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 #define MAX_FRAMES_IN_PAYLOAD 15
44 static const uint8_t a2dp_src_uuid[] = {
45 0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
46 0x80, 0x00, 0x00, 0x80, 0x5f, 0x9b, 0x34, 0xfb };
48 static int listen_sk = -1;
49 static int audio_sk = -1;
51 static pthread_t ipc_th = 0;
52 static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
54 #if __BYTE_ORDER == __LITTLE_ENDIAN
65 uint16_t sequence_number;
69 } __attribute__ ((packed));
72 unsigned frame_count:4;
74 unsigned is_last_fragment:1;
75 unsigned is_first_fragment:1;
76 unsigned is_fragmented:1;
77 } __attribute__ ((packed));
79 #elif __BYTE_ORDER == __BIG_ENDIAN
90 uint16_t sequence_number;
94 } __attribute__ ((packed));
97 unsigned is_fragmented:1;
98 unsigned is_first_fragment:1;
99 unsigned is_last_fragment:1;
101 unsigned frame_count:4;
102 } __attribute__ ((packed));
105 #error "Unknown byte order"
108 struct media_packet {
109 struct rtp_header hdr;
110 struct rtp_payload payload;
114 struct audio_input_config {
117 audio_format_t format;
131 unsigned frame_duration;
132 unsigned frames_per_packet;
134 struct timespec start;
135 unsigned frames_sent;
141 static inline void timespec_diff(struct timespec *a, struct timespec *b,
142 struct timespec *res)
144 res->tv_sec = a->tv_sec - b->tv_sec;
145 res->tv_nsec = a->tv_nsec - b->tv_nsec;
147 if (res->tv_nsec < 0) {
149 res->tv_nsec += 1000000000; /* 1sec */
153 static int sbc_get_presets(struct audio_preset *preset, size_t *len);
154 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
156 static int sbc_cleanup(void *codec_data);
157 static int sbc_get_config(void *codec_data, struct audio_input_config *config);
158 static size_t sbc_get_buffer_size(void *codec_data);
159 static size_t sbc_get_mediapacket_duration(void *codec_data);
160 static void sbc_resume(void *codec_data);
161 static ssize_t sbc_write_data(void *codec_data, const void *buffer,
162 size_t bytes, int fd);
167 int (*get_presets) (struct audio_preset *preset, size_t *len);
169 int (*init) (struct audio_preset *preset, uint16_t mtu,
171 int (*cleanup) (void *codec_data);
172 int (*get_config) (void *codec_data,
173 struct audio_input_config *config);
174 size_t (*get_buffer_size) (void *codec_data);
175 size_t (*get_mediapacket_duration) (void *codec_data);
176 void (*resume) (void *codec_data);
177 ssize_t (*write_data) (void *codec_data, const void *buffer,
178 size_t bytes, int fd);
181 static const struct audio_codec audio_codecs[] = {
183 .type = A2DP_CODEC_SBC,
185 .get_presets = sbc_get_presets,
187 .init = sbc_codec_init,
188 .cleanup = sbc_cleanup,
189 .get_config = sbc_get_config,
190 .get_buffer_size = sbc_get_buffer_size,
191 .get_mediapacket_duration = sbc_get_mediapacket_duration,
192 .resume = sbc_resume,
193 .write_data = sbc_write_data,
197 #define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
199 #define MAX_AUDIO_ENDPOINTS NUM_CODECS
201 struct audio_endpoint {
203 const struct audio_codec *codec;
208 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
211 AUDIO_A2DP_STATE_NONE,
212 AUDIO_A2DP_STATE_STANDBY,
213 AUDIO_A2DP_STATE_SUSPENDED,
214 AUDIO_A2DP_STATE_STARTED
217 struct a2dp_stream_out {
218 struct audio_stream_out stream;
220 struct audio_endpoint *ep;
