2 * Copyright (C) 2013 Intel Corporation
4 * Licensed under the Apache License, Version 2.0 (the "License");
5 * you may not use this file except in compliance with the License.
6 * You may obtain a copy of the License at
8 * http://www.apache.org/licenses/LICENSE-2.0
10 * Unless required by applicable law or agreed to in writing, software
11 * distributed under the License is distributed on an "AS IS" BASIS,
12 * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
13 * See the License for the specific language governing permissions and
14 * limitations under the License.
24 #include <sys/socket.h>
27 #include <arpa/inet.h>
30 #include <hardware/audio.h>
31 #include <hardware/hardware.h>
35 #include "audio-msg.h"
38 #include "../profiles/audio/a2dp-codecs.h"
40 #define FIXED_A2DP_PLAYBACK_LATENCY_MS 25
42 static const uint8_t a2dp_src_uuid[] = {
43 0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
44 0x80, 0x00, 0x00, 0x80, 0x5f, 0x9b, 0x34, 0xfb };
46 static int listen_sk = -1;
47 static int audio_sk = -1;
49 static pthread_t ipc_th = 0;
50 static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
52 #if __BYTE_ORDER == __LITTLE_ENDIAN
63 uint16_t sequence_number;
67 } __attribute__ ((packed));
70 unsigned frame_count:4;
72 unsigned is_last_fragment:1;
73 unsigned is_first_fragment:1;
74 unsigned is_fragmented:1;
75 } __attribute__ ((packed));
77 #elif __BYTE_ORDER == __BIG_ENDIAN
88 uint16_t sequence_number;
92 } __attribute__ ((packed));
95 unsigned is_fragmented:1;
96 unsigned is_first_fragment:1;
97 unsigned is_last_fragment:1;
99 unsigned frame_count:4;
100 } __attribute__ ((packed));
103 #error "Unknown byte order"
106 struct media_packet {
107 struct rtp_header hdr;
108 struct rtp_payload payload;
112 struct audio_input_config {
115 audio_format_t format;
129 unsigned frame_duration;
130 unsigned frames_per_packet;
132 struct timespec start;
133 unsigned frames_sent;
138 static inline void timespec_diff(struct timespec *a, struct timespec *b,
139 struct timespec *res)
141 res->tv_sec = a->tv_sec - b->tv_sec;
142 res->tv_nsec = a->tv_nsec - b->tv_nsec;
144 if (res->tv_nsec < 0) {
146 res->tv_nsec += 1000000000; /* 1sec */
150 static int sbc_get_presets(struct audio_preset *preset, size_t *len);
151 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
153 static int sbc_cleanup(void *codec_data);
154 static int sbc_get_config(void *codec_data, struct audio_input_config *config);
155 static size_t sbc_get_buffer_size(void *codec_data);
156 static size_t sbc_get_mediapacket_duration(void *codec_data);
157 static void sbc_resume(void *codec_data);
158 static ssize_t sbc_write_data(void *codec_data, const void *buffer,
159 size_t bytes, int fd);
164 int (*get_presets) (struct audio_preset *preset, size_t *len);
166 int (*init) (struct audio_preset *preset, uint16_t mtu,
168 int (*cleanup) (void *codec_data);
169 int (*get_config) (void *codec_data,
170 struct audio_input_config *config);
171 size_t (*get_buffer_size) (void *codec_data);
172 size_t (*get_mediapacket_duration) (void *codec_data);
173 void (*resume) (void *codec_data);
174 ssize_t (*write_data) (void *codec_data, const void *buffer,
175 size_t bytes, int fd);
178 static const struct audio_codec audio_codecs[] = {
180 .type = A2DP_CODEC_SBC,
182 .get_presets = sbc_get_presets,
184 .init = sbc_codec_init,
185 .cleanup = sbc_cleanup,
186 .get_config = sbc_get_config,
187 .get_buffer_size = sbc_get_buffer_size,
