From 5738f83aeb59361a0a2eda2460113f6dc9194271 Mon Sep 17 00:00:00 2001 From: The Android Open Source Project Date: Wed, 12 Dec 2012 16:00:35 -0800 Subject: Snapshot cdeccf6fdd8c2d494ea2867cb37a025bf8879baf Change-Id: Ia2de32ccb97a9641462c72363b0a8c4288f4f36d --- audio_a2dp_hw/audio_a2dp_hw.c | 1103 +++++++++++++++++++++++++++++++++++++++++ 1 file changed, 1103 insertions(+) create mode 100644 audio_a2dp_hw/audio_a2dp_hw.c (limited to 'audio_a2dp_hw/audio_a2dp_hw.c') diff --git a/audio_a2dp_hw/audio_a2dp_hw.c b/audio_a2dp_hw/audio_a2dp_hw.c new file mode 100644 index 0000000..42e416e --- /dev/null +++ b/audio_a2dp_hw/audio_a2dp_hw.c @@ -0,0 +1,1103 @@ +/****************************************************************************** + * + * 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 + * + *****************************************************************************/ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include +#include + +#include +#include "audio_a2dp_hw.h" + +#define LOG_TAG "audio_a2dp_hw" +/* #define LOG_NDEBUG 0 */ +#include + +/***************************************************************************** +** Constants & Macros +******************************************************************************/ + +#define CTRL_CHAN_RETRY_COUNT 3 +#define USEC_PER_SEC 1000000L + +#define CASE_RETURN_STR(const) case const: return #const; + +#define FNLOG() ALOGV("%s", __FUNCTION__); +#define DEBUG(fmt, ...) ALOGV("%s: " fmt,__FUNCTION__, ## __VA_ARGS__) +#define INFO(fmt, ...) ALOGI("%s: " fmt,__FUNCTION__, ## __VA_ARGS__) +#define ERROR(fmt, ...) ALOGE("%s: " fmt,__FUNCTION__, ## __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, + AUDIO_A2DP_STATE_SUSPENDED, /* need explicit set param call to resume (suspend=false) */ + AUDIO_A2DP_STATE_STANDBY /* allows write to autoresume */ +} a2dp_state_t; + +struct a2dp_stream_out; + +struct a2dp_audio_device { + struct audio_hw_device device; + struct a2dp_stream_out *output; +}; + +struct a2dp_config { + uint32_t rate; + uint32_t channel_flags; + int format; +}; + +/* move ctrl_fd outside output stream and keep open until HAL unloaded ? */ + +struct a2dp_stream_out { + struct audio_stream_out stream; + pthread_mutex_t lock; + int ctrl_fd; + int audio_fd; + size_t buffer_sz; + a2dp_state_t state; + struct a2dp_config cfg; +}; + +struct a2dp_stream_in { + struct audio_stream_in stream; +}; + +/***************************************************************************** +** Static variables +******************************************************************************/ + +/***************************************************************************** +** Static functions +******************************************************************************/ + +static size_t out_get_buffer_size(const struct audio_stream *stream); + +/***************************************************************************** +** Externs +******************************************************************************/ + +/***************************************************************************** +** Functions +******************************************************************************/ + +/***************************************************************************** +** Miscellaneous helper functions +******************************************************************************/ + +static const char* dump_a2dp_ctrl_event(char event) +{ + switch(event) + { + CASE_RETURN_STR(A2DP_CTRL_CMD_NONE) + CASE_RETURN_STR(A2DP_CTRL_CMD_CHECK_READY) + CASE_RETURN_STR(A2DP_CTRL_CMD_START) + CASE_RETURN_STR(A2DP_CTRL_CMD_STOP) + CASE_RETURN_STR(A2DP_CTRL_CMD_SUSPEND) + default: + return "UNKNOWN MSG ID"; + } +} + +/* logs timestamp with microsec precision + pprev is optional in case a dedicated diff is required */ +static void