221 enum a2dp_state_t audio_state;
222 struct audio_input_config cfg;
225 struct a2dp_audio_dev {
226 struct audio_hw_device dev;
227 struct a2dp_stream_out *out;
230 static const a2dp_sbc_t sbc_presets[] = {
232 .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
233 .channel_mode = SBC_CHANNEL_MODE_DUAL_CHANNEL |
234 SBC_CHANNEL_MODE_STEREO |
235 SBC_CHANNEL_MODE_JOINT_STEREO,
236 .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
237 .allocation_method = SBC_ALLOCATION_SNR |
238 SBC_ALLOCATION_LOUDNESS,
239 .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
240 SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
241 .min_bitpool = MIN_BITPOOL,
242 .max_bitpool = MAX_BITPOOL
245 .frequency = SBC_SAMPLING_FREQ_44100,
246 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
247 .subbands = SBC_SUBBANDS_8,
248 .allocation_method = SBC_ALLOCATION_LOUDNESS,
249 .block_length = SBC_BLOCK_LENGTH_16,
250 .min_bitpool = MIN_BITPOOL,
251 .max_bitpool = MAX_BITPOOL
254 .frequency = SBC_SAMPLING_FREQ_48000,
255 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
256 .subbands = SBC_SUBBANDS_8,
257 .allocation_method = SBC_ALLOCATION_LOUDNESS,
258 .block_length = SBC_BLOCK_LENGTH_16,
259 .min_bitpool = MIN_BITPOOL,
260 .max_bitpool = MAX_BITPOOL
264 static int sbc_get_presets(struct audio_preset *preset, size_t *len)
269 uint8_t *ptr = (uint8_t *) preset;
270 size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
272 count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
274 for (i = 0; i < count; i++) {
275 preset = (struct audio_preset *) ptr;
277 if (new_len + preset_size > *len)
280 preset->len = sizeof(a2dp_sbc_t);
281 memcpy(preset->data, &sbc_presets[i], preset->len);
283 new_len += preset_size;
292 static int sbc_freq2int(uint8_t freq)
295 case SBC_SAMPLING_FREQ_16000:
297 case SBC_SAMPLING_FREQ_32000:
299 case SBC_SAMPLING_FREQ_44100:
301 case SBC_SAMPLING_FREQ_48000:
308 static const char *sbc_mode2str(uint8_t mode)
311 case SBC_CHANNEL_MODE_MONO:
313 case SBC_CHANNEL_MODE_DUAL_CHANNEL:
314 return "DualChannel";
315 case SBC_CHANNEL_MODE_STEREO:
317 case SBC_CHANNEL_MODE_JOINT_STEREO:
318 return "JointStereo";
324 static int sbc_blocks2int(uint8_t blocks)
327 case SBC_BLOCK_LENGTH_4:
329 case SBC_BLOCK_LENGTH_8:
331 case SBC_BLOCK_LENGTH_12:
333 case SBC_BLOCK_LENGTH_16:
340 static int sbc_subbands2int(uint8_t subbands)
352 static const char *sbc_allocation2str(uint8_t allocation)
354 switch (allocation) {
355 case SBC_ALLOCATION_SNR:
357 case SBC_ALLOCATION_LOUDNESS:
364 static void sbc_init_encoder(struct sbc_data *sbc_data)
366 a2dp_sbc_t *in = &sbc_data->sbc;
367 sbc_t *out = &sbc_data->enc;
369 sbc_init_a2dp(out, 0L, in, sizeof(*in));
371 out->endian = SBC_LE;
372 out->bitpool = in->max_bitpool;
374 DBG("frequency=%d channel_mode=%s block_length=%d subbands=%d "
375 "allocation=%s bitpool=%d-%d",
376 sbc_freq2int(in->frequency),
377 sbc_mode2str(in->channel_mode),
378 sbc_blocks2int(in->block_length),
379 sbc_subbands2int(in->subbands),
380 sbc_allocation2str(in->allocation_method),
381 in->min_bitpool, in->max_bitpool);
384 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
387 struct sbc_data *sbc_data;
388 size_t hdr_len = sizeof(struct media_packet);
390 size_t out_frame_len;