188 .get_mediapacket_duration = sbc_get_mediapacket_duration,
189 .resume = sbc_resume,
190 .write_data = sbc_write_data,
194 #define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
196 #define MAX_AUDIO_ENDPOINTS NUM_CODECS
198 struct audio_endpoint {
200 const struct audio_codec *codec;
205 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
208 AUDIO_A2DP_STATE_NONE,
209 AUDIO_A2DP_STATE_STANDBY,
210 AUDIO_A2DP_STATE_SUSPENDED,
211 AUDIO_A2DP_STATE_STARTED
214 struct a2dp_stream_out {
215 struct audio_stream_out stream;
217 struct audio_endpoint *ep;
218 enum a2dp_state_t audio_state;
219 struct audio_input_config cfg;
222 struct a2dp_audio_dev {
223 struct audio_hw_device dev;
224 struct a2dp_stream_out *out;
227 static const a2dp_sbc_t sbc_presets[] = {
229 .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
230 .channel_mode = SBC_CHANNEL_MODE_MONO |
231 SBC_CHANNEL_MODE_DUAL_CHANNEL |
232 SBC_CHANNEL_MODE_STEREO |
233 SBC_CHANNEL_MODE_JOINT_STEREO,
234 .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
235 .allocation_method = SBC_ALLOCATION_SNR |
236 SBC_ALLOCATION_LOUDNESS,
237 .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
238 SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
239 .min_bitpool = MIN_BITPOOL,
240 .max_bitpool = MAX_BITPOOL
243 .frequency = SBC_SAMPLING_FREQ_44100,
244 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
245 .subbands = SBC_SUBBANDS_8,
246 .allocation_method = SBC_ALLOCATION_LOUDNESS,
247 .block_length = SBC_BLOCK_LENGTH_16,
248 .min_bitpool = MIN_BITPOOL,
249 .max_bitpool = MAX_BITPOOL
252 .frequency = SBC_SAMPLING_FREQ_48000,
253 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
254 .subbands = SBC_SUBBANDS_8,
255 .allocation_method = SBC_ALLOCATION_LOUDNESS,
256 .block_length = SBC_BLOCK_LENGTH_16,
257 .min_bitpool = MIN_BITPOOL,
258 .max_bitpool = MAX_BITPOOL
262 static int sbc_get_presets(struct audio_preset *preset, size_t *len)
267 uint8_t *ptr = (uint8_t *) preset;
268 size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
270 count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
272 for (i = 0; i < count; i++) {
273 preset = (struct audio_preset *) ptr;
275 if (new_len + preset_size > *len)
278 preset->len = sizeof(a2dp_sbc_t);
279 memcpy(preset->data, &sbc_presets[i], preset->len);
281 new_len += preset_size;
290 static void sbc_init_encoder(struct sbc_data *sbc_data)
292 a2dp_sbc_t *in = &sbc_data->sbc;
293 sbc_t *out = &sbc_data->enc;
295 sbc_init_a2dp(out, 0L, in, sizeof(*in));
297 out->endian = SBC_LE;
298 out->bitpool = in->max_bitpool;
301 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
304 struct sbc_data *sbc_data;
305 size_t hdr_len = sizeof(struct media_packet);
307 size_t out_frame_len;
310 if (preset->len != sizeof(a2dp_sbc_t)) {
311 error("SBC: preset size mismatch");
312 return AUDIO_STATUS_FAILED;
315 sbc_data = calloc(sizeof(struct sbc_data), 1);
317 return AUDIO_STATUS_FAILED;
319 memcpy(&sbc_data->sbc, preset->data, preset->len);
321 sbc_init_encoder(sbc_data);
323 in_frame_len = sbc_get_codesize(&sbc_data->enc);
324 out_frame_len = sbc_get_frame_length(&sbc_data->enc);
325 num_frames = (mtu - hdr_len) / out_frame_len;
327 sbc_data->in_frame_len = in_frame_len;
328 sbc_data->in_buf_size = num_frames * in_frame_len;
330 sbc_data->out_buf_size = hdr_len + num_frames * out_frame_len;
331 sbc_data->out_buf = calloc(1, sbc_data->out_buf_size);