ts_log(char *tag, 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(struct a2dp_config cfg, int bytes) +{ + int chan_count = popcount(cfg.channel_flags); + + ASSERTC(cfg.format == AUDIO_FORMAT_PCM_16_BIT, + "unsupported sample sz", cfg.format); + + return bytes*(1000000/(chan_count*2))/cfg.rate; +} + +/***************************************************************************** +** +** bluedroid stack adaptation +** +*****************************************************************************/ + +static int skt_connect(struct a2dp_stream_out *out, char *path) +{ + int ret; + int skt_fd; + struct sockaddr_un remote; + int len; + + INFO("connect to %s (sz %d)", path, out->buffer_sz); + + skt_fd = socket(AF_LOCAL, SOCK_STREAM, 0); + + if(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 = out->buffer_sz; + ret = setsockopt(skt_fd, SOL_SOCKET, SO_SNDBUF, (char*)&len, (int)sizeof(len)); + + /* only issue warning if failed */ + if (ret < 0) + ERROR("setsockopt failed (%s)", strerror(errno)); + + INFO("connected to stack fd = %d", skt_fd); + + return skt_fd; +} + +static int skt_write(int fd, const void *p, size_t len) +{ + int sent; + struct pollfd pfd; + + FNLOG(); + + pfd.fd = fd; + pfd.events = POLLOUT; + + /* poll for 500 ms */ + + /* send time out */ + if (poll(&pfd, 1, 500) == 0) + return 0; + + ts_log("skt_write", len, NULL); + + if ((sent = send(fd, p, len, MSG_NOSIGNAL)) == -1) + { + ERROR("write failed with errno=%d\n", errno); + return -1; + } + + return sent; +} + +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_command(struct a2dp_stream_out *out, char cmd) +{ + char ack; + + DEBUG("A2DP COMMAND %s", dump_a2dp_ctrl_event(cmd)); + + /* send command */ + if (send(out->ctrl_fd, &cmd, 1, MSG_NOSIGNAL) == -1) + { + ERROR("cmd failed (%s)", strerror(errno)); + skt_disconnect(out->ctrl_fd); + out->ctrl_fd = AUDIO_SKT_DISCONNECTED; + return -1; + } + + /* wait for ack byte */ + if (recv(out->ctrl_fd, &ack, 1, MSG_NOSIGNAL) < 0) + { + ERROR("ack failed (%s)", strerror(errno)); + skt_disconnect(out->ctrl_fd); + out->ctrl_fd = AUDIO_SKT_DISCONNECTED; + return -1; + } + + DEBUG("A2DP COMMAND %s DONE STATUS %d", dump_a2dp_ctrl_event(cmd), ack); + + if (ack != A2DP_CTRL_ACK_SUCCESS) + return -1; + + return 0; +} + +/***************************************************************************** +** +** AUDIO DATA PATH +** +*****************************************************************************/ + +static void a2dp_stream_out_init(struct a2dp_stream_out *out) +{ + pthread_mutexattr_t lock_attr; + + FNLOG(); + + pthread_mutexattr_init(&lock_attr); + pthread_mutexattr_settype(&lock_attr, PTHREAD_MUTEX_RECURSIVE); + pthread_mutex_init(&out->lock, &lock_attr); + + out->ctrl_fd = AUDIO_SKT_DISCONNECTED; + out->audio_fd = AUDIO_SKT_DISCONNECTED; + out->state = AUDIO_A2DP_STATE_STOPPED; + + out->cfg.channel_flags = AUDIO_STREAM_DEFAULT_CHANNEL_FLAG; + out->cfg.format = AUDIO_STREAM_DEFAULT_FORMAT; + out->cfg.rate = AUDIO_STREAM_DEFAULT_RATE; + + /* manages max capacity of socket pipe */ + out->buffer_sz = AUDIO_STREAM_OUTPUT_BUFFER_SZ; +} + +static int start_audio_datapath(struct a2dp_stream_out *out) +{ + int oldstate = out->state; + + INFO("state %d", out->state); + + if (out->ctrl_fd == AUDIO_SKT_DISCONNECTED) + return -1; + + out->state = AUDIO_A2DP_STATE_STARTING; + + if (a2dp_command(out, A2DP_CTRL_CMD_START) < 0) + { + ERROR("audiopath start failed"); + + out->state = oldstate; + return -1; + } + + /* connect socket if not yet connected */ + if (out->audio_fd == AUDIO_SKT_DISCONNECTED) + { + out->audio_fd = skt_connect(out, A2DP_DATA_PATH); + + if (out->audio_fd < 0) + { + out->state = oldstate; + return -1; + } + + out->state = AUDIO_A2DP_STATE_STARTED; + } + + return 0; +} + + +static int stop_audio_datapath(struct a2dp_stream_out *out) +{ + int oldstate = out->state; + + INFO("state %d", out->state); + + if (out->ctrl_fd == AUDIO_SKT_DISCONNECTED) + return -1; + + /* prevent any stray output writes from autostarting the stream + while stopping audiopath */ + out->state = AUDIO_A2DP_STATE_STOPPING; + + if (a2dp_command(out, A2DP_CTRL_CMD_STOP) < 0) + { + ERROR("audiopath stop failed"); + out->state = oldstate; + return -1; + } + + out->state = AUDIO_A2DP_STATE_STOPPED; + + /* disconnect audio path */ + skt_disconnect(out->audio_fd); + out->audio_fd = AUDIO_SKT_DISCONNECTED; + + return 0; +} + +static int suspend_audio_datapath(struct a2dp_stream_out *out, bool standby) +{ + INFO("state %d", out->state); + + if (out->ctrl_fd == AUDIO_SKT_DISCONNECTED) + return -1; + + if (out->state == AUDIO_A2DP_STATE_STOPPING) + return -1; + + if (a2dp_command(out, A2DP_CTRL_CMD_SUSPEND) < 0) + return -1; + + if (standby) + out->state = AUDIO_A2DP_STATE_STANDBY; + else + out->state = AUDIO_A2DP_STATE_SUSPENDED; + + /* disconnect audio path */ + skt_disconnect(out->audio_fd); + + out->audio_fd = AUDIO_SKT_DISCONNECTED; + + return 0; +} + +static int check_a2dp_ready(struct a2dp_stream_out *out) +{ + INFO("state %d", out->state); + + if (a2dp_command(out, A2DP_CTRL_CMD_CHECK_READY) < 0) + { + ERROR("check a2dp ready failed"); + return -1; + } + 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; + + DEBUG("write %d bytes (fd %d)", bytes, out->audio_fd); + + if (out->state == AUDIO_A2DP_STATE_SUSPENDED) + { + DEBUG("stream suspended"); + return -1; + } + + /* only allow autostarting if we are in stopped or standby */ + if ((out->state == AUDIO_A2DP_STATE_STOPPED) || + (out->state == AUDIO_A2DP_STATE_STANDBY)) + { + pthread_mutex_lock(&out->lock); + + if (start_audio_datapath(out) < 0) + { + /* emulate time this write represents to avoid very fast write + failures during transition periods or remote suspend */ + + int us_delay = calc_audiotime(out->cfg, bytes); + + DEBUG("emulate a2dp write delay (%d us)", us_delay); + + usleep(us_delay); + pthread_mutex_unlock(&out->lock); + return -1; + } + + pthread_mutex_unlock(&out->lock); + } + else if (out->state != AUDIO_A2DP_STATE_STARTED) + { + ERROR("stream not in stopped or standby"); + return -1; + } + + sent = skt_write(out->audio_fd, buffer, bytes); + + if (sent == -1) + { + skt_disconnect(out->audio_fd); + out->audio_fd = AUDIO_SKT_DISCONNECTED; + out->state = AUDIO_A2DP_STATE_STOPPED; + } + + DEBUG("wrote %d bytes out of %d bytes", sent, bytes); + return sent; +} + + +static uint32_t out_get_sample_rate(const struct audio_stream *stream) +{ + struct a2dp_stream_out *out = (struct a2dp_stream_out *)stream; + + DEBUG("rate %d", out->cfg.rate); + + return out->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 : %d", rate); + + if (rate != AUDIO_STREAM_DEFAULT_RATE) + { + ERROR("only rate %d supported", AUDIO_STREAM_DEFAULT_RATE); + return -1; + } + + out->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; + + DEBUG("buffer_size : %d", out->buffer_sz); + + return out->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%x", out->cfg.channel_flags); + + return out->cfg.channel_flags; +} + +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->cfg.format); + return out->cfg.format; +} + +static int out_set_format(struct audio_stream *stream, audio_format_t format) +{ + struct a2dp_stream_out *out = (struct a2dp_stream_out *)stream; + 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; + + int retVal = 0; + + FNLOG(); + + pthread_mutex_lock(&out->lock); + + if (out->state == AUDIO_A2DP_STATE_STARTED) + retVal = suspend_audio_datapath(out, true); + else + retVal = 0; + pthread_mutex_unlock (&out->lock); + + return retVal; +} + +static int out_dump(const struct audio_stream *stream, int fd) +{ + struct a2dp_stream_out *out = (struct a2dp_stream_out *)stream; + 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; + struct str_parms *parms; + char keyval[16]; + int retval = 0; + + INFO("state %d", out->state); + + pthread_mutex_lock(&out->lock); + + parms = str_parms_create_str(kvpairs); + + /* dump params */ + str_parms_dump(parms); + + retval = str_parms_get_str(parms, "closing", keyval, sizeof(keyval)); + + if (retval >= 0) + { + if (strcmp(keyval, "true") == 0) + { + DEBUG("stream closing, disallow any writes"); + out->state = AUDIO_A2DP_STATE_STOPPING; + } + } + + retval = str_parms_get_str(parms, "A2dpSuspended", keyval, sizeof(keyval)); + + if (retval >= 0) + { + if (strcmp(keyval, "true") == 0) + { + if (out->state == AUDIO_A2DP_STATE_STARTED) + retval = suspend_audio_datapath(out, 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->state == AUDIO_A2DP_STATE_SUSPENDED) + out->state = AUDIO_A2DP_STATE_STANDBY; + /* Irrespective of the state, return 0 */ + retval = 0; + } + } + + pthread_mutex_unlock(&out->lock); + str_parms_destroy(parms); + + return retval; +} + +static char * out_get_parameters(const struct audio_stream *stream, const char *keys) +{ + struct a2dp_stream_out *out = (struct a2dp_stream_out *)stream; + + FNLOG(); + + /* add populating param here */ + + return strdup(""); +} + +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->buffer_sz * 1000 ) / + audio_stream_frame_size(&out->stream.common) / + out->cfg.rate) * 1000; + + + return (latency_us / 1000) + 200; +} + +static int out_set_volume(struct audio_stream_out *stream, float left, + float right) +{ + FNLOG(); + + /* volume controlled in audioflinger mixer (digital) */ + + return -ENOSYS; +} + + + +static int out_get_render_position(const struct audio_stream_out *stream, + uint32_t *dsp_frames) +{ + FNLOG(); + return -EINVAL; +} + +static int out_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) +{ + FNLOG(); + return 0; +} + +static int out_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) +{ + FNLOG(); + return 0; +} + +/* + * AUDIO INPUT STREAM + */ + +static uint32_t in_get_sample_rate(const struct audio_stream *stream) +{ + FNLOG(); + return 8000; +} + +static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate) +{ + FNLOG(); + return 0; +} + +static size_t in_get_buffer_size(const struct audio_stream *stream) +{ + FNLOG(); + return 320; +} + +static uint32_t in_get_channels(const struct audio_stream *stream) +{ + FNLOG(); + return AUDIO_CHANNEL_IN_MONO; +} + +static audio_format_t in_get_format(const struct audio_stream *stream) +{ + FNLOG(); + return AUDIO_FORMAT_PCM_16_BIT; +} + +static int in_set_format(struct audio_stream *stream, audio_format_t format) +{ + FNLOG(); + return 0; +} + +static int in_standby(struct audio_stream *stream) +{ + FNLOG(); + return 0; +} + +static int in_dump(const struct audio_stream *stream, int fd) +{ + FNLOG(); + return 0; +} + +static int in_set_parameters(struct audio_stream *stream, const char *kvpairs) +{ + FNLOG(); + return 0; +} + +static char * in_get_parameters(const struct audio_stream *stream, + const char *keys) +{ + FNLOG(); + return strdup(""); +} + +static int in_set_gain(struct audio_stream_in *stream, float gain) +{ + FNLOG(); + return 0; +} + +static ssize_t in_read(struct audio_stream_in *stream, void* buffer, + size_t bytes) +{ + FNLOG(); + return bytes; +} + +static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream) +{ + FNLOG(); + return 0; +} + +static int in_add_audio_effect(const struct audio_stream *stream, effect_handle_t effect) +{ + FNLOG(); + return 0; +} + +static int in_remove_audio_effect(const struct audio_stream *stream, effect_handle_t effect) +{ + FNLOG(); + + 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) + +{ + struct a2dp_audio_device *a2dp_dev = (struct a2dp_audio_device *)dev; + struct a2dp_stream_out *out; + int ret = 0; + int i; + + INFO("opening output"); + + 