393 if (preset->len != sizeof(a2dp_sbc_t)) {
394 error("SBC: preset size mismatch");
395 return AUDIO_STATUS_FAILED;
398 sbc_data = calloc(sizeof(struct sbc_data), 1);
400 return AUDIO_STATUS_FAILED;
402 memcpy(&sbc_data->sbc, preset->data, preset->len);
404 sbc_init_encoder(sbc_data);
406 in_frame_len = sbc_get_codesize(&sbc_data->enc);
407 out_frame_len = sbc_get_frame_length(&sbc_data->enc);
408 num_frames = (mtu - hdr_len) / out_frame_len;
410 sbc_data->in_frame_len = in_frame_len;
411 sbc_data->in_buf_size = num_frames * in_frame_len;
413 sbc_data->out_buf_size = hdr_len + num_frames * out_frame_len;
414 sbc_data->out_buf = calloc(1, sbc_data->out_buf_size);
416 sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
417 sbc_data->frames_per_packet = num_frames;
419 DBG("mtu=%u in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
420 mtu, in_frame_len, out_frame_len, num_frames);
422 *codec_data = sbc_data;
424 return AUDIO_STATUS_SUCCESS;
427 static int sbc_cleanup(void *codec_data)
429 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
431 sbc_finish(&sbc_data->enc);
432 free(sbc_data->out_buf);
435 return AUDIO_STATUS_SUCCESS;
438 static int sbc_get_config(void *codec_data, struct audio_input_config *config)
440 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
442 switch (sbc_data->sbc.frequency) {
443 case SBC_SAMPLING_FREQ_16000:
444 config->rate = 16000;
446 case SBC_SAMPLING_FREQ_32000:
447 config->rate = 32000;
449 case SBC_SAMPLING_FREQ_44100:
450 config->rate = 44100;
452 case SBC_SAMPLING_FREQ_48000:
453 config->rate = 48000;
456 return AUDIO_STATUS_FAILED;
458 config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
459 AUDIO_CHANNEL_OUT_MONO :
460 AUDIO_CHANNEL_OUT_STEREO;
461 config->format = AUDIO_FORMAT_PCM_16_BIT;
463 return AUDIO_STATUS_SUCCESS;
466 static size_t sbc_get_buffer_size(void *codec_data)
468 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
470 return sbc_data->in_buf_size;
473 static size_t sbc_get_mediapacket_duration(void *codec_data)
475 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
477 return sbc_data->frame_duration * sbc_data->frames_per_packet;
480 static void sbc_resume(void *codec_data)
482 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
486 clock_gettime(CLOCK_MONOTONIC, &sbc_data->start);
488 sbc_data->frames_sent = 0;
489 sbc_data->timestamp = 0;
492 static int write_media_packet(int fd, struct sbc_data *sbc_data,
493 struct media_packet *mp, size_t data_len)
496 struct timespec diff;
497 unsigned expected_frames;
501 ret = write(fd, mp, sizeof(*mp) + data_len);
509 sbc_data->frames_sent += mp->payload.frame_count;
511 clock_gettime(CLOCK_MONOTONIC, &cur);
512 timespec_diff(&cur, &sbc_data->start, &diff);
513 expected_frames = (diff.tv_sec * 1000000 + diff.tv_nsec / 1000) /
514 sbc_data->frame_duration;
516 /* AudioFlinger does not seem to provide any *working*
517 * API to provide data in some interval and will just
518 * send another buffer as soon as we process current
519 * one. To prevent overflowing L2CAP socket, we need to
520 * introduce some artificial delay here base on how many
521 * audio frames were sent so far, i.e. if we're not
522 * lagging behind audio stream, we can sleep for
523 * duration of single media packet.