333 sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
334 sbc_data->frames_per_packet = num_frames;
336 *codec_data = sbc_data;
338 return AUDIO_STATUS_SUCCESS;
341 static int sbc_cleanup(void *codec_data)
343 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
345 sbc_finish(&sbc_data->enc);
346 free(sbc_data->out_buf);
349 return AUDIO_STATUS_SUCCESS;
352 static int sbc_get_config(void *codec_data, struct audio_input_config *config)
354 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
356 switch (sbc_data->sbc.frequency) {
357 case SBC_SAMPLING_FREQ_16000:
358 config->rate = 16000;
360 case SBC_SAMPLING_FREQ_32000:
361 config->rate = 32000;
363 case SBC_SAMPLING_FREQ_44100:
364 config->rate = 44100;
366 case SBC_SAMPLING_FREQ_48000:
367 config->rate = 48000;
370 return AUDIO_STATUS_FAILED;
372 config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
373 AUDIO_CHANNEL_OUT_MONO :
374 AUDIO_CHANNEL_OUT_STEREO;
375 config->format = AUDIO_FORMAT_PCM_16_BIT;
377 return AUDIO_STATUS_SUCCESS;
380 static size_t sbc_get_buffer_size(void *codec_data)
382 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
384 return sbc_data->in_buf_size;
387 static size_t sbc_get_mediapacket_duration(void *codec_data)
389 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
391 return sbc_data->frame_duration * sbc_data->frames_per_packet;
394 static void sbc_resume(void *codec_data)
396 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
400 clock_gettime(CLOCK_MONOTONIC, &sbc_data->start);
402 sbc_data->frames_sent = 0;
405 static int write_media_packet(int fd, struct sbc_data *sbc_data,
406 struct media_packet *mp, size_t data_len)
409 struct timespec diff;
410 unsigned expected_frames;
414 ret = write(fd, mp, sizeof(*mp) + data_len);
422 sbc_data->frames_sent += mp->payload.frame_count;
424 clock_gettime(CLOCK_MONOTONIC, &cur);
425 timespec_diff(&cur, &sbc_data->start, &diff);
426 expected_frames = (diff.tv_sec * 1000000 + diff.tv_nsec / 1000) /
427 sbc_data->frame_duration;
429 /* AudioFlinger does not seem to provide any *working*
430 * API to provide data in some interval and will just
431 * send another buffer as soon as we process current
432 * one. To prevent overflowing L2CAP socket, we need to
433 * introduce some artificial delay here base on how many
434 * audio frames were sent so far, i.e. if we're not
435 * lagging behind audio stream, we can sleep for
436 * duration of single media packet.
438 if (sbc_data->frames_sent >= expected_frames)
439 usleep(sbc_data->frame_duration *
440 mp->payload.frame_count);
445 static ssize_t sbc_write_data(void *codec_data, const void *buffer,
446 size_t bytes, int fd)
448 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
451 struct media_packet *mp = (struct media_packet *) sbc_data->out_buf;
452 size_t free_space = sbc_data->out_buf_size - sizeof(*mp);
457 mp->hdr.sequence_number = htons(sbc_data->seq++);
458 mp->hdr.ssrc = htonl(1);
459 mp->payload.frame_count = 0;
461 while (bytes - consumed >= sbc_data->in_frame_len) {
464 ret = sbc_encode(&sbc_data->enc, buffer + consumed,
465 sbc_data->in_frame_len,
466 mp->data + encoded, free_space,
470 error("SBC: failed to encode block");
474 mp->payload.frame_count++;
478 free_space -= written;
480 /* write data if we either filled media packed or encoded all
483 if (mp->payload.frame_count == sbc_data->frames_per_packet ||
485 ret = write_media_packet(fd, sbc_data, mp, encoded);