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; + + /* initialize a2dp specifics */ + a2dp_stream_out_init(out); + + /* set output config values */ + if (config) + { + 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); + } + *stream_out = &out->stream; + a2dp_dev->output = out; + + /* 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 ((out->ctrl_fd = skt_connect(out, A2DP_CTRL_PATH)) > 0) + { + /* success, now check if stack is ready */ + if (check_a2dp_ready(out) == 0) + break; + + ERROR("error : a2dp not ready, wait 250 ms and retry"); + usleep(250000); + skt_disconnect(out->ctrl_fd); + } + + /* ctrl channel not ready, wait a bit */ + usleep(250000); + } + + if (out->ctrl_fd == AUDIO_SKT_DISCONNECTED) + { + ERROR("ctrl socket failed to connect (%s)", strerror(errno)); + ret = -1; + goto err_open; + } + + DEBUG("success"); + return 0; + +err_open: + free(out); + *stream_out = 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; + + INFO("closing output (state %d)", out->state); + + if ((out->state == AUDIO_A2DP_STATE_STARTED) || (out->state == AUDIO_A2DP_STATE_STOPPING)) + stop_audio_datapath(out); + + skt_disconnect(out->ctrl_fd); + 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; + struct a2dp_stream_out *out = a2dp_dev->output; + int retval = 0; + + if (out == NULL) + return retval; + + INFO("state %d", out->state); + + retval = out->stream.common.set_parameters((struct audio_stream *)out, kvpairs); + + return retval; +} + +static char * adev_get_parameters(const struct audio_hw_device *dev, + const char *keys) +{ + struct str_parms *parms; + + FNLOG(); + + parms = str_parms_create_str(keys); + + str_parms_dump(parms); + + str_parms_destroy(parms); + + return strdup(""); +} + +static int adev_init_check(const struct audio_hw_device *dev) +{ + struct a2dp_audio_device *a2dp_dev = (struct a2dp_audio_device*)dev; + + FNLOG(); + + return 0; +} + +static int adev_set_voice_volume(struct audio_hw_device *dev, float volume) +{ + FNLOG(); + + return -ENOSYS; +} + +static int adev_set_master_volume(struct audio_hw_device *dev, float volume) +{ + FNLOG(); + + return -ENOSYS; +} + +static int adev_set_mode(struct audio_hw_device *dev, int mode) +{ + FNLOG(); + + return 0; +} + +static int adev_set_mic_mute(struct audio_hw_device *dev, bool state) +{ + FNLOG(); + + return -ENOSYS; +} + +static int adev_get_mic_mute(const struct audio_hw_device *dev, bool *state) +{ + FNLOG(); + + return -ENOSYS; +} + +static size_t adev_get_input_buffer_size(const struct audio_hw_device *dev, + const struct audio_config *config) +{ + FNLOG(); + + return 320; +} + +static int adev_open_input_stream(struct audio_hw_device *dev, + audio_io_handle_t handle, + audio_devices_t devices, + struct audio_config *config, + struct audio_stream_in **stream_in) +{ + struct a2dp_audio_device *ladev = (struct a2dp_audio_device *)dev; + struct a2dp_stream_in *in; + int ret; + + FNLOG(); + + 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; + + *stream_in = &in->stream; + return 0; + +err_open: + free(in); + *stream_in = NULL; + return ret; +} + +static void adev_close_input_stream(struct audio_hw_device *dev, + struct audio_stream_in *in) +{ + FNLOG(); + + return; +} + +static int adev_dump(const audio_hw_device_t *device, int fd) +{ + FNLOG(); + + return 0; +} + +static int adev_close(hw_device_t *device) +{ + FNLOG(); + + 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; + int ret; + + 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 = calloc(1, sizeof(struct a2dp_audio_device)); + + if (!adev) + return -ENOMEM; + + adev->device.common.tag = HARDWARE_DEVICE_TAG; + adev->device.common.version = AUDIO_DEVICE_API_VERSION_CURRENT; + 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, +}; + +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, + }, +}; + -- cgit v1.1