525 if (sbc_data->frames_sent >= expected_frames)
526 usleep(sbc_data->frame_duration *
527 mp->payload.frame_count);
532 static ssize_t sbc_write_data(void *codec_data, const void *buffer,
533 size_t bytes, int fd)
535 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
538 struct media_packet *mp = (struct media_packet *) sbc_data->out_buf;
539 size_t free_space = sbc_data->out_buf_size - sizeof(*mp);
545 mp->hdr.ssrc = htonl(1);
546 mp->hdr.timestamp = htonl(sbc_data->timestamp);
547 mp->payload.frame_count = 0;
549 while (bytes - consumed >= sbc_data->in_frame_len) {
552 bytes_read = sbc_encode(&sbc_data->enc, buffer + consumed,
553 sbc_data->in_frame_len,
554 mp->data + encoded, free_space,
557 if (bytes_read < 0) {
558 error("SBC: failed to encode block (%zd)", bytes_read);
562 mp->payload.frame_count++;
564 consumed += bytes_read;
566 free_space -= written;
568 /* AudioFlinger provides PCM 16bit stereo only, thus sample size
571 sbc_data->timestamp += (bytes_read / 4);
573 /* write data if we either filled media packed or encoded all
576 if (mp->payload.frame_count == sbc_data->frames_per_packet ||
578 mp->payload.frame_count ==
579 MAX_FRAMES_IN_PAYLOAD) {
580 mp->hdr.sequence_number = htons(sbc_data->seq++);
582 ret = write_media_packet(fd, sbc_data, mp, encoded);
587 free_space = sbc_data->out_buf_size - sizeof(*mp);
588 mp->hdr.timestamp = htonl(sbc_data->timestamp);
589 mp->payload.frame_count = 0;
593 if (consumed != bytes) {
594 /* we should encode all input data
595 * if we did not, something went wrong but we can't really
596 * handle this so this is just sanity check
598 error("SBC: failed to encode complete input buffer");
601 /* we always assume that all data was processed and sent */
605 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
606 void *param, size_t *rsp_len, void *rsp, int *fd)
612 char cmsgbuf[CMSG_SPACE(sizeof(int))];
614 size_t s_len = sizeof(s);
616 pthread_mutex_lock(&sk_mutex);
619 error("audio: Invalid cmd socket passed to audio_ipc_cmd");
623 if (!rsp || !rsp_len) {
624 memset(&s, 0, s_len);
629 memset(&msg, 0, sizeof(msg));
630 memset(&cmd, 0, sizeof(cmd));
632 cmd.service_id = service_id;
636 iv[0].iov_base = &cmd;
637 iv[0].iov_len = sizeof(cmd);
639 iv[1].iov_base = param;
645 ret = sendmsg(audio_sk, &msg, 0);
647 error("audio: Sending command failed:%s", strerror(errno));
651 /* socket was shutdown */
653 error("audio: Command socket closed");
657 memset(&msg, 0, sizeof(msg));
658 memset(&cmd, 0, sizeof(cmd));
660 iv[0].iov_base = &cmd;
661 iv[0].iov_len = sizeof(cmd);
663 iv[1].iov_base = rsp;
664 iv[1].iov_len = *rsp_len;
670 memset(cmsgbuf, 0, sizeof(cmsgbuf));
671 msg.msg_control = cmsgbuf;
672 msg.msg_controllen = sizeof(cmsgbuf);
675 ret = recvmsg(audio_sk, &msg, 0);
677 error("audio: Receiving command response failed:%s",
682 if (ret < (ssize_t) sizeof(cmd)) {
683 error("audio: Too small response received(%zd bytes)", ret);
687 if (cmd.service_id != service_id) {
688 error("audio: Invalid service id (%u vs %u)", cmd.service_id,
693 if (ret != (ssize_t) (sizeof(cmd) + cmd.len)) {
694 error("audio: Malformed response received(%zd bytes)", ret);
698 if (cmd.opcode != opcode && cmd.opcode != AUDIO_OP_STATUS) {