490 free_space = sbc_data->out_buf_size - sizeof(*mp);
491 mp->payload.frame_count = 0;
495 if (consumed != bytes) {
496 /* we should encode all input data
497 * if we did not, something went wrong but we can't really
498 * handle this so this is just sanity check
500 error("SBC: failed to encode complete input buffer");
503 /* we always assume that all data was processed and sent */
507 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
508 void *param, size_t *rsp_len, void *rsp, int *fd)
514 char cmsgbuf[CMSG_SPACE(sizeof(int))];
516 size_t s_len = sizeof(s);
518 pthread_mutex_lock(&sk_mutex);
521 error("audio: Invalid cmd socket passed to audio_ipc_cmd");
525 if (!rsp || !rsp_len) {
526 memset(&s, 0, s_len);
531 memset(&msg, 0, sizeof(msg));
532 memset(&cmd, 0, sizeof(cmd));
534 cmd.service_id = service_id;
538 iv[0].iov_base = &cmd;
539 iv[0].iov_len = sizeof(cmd);
541 iv[1].iov_base = param;
547 ret = sendmsg(audio_sk, &msg, 0);
549 error("audio: Sending command failed:%s", strerror(errno));
553 /* socket was shutdown */
555 error("audio: Command socket closed");
559 memset(&msg, 0, sizeof(msg));
560 memset(&cmd, 0, sizeof(cmd));
562 iv[0].iov_base = &cmd;
563 iv[0].iov_len = sizeof(cmd);
565 iv[1].iov_base = rsp;
566 iv[1].iov_len = *rsp_len;
572 memset(cmsgbuf, 0, sizeof(cmsgbuf));
573 msg.msg_control = cmsgbuf;
574 msg.msg_controllen = sizeof(cmsgbuf);
577 ret = recvmsg(audio_sk, &msg, 0);
579 error("audio: Receiving command response failed:%s",
584 if (ret < (ssize_t) sizeof(cmd)) {
585 error("audio: Too small response received(%zd bytes)", ret);
589 if (cmd.service_id != service_id) {
590 error("audio: Invalid service id (%u vs %u)", cmd.service_id,
595 if (ret != (ssize_t) (sizeof(cmd) + cmd.len)) {
596 error("audio: Malformed response received(%zd bytes)", ret);
600 if (cmd.opcode != opcode && cmd.opcode != AUDIO_OP_STATUS) {
601 error("audio: Invalid opcode received (%u vs %u)",
606 if (cmd.opcode == AUDIO_OP_STATUS) {
607 struct hal_status *s = rsp;
609 if (sizeof(*s) != cmd.len) {
610 error("audio: Invalid status length");
614 if (s->code == AUDIO_STATUS_SUCCESS) {
615 error("audio: Invalid success status response");
619 pthread_mutex_unlock(&sk_mutex);
624 pthread_mutex_unlock(&sk_mutex);
626 /* Receive auxiliary data in msg */
628 struct cmsghdr *cmsg;
632 for (cmsg = CMSG_FIRSTHDR(&msg); cmsg;
633 cmsg = CMSG_NXTHDR(&msg, cmsg)) {
634 if (cmsg->cmsg_level == SOL_SOCKET
635 && cmsg->cmsg_type == SCM_RIGHTS) {
636 memcpy(fd, CMSG_DATA(cmsg), sizeof(int));
648 return AUDIO_STATUS_SUCCESS;
651 /* Some serious issue happen on IPC - recover */
652 shutdown(audio_sk, SHUT_RDWR);
653 pthread_mutex_unlock(&sk_mutex);
655 return AUDIO_STATUS_FAILED;
658 static int ipc_open_cmd(const struct audio_codec *codec)
660 uint8_t buf[BLUEZ_AUDIO_MTU];
661 struct audio_cmd_open *cmd = (struct audio_cmd_open *) buf;
662 struct audio_rsp_open rsp;
663 size_t cmd_len = sizeof(buf) - sizeof(*cmd);
664 size_t rsp_len = sizeof(rsp);
669 memcpy(cmd->uuid, a2dp_src_uuid, sizeof(a2dp_src_uuid));
671 cmd->codec = codec->type;
672 cmd->presets = codec->get_presets(cmd->preset, &cmd_len);
674 cmd_len += sizeof(*cmd);
676 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN, cmd_len, cmd,
677 &rsp_len, &rsp, NULL);
679 if (result != AUDIO_STATUS_SUCCESS)
685 static int ipc_close_cmd(uint8_t endpoint_id)
687 struct audio_cmd_close cmd;
692 cmd.id = endpoint_id;
694 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE,
695 sizeof(cmd), &cmd, NULL, NULL, NULL);
700 static int ipc_open_stream_cmd(uint8_t endpoint_id, uint16_t *mtu, int *fd,
701 struct audio_preset **caps)
703 char buf[BLUEZ_AUDIO_MTU];
704 struct audio_cmd_open_stream cmd;
705 struct audio_rsp_open_stream *rsp =
706 (struct audio_rsp_open_stream *) &buf;
707 size_t rsp_len = sizeof(buf);
713 return AUDIO_STATUS_FAILED;
715 cmd.id = endpoint_id;
717 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN_STREAM,
718 sizeof(cmd), &cmd, &rsp_len, rsp, fd);
719 if (result == AUDIO_STATUS_SUCCESS) {
720 size_t buf_len = sizeof(struct audio_preset) +
723 *caps = malloc(buf_len);
724 memcpy(*caps, &rsp->preset, buf_len);
732 static int ipc_close_stream_cmd(uint8_t endpoint_id)
734 struct audio_cmd_close_stream cmd;
739 cmd.id = endpoint_id;
741 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE_STREAM,
742 sizeof(cmd), &cmd, NULL, NULL, NULL);
747 static int ipc_resume_stream_cmd(uint8_t endpoint_id)
749 struct audio_cmd_resume_stream cmd;
754 cmd.id = endpoint_id;
756 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_RESUME_STREAM,
757 sizeof(cmd), &cmd, NULL, NULL, NULL);
762 static int ipc_suspend_stream_cmd(uint8_t endpoint_id)
764 struct audio_cmd_suspend_stream cmd;
769 cmd.id = endpoint_id;
771 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_SUSPEND_STREAM,
772 sizeof(cmd), &cmd, NULL, NULL, NULL);
777 static int register_endpoints(void)
779 struct audio_endpoint *ep = &audio_endpoints[0];
782 for (i = 0; i < NUM_CODECS; i++, ep++) {
783 const struct audio_codec *codec = &audio_codecs[i];
785 ep->id = ipc_open_cmd(codec);
788 return AUDIO_STATUS_FAILED;
791 ep->codec_data = NULL;
795 return AUDIO_STATUS_SUCCESS;
798 static void unregister_endpoints(void)
802 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++) {
803 struct audio_endpoint *ep = &audio_endpoints[i];
806 ipc_close_cmd(ep->id);
807 memset(ep, 0, sizeof(*ep));
812 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
815 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
817 /* We can auto-start only from standby */
818 if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
819 DBG("stream in standby, auto-start");
821 if (ipc_resume_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
824 out->ep->codec->resume(out->ep->codec_data);
826 out->audio_state = AUDIO_A2DP_STATE_STARTED;
829 if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
830 error("audio: stream not started");
834 if (out->ep->fd < 0) {
835 error("audio: no transport socket");
839 return out->ep->codec->write_data(out->ep->codec_data, buffer,
843 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
845 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
849 return out->cfg.rate;
852 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
854 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
858 if (rate != out->cfg.rate) {
859 warn("audio: cannot set sample rate to %d", rate);
866 static size_t out_get_buffer_size(const struct audio_stream *stream)
870 /* We should return proper buffer size calculated by codec (so each
871 * input buffer is encoded into single media packed) but this does not
872 * work well with AudioFlinger and causes problems. For this reason we
873 * use magic value here and out_write code takes care of splitting
874 * input buffer into multiple media packets.
879 static uint32_t out_get_channels(const struct audio_stream *stream)
881 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
885 return out->cfg.channels;
888 static audio_format_t out_get_format(const struct audio_stream *stream)
890 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
894 return out->cfg.format;
897 static int out_set_format(struct audio_stream *stream, audio_format_t format)
903 static int out_standby(struct audio_stream *stream)
905 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
909 if (out->audio_state == AUDIO_A2DP_STATE_STARTED) {
910 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
912 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
918 static int out_dump(const struct audio_stream *stream, int fd)
924 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
926 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
930 bool enter_suspend = false;
931 bool exit_suspend = false;
935 str = strdup(kvpairs);
936 kvpair = strtok_r(str, ";", &saveptr);
938 for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
941 keyval = strchr(kvpair, '=');
948 if (!strcmp(kvpair, "closing")) {
949 if (!strcmp(keyval, "true"))
950 out->audio_state = AUDIO_A2DP_STATE_NONE;
951 } else if (!strcmp(kvpair, "A2dpSuspended")) {
952 if (!strcmp(keyval, "true"))
953 enter_suspend = true;
961 if (enter_suspend && out->audio_state == AUDIO_A2DP_STATE_STARTED) {
962 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
964 out->audio_state = AUDIO_A2DP_STATE_SUSPENDED;
967 if (exit_suspend && out->audio_state == AUDIO_A2DP_STATE_SUSPENDED)
968 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
973 static char *out_get_parameters(const struct audio_stream *stream,
980 static uint32_t out_get_latency(const struct audio_stream_out *stream)
982 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
983 struct audio_endpoint *ep = out->ep;
988 pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
990 return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
993 static int out_set_volume(struct audio_stream_out *stream, float left,
997 /* volume controlled in audioflinger mixer (digital) */
1001 static int out_get_render_position(const struct audio_stream_out *stream,
1002 uint32_t *dsp_frames)
1008 static int out_add_audio_effect(const struct audio_stream *stream,
1009 effect_handle_t effect)
1015 static int out_remove_audio_effect(const struct audio_stream *stream,
1016 effect_handle_t effect)
1022 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1028 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1034 static size_t in_get_buffer_size(const struct audio_stream *stream)
1040 static uint32_t in_get_channels(const struct audio_stream *stream)
1046 static audio_format_t in_get_format(const struct audio_stream *stream)
1052 static int in_set_format(struct audio_stream *stream, audio_format_t format)
1058 static int in_standby(struct audio_stream *stream)
1064 static int in_dump(const struct audio_stream *stream, int fd)
1070 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1076 static char *in_get_parameters(const struct audio_stream *stream,
1083 static int in_set_gain(struct audio_stream_in *stream, float gain)
1089 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1096 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1102 static int in_add_audio_effect(const struct audio_stream *stream,
1103 effect_handle_t effect)
1109 static int in_remove_audio_effect(const struct audio_stream *stream,
1110 effect_handle_t effect)
1116 static int set_blocking(int fd)
1120 flags = fcntl(fd, F_GETFL, 0);
1122 error("fcntl(F_GETFL): %s (%d)", strerror(errno), errno);
1126 if (fcntl(fd, F_SETFL, flags & ~O_NONBLOCK) < 0) {
1127 error("fcntl(F_SETFL): %s (%d)", strerror(errno), errno);
1134 static int audio_open_output_stream(struct audio_hw_device *dev,
1135 audio_io_handle_t handle,
1136 audio_devices_t devices,
1137 audio_output_flags_t flags,
1138 struct audio_config *config,
1139 struct audio_stream_out **stream_out)
1142 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1143 struct a2dp_stream_out *out;
1144 struct audio_preset *preset;
1145 const struct audio_codec *codec;
1149 out = calloc(1, sizeof(struct a2dp_stream_out));
1155 out->stream.common.get_sample_rate = out_get_sample_rate;
1156 out->stream.common.set_sample_rate = out_set_sample_rate;
1157 out->stream.common.get_buffer_size = out_get_buffer_size;
1158 out->stream.common.get_channels = out_get_channels;
1159 out->stream.common.get_format = out_get_format;
1160 out->stream.common.set_format = out_set_format;
1161 out->stream.common.standby = out_standby;
1162 out->stream.common.dump = out_dump;
1163 out->stream.common.set_parameters = out_set_parameters;
1164 out->stream.common.get_parameters = out_get_parameters;
1165 out->stream.common.add_audio_effect = out_add_audio_effect;
1166 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1167 out->stream.get_latency = out_get_latency;
1168 out->stream.set_volume = out_set_volume;
1169 out->stream.write = out_write;
1170 out->stream.get_render_position = out_get_render_position;
1172 /* TODO: for now we always use endpoint 0 */
1173 out->ep = &audio_endpoints[0];
1175 if (ipc_open_stream_cmd(out->ep->id, &mtu, &fd, &preset) !=
1176 AUDIO_STATUS_SUCCESS)
1179 if (!preset || fd < 0)
1182 if (set_blocking(fd) < 0)
1186 codec = out->ep->codec;
1188 codec->init(preset, mtu, &out->ep->codec_data);
1189 codec->get_config(out->ep->codec_data, &out->cfg);
1191 DBG("rate=%d channels=%d format=%d", out->cfg.rate,
1192 out->cfg.channels, out->cfg.format);
1196 *stream_out = &out->stream;
1197 a2dp_dev->out = out;
1199 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1204 error("audio: cannot open output stream");
1210 static void audio_close_output_stream(struct audio_hw_device *dev,
1211 struct audio_stream_out *stream)
1213 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1214 struct audio_endpoint *ep = a2dp_dev->out->ep;
1218 ipc_close_stream_cmd(ep->id);
1225 ep->codec->cleanup(ep->codec_data);
1226 ep->codec_data = NULL;
1229 a2dp_dev->out = NULL;
1232 static int audio_set_parameters(struct audio_hw_device *dev,
1233 const char *kvpairs)
1235 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1236 struct a2dp_stream_out *out = a2dp_dev->out;
1243 return out->stream.common.set_parameters((struct audio_stream *) out,
1247 static char *audio_get_parameters(const struct audio_hw_device *dev,
1254 static int audio_init_check(const struct audio_hw_device *dev)
1260 static int audio_set_voice_volume(struct audio_hw_device *dev, float volume)
1266 static int audio_set_master_volume(struct audio_hw_device *dev, float volume)
1272 static int audio_set_mode(struct audio_hw_device *dev, int mode)
1278 static int audio_set_mic_mute(struct audio_hw_device *dev, bool state)
1284 static int audio_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1290 static size_t audio_get_input_buffer_size(const struct audio_hw_device *dev,
1291 const struct audio_config *config)
1297 static int audio_open_input_stream(struct audio_hw_device *dev,
1298 audio_io_handle_t handle,
1299 audio_devices_t devices,
1300 struct audio_config *config,
1301 struct audio_stream_in **stream_in)
1303 struct audio_stream_in *in;
1307 in = calloc(1, sizeof(struct audio_stream_in));
1311 in->common.get_sample_rate = in_get_sample_rate;
1312 in->common.set_sample_rate = in_set_sample_rate;
1313 in->common.get_buffer_size = in_get_buffer_size;
1314 in->common.get_channels = in_get_channels;
1315 in->common.get_format = in_get_format;
1316 in->common.set_format = in_set_format;
1317 in->common.standby = in_standby;
1318 in->common.dump = in_dump;
1319 in->common.set_parameters = in_set_parameters;
1320 in->common.get_parameters = in_get_parameters;
1321 in->common.add_audio_effect = in_add_audio_effect;
1322 in->common.remove_audio_effect = in_remove_audio_effect;
1323 in->set_gain = in_set_gain;
1325 in->get_input_frames_lost = in_get_input_frames_lost;
1332 static void audio_close_input_stream(struct audio_hw_device *dev,
1333 struct audio_stream_in *stream_in)
1339 static int audio_dump(const audio_hw_device_t *device, int fd)
1345 static int audio_close(hw_device_t *device)
1347 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
1351 unregister_endpoints();
1353 shutdown(listen_sk, SHUT_RDWR);
1354 shutdown(audio_sk, SHUT_RDWR);
1356 pthread_join(ipc_th, NULL);
1365 static void *ipc_handler(void *data)
1374 DBG("Waiting for connection ...");
1376 sk = accept(listen_sk, NULL, NULL);
1383 if (err != ECONNABORTED && err != EINVAL)
1384 error("audio: Failed to accept socket: %d (%s)",
1385 err, strerror(err));
1390 pthread_mutex_lock(&sk_mutex);
1392 pthread_mutex_unlock(&sk_mutex);
1394 DBG("Audio IPC: Connected");
1396 if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
1397 error("audio: Failed to register endpoints");
1399 unregister_endpoints();
1401 pthread_mutex_lock(&sk_mutex);
1402 shutdown(audio_sk, SHUT_RDWR);
1405 pthread_mutex_unlock(&sk_mutex);
1410 memset(&pfd, 0, sizeof(pfd));
1412 pfd.events = POLLHUP | POLLERR | POLLNVAL;
1414 /* Check if socket is still alive. Empty while loop.*/
1415 while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
1417 if (pfd.revents & (POLLHUP | POLLERR | POLLNVAL)) {
1418 info("Audio HAL: Socket closed");
1420 pthread_mutex_lock(&sk_mutex);
1423 pthread_mutex_unlock(&sk_mutex);
1427 /* audio_sk is closed at this point, just cleanup endpoints states */
1428 memset(audio_endpoints, 0, sizeof(audio_endpoints));
1430 info("Closing Audio IPC thread");
1434 static int audio_ipc_init(void)
1436 struct sockaddr_un addr;
1442 sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
1445 error("audio: Failed to create socket: %d (%s)", err,
1450 memset(&addr, 0, sizeof(addr));
1451 addr.sun_family = AF_UNIX;
1453 memcpy(addr.sun_path, BLUEZ_AUDIO_SK_PATH,
1454 sizeof(BLUEZ_AUDIO_SK_PATH));
1456 if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
1458 error("audio: Failed to bind socket: %d (%s)", err,
1463 if (listen(sk, 1) < 0) {
1465 error("audio: Failed to listen on the socket: %d (%s)", err,
1472 err = pthread_create(&ipc_th, NULL, ipc_handler, NULL);
1476 error("audio: Failed to start Audio IPC thread: %d (%s)",
1477 err, strerror(err));
1488 static int audio_open(const hw_module_t *module, const char *name,
1489 hw_device_t **device)
1491 struct a2dp_audio_dev *a2dp_dev;
1496 if (strcmp(name, AUDIO_HARDWARE_INTERFACE)) {
1497 error("audio: interface %s not matching [%s]", name,
1498 AUDIO_HARDWARE_INTERFACE);
1502 err = audio_ipc_init();
1506 a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
1510 a2dp_dev->dev.common.tag = HARDWARE_DEVICE_TAG;
1511 a2dp_dev->dev.common.version = AUDIO_DEVICE_API_VERSION_CURRENT;
1512 a2dp_dev->dev.common.module = (struct hw_module_t *) module;
1513 a2dp_dev->dev.common.close = audio_close;
1515 a2dp_dev->dev.init_check = audio_init_check;
1516 a2dp_dev->dev.set_voice_volume = audio_set_voice_volume;
1517 a2dp_dev->dev.set_master_volume = audio_set_master_volume;
1518 a2dp_dev->dev.set_mode = audio_set_mode;
1519 a2dp_dev->dev.set_mic_mute = audio_set_mic_mute;
1520 a2dp_dev->dev.get_mic_mute = audio_get_mic_mute;
1521 a2dp_dev->dev.set_parameters = audio_set_parameters;
1522 a2dp_dev->dev.get_parameters = audio_get_parameters;
1523 a2dp_dev->dev.get_input_buffer_size = audio_get_input_buffer_size;
1524 a2dp_dev->dev.open_output_stream = audio_open_output_stream;
1525 a2dp_dev->dev.close_output_stream = audio_close_output_stream;
1526 a2dp_dev->dev.open_input_stream = audio_open_input_stream;
1527 a2dp_dev->dev.close_input_stream = audio_close_input_stream;
1528 a2dp_dev->dev.dump = audio_dump;
1530 /* Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
1531 * This results from the structure of following structs:a2dp_audio_dev,
1532 * audio_hw_device. We will rely on this later in the code.*/
1533 *device = &a2dp_dev->dev.common;
1538 static struct hw_module_methods_t hal_module_methods = {
1542 struct audio_module HAL_MODULE_INFO_SYM = {
1544 .tag = HARDWARE_MODULE_TAG,
1547 .id = AUDIO_HARDWARE_MODULE_ID,
1548 .name = "A2DP Bluez HW HAL",
1549 .author = "Intel Corporation",
1550 .methods = &hal_module_methods,