection due to ongoing music playback. */
usleep(250000);
return 0;
err_open:
a2dp_stream_common_destroy(&out->common);
free(out);
*stream_out = NULL;
a2dp_dev->output = NULL;
ERROR("failed");
return ret;
}
static void adev_close_output_stream(struct audio_hw_device* dev,
struct audio_stream_out* stream) {
struct a2dp_audio_device* a2dp_dev = (struct a2dp_audio_device*)dev;
struct a2dp_stream_out* out = (struct a2dp_stream_out*)stream;
// prevent interference with adev_set_parameters.
std::lock_guard<std::recursive_mutex> lock(*a2dp_dev->mutex);
{
std::lock_guard<std::recursive_mutex> lock(*out->common.mutex);
const a2dp_state_t state = out->common.state;
INFO("closing output (state %d)", (int)state);
if ((state == AUDIO_A2DP_STATE_STARTED) ||
(state == AUDIO_A2DP_STATE_STOPPING)) {
stop_audio_datapath(&out->common);
}
skt_disconnect(out->common.ctrl_fd);
out->common.ctrl_fd = AUDIO_SKT_DISCONNECTED;
}
a2dp_stream_common_destroy(&out->common);
free(stream);
a2dp_dev->output = NULL;
DEBUG("done");
}
static int adev_set_parameters(struct audio_hw_device* dev,
const char* kvpairs) {
struct a2dp_audio_device* a2dp_dev = (struct a2dp_audio_device*)dev;
int retval = 0;
// prevent interference with adev_close_output_stream
std::lock_guard<std::recursive_mutex> lock(*a2dp_dev->mutex);
struct a2dp_stream_out* out = a2dp_dev->output;
if (out == NULL) return retval;
INFO("state %d", out->common.state);
retval =
out->stream.common.set_parameters((struct audio_stream*)out, kvpairs);
return retval;
}
static char* adev_get_parameters(UNUSED_ATTR const struct audio_hw_device* dev,
const char* keys) {
FNLOG();
std::unordered_map<std::string, std::string> params =
hash_map_utils_new_from_string_params(keys);
hash_map_utils_dump_string_keys_string_values(params);
return strdup("");
}
static int adev_init_check(UNUSED_ATTR const struct audio_hw_device* dev) {
FNLOG();
return 0;
}
static int adev_set_voice_volume(UNUSED_ATTR struct audio_hw_device* dev,
UNUSED_ATTR float volume) {
FNLOG();
return -ENOSYS;
}
static int adev_set_master_volume(UNUSED_ATTR struct audio_hw_device* dev,
UNUSED_ATTR float volume) {
FNLOG();
return -ENOSYS;
}
static int adev_set_mode(UNUSED_ATTR struct audio_hw_device* dev,
UNUSED_ATTR audio_mode_t mode) {
FNLOG();
return 0;
}
static int adev_set_mic_mute(UNUSED_ATTR struct audio_hw_device* dev,
UNUSED_ATTR bool state) {
FNLOG();
return -ENOSYS;
}
static int adev_get_mic_mute(UNUSED_ATTR const struct audio_hw_device* dev,
UNUSED_ATTR bool* state) {
FNLOG();
return -ENOSYS;
}
static size_t adev_get_input_buffer_size(
UNUSED_ATTR const struct audio_hw_device* dev,
UNUSED_ATTR const struct audio_config* config) {
FNLOG();
return 320;
}
static int adev_open_input_stream(struct audio_hw_device* dev,
UNUSED_ATTR audio_io_handle_t handle,
UNUSED_ATTR audio_devices_t devices,
UNUSED_ATTR struct audio_config* config,
struct audio_stream_in** stream_in,
UNUSED_ATTR audio_input_flags_t flags,
UNUSED_ATTR const char* address,
UNUSED_ATTR audio_source_t source) {
struct a2dp_audio_device* a2dp_dev = (struct a2dp_audio_device*)dev;
struct a2dp_stream_in* in;
int ret;
FNLOG();
// protect against adev->input and stream_in from being inconsistent
std::lock_guard<std::recursive_mutex> lock(*a2dp_dev->mutex);
in = (struct a2dp_stream_in*)calloc(1, sizeof(struct a2dp_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;
/* initialize a2dp specifics */
a2dp_stream_common_init(&in->common);
*stream_in = &in->stream;
a2dp_dev->input = in;
if (a2dp_read_input_audio_config(&in->common) < 0) {
ERROR("a2dp_read_input_audio_config failed (%s)", strerror(errno));
ret = -1;
goto err_open;
}
// a2dp_read_input_audio_config() opens socket control path (or fails)
DEBUG("success");
return 0;
err_open:
a2dp_stream_common_destroy(&in->common);
free(in);
*stream_in = NULL;
a2dp_dev->input = NULL;
ERROR("failed");
return ret;
}
static void adev_close_input_stream(struct audio_hw_device* dev,
struct audio_stream_in* stream) {
struct a2dp_audio_device* a2dp_dev = (struct a2dp_audio_device*)dev;
struct a2dp_stream_in* in = (struct a2dp_stream_in*)stream;
std::lock_guard<std::recursive_mutex> lock(*a2dp_dev->mutex);
{
std::lock_guard<std::recursive_mutex> lock(*in->common.mutex);
const a2dp_state_t state = in->common.state;
INFO("closing input (state %d)", (int)state);
if ((state == AUDIO_A2DP_STATE_STARTED) ||
(state == AUDIO_A2DP_STATE_STOPPING))
stop_audio_datapath(&in->common);
skt_disconnect(in->common.ctrl_fd);
in->common.ctrl_fd = AUDIO_SKT_DISCONNECTED;
}
a2dp_stream_common_destroy(&in->common);
free(stream);
a2dp_dev->input = NULL;
DEBUG("done");
}
static int adev_dump(UNUSED_ATTR const audio_hw_device_t* device,
UNUSED_ATTR int fd) {
FNLOG();
return 0;
}
static int adev_close(hw_device_t* device) {
struct a2dp_audio_device* a2dp_dev = (struct a2dp_audio_device*)device;
FNLOG();
delete a2dp_dev->mutex;
a2dp_dev->mutex = nullptr;
free(device);
return 0;
}
static int adev_open(const hw_module_t* module, const char* name,
hw_device_t** device) {
struct a2dp_audio_device* adev;
INFO(" adev_open in A2dp_hw module");
FNLOG();
if (strcmp(name, AUDIO_HARDWARE_INTERFACE) != 0) {
ERROR("interface %s not matching [%s]", name, AUDIO_HARDWARE_INTERFACE);
return -EINVAL;
}
adev = (struct a2dp_audio_device*)calloc(1, sizeof(struct a2dp_audio_device));
if (!adev) return -ENOMEM;
adev->mutex = new std::recursive_mutex;
adev->device.common.tag = HARDWARE_DEVICE_TAG;
adev->device.common.version = AUDIO_DEVICE_API_VERSION_2_0;
adev->device.common.module = (struct hw_module_t*)module;
adev->device.common.close = adev_close;
adev->device.init_check = adev_init_check;
adev->device.set_voice_volume = adev_set_voice_volume;
adev->device.set_master_volume = adev_set_master_volume;
adev->device.set_mode = adev_set_mode;
adev->device.set_mic_mute = adev_set_mic_mute;
adev->device.get_mic_mute = adev_get_mic_mute;
adev->device.set_parameters = adev_set_parameters;
adev->device.get_parameters = adev_get_parameters;
adev->device.get_input_buffer_size = adev_get_input_buffer_size;
adev->device.open_output_stream = adev_open_output_stream;
adev->device.close_output_stream = adev_close_output_stream;
adev->device.open_input_stream = adev_open_input_stream;
adev->device.close_input_stream = adev_close_input_stream;
adev->device.dump = adev_dump;
adev->output = NULL;
*device = &adev->device.common;
return 0;
}
static struct hw_module_methods_t hal_module_methods = {
.open = adev_open,
};
__attribute__((
visibility("default"))) struct audio_module HAL_MODULE_INFO_SYM = {
.common =
{
.tag = HARDWARE_MODULE_TAG,
.version_major = 1,
.version_minor = 0,
.id = AUDIO_HARDWARE_MODULE_ID,
.name = "A2DP Audio HW HAL",
.author = "The Android Open Source Project",
.methods = &hal_module_methods,
},
};