699 error("audio: Invalid opcode received (%u vs %u)",
704 if (cmd.opcode == AUDIO_OP_STATUS) {
705 struct hal_status *s = rsp;
707 if (sizeof(*s) != cmd.len) {
708 error("audio: Invalid status length");
712 if (s->code == AUDIO_STATUS_SUCCESS) {
713 error("audio: Invalid success status response");
717 pthread_mutex_unlock(&sk_mutex);
722 pthread_mutex_unlock(&sk_mutex);
724 /* Receive auxiliary data in msg */
726 struct cmsghdr *cmsg;
730 for (cmsg = CMSG_FIRSTHDR(&msg); cmsg;
731 cmsg = CMSG_NXTHDR(&msg, cmsg)) {
732 if (cmsg->cmsg_level == SOL_SOCKET
733 && cmsg->cmsg_type == SCM_RIGHTS) {
734 memcpy(fd, CMSG_DATA(cmsg), sizeof(int));
746 return AUDIO_STATUS_SUCCESS;
749 /* Some serious issue happen on IPC - recover */
750 shutdown(audio_sk, SHUT_RDWR);
751 pthread_mutex_unlock(&sk_mutex);
753 return AUDIO_STATUS_FAILED;
756 static int ipc_open_cmd(const struct audio_codec *codec)
758 uint8_t buf[BLUEZ_AUDIO_MTU];
759 struct audio_cmd_open *cmd = (struct audio_cmd_open *) buf;
760 struct audio_rsp_open rsp;
761 size_t cmd_len = sizeof(buf) - sizeof(*cmd);
762 size_t rsp_len = sizeof(rsp);
767 memcpy(cmd->uuid, a2dp_src_uuid, sizeof(a2dp_src_uuid));
769 cmd->codec = codec->type;
770 cmd->presets = codec->get_presets(cmd->preset, &cmd_len);
772 cmd_len += sizeof(*cmd);
774 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN, cmd_len, cmd,
775 &rsp_len, &rsp, NULL);
777 if (result != AUDIO_STATUS_SUCCESS)
783 static int ipc_close_cmd(uint8_t endpoint_id)
785 struct audio_cmd_close cmd;
790 cmd.id = endpoint_id;
792 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE,
793 sizeof(cmd), &cmd, NULL, NULL, NULL);
798 static int ipc_open_stream_cmd(uint8_t endpoint_id, uint16_t *mtu, int *fd,
799 struct audio_preset **caps)
801 char buf[BLUEZ_AUDIO_MTU];
802 struct audio_cmd_open_stream cmd;
803 struct audio_rsp_open_stream *rsp =
804 (struct audio_rsp_open_stream *) &buf;
805 size_t rsp_len = sizeof(buf);
811 return AUDIO_STATUS_FAILED;
813 cmd.id = endpoint_id;
815 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN_STREAM,
816 sizeof(cmd), &cmd, &rsp_len, rsp, fd);
817 if (result == AUDIO_STATUS_SUCCESS) {
818 size_t buf_len = sizeof(struct audio_preset) +
821 *caps = malloc(buf_len);
822 memcpy(*caps, &rsp->preset, buf_len);
830 static int ipc_close_stream_cmd(uint8_t endpoint_id)
832 struct audio_cmd_close_stream cmd;
837 cmd.id = endpoint_id;
839 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE_STREAM,
840 sizeof(cmd), &cmd, NULL, NULL, NULL);
845 static int ipc_resume_stream_cmd(uint8_t endpoint_id)
847 struct audio_cmd_resume_stream cmd;
852 cmd.id = endpoint_id;
854 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_RESUME_STREAM,
855 sizeof(cmd), &cmd, NULL, NULL, NULL);
860 static int ipc_suspend_stream_cmd(uint8_t endpoint_id)
862 struct audio_cmd_suspend_stream cmd;
867 cmd.id = endpoint_id;
869 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_SUSPEND_STREAM,
870 sizeof(cmd), &cmd, NULL, NULL, NULL);
875 static int register_endpoints(void)
877 struct audio_endpoint *ep = &audio_endpoints[0];
880 for (i = 0; i < NUM_CODECS; i++, ep++) {
881 const struct audio_codec *codec = &audio_codecs[i];
883 ep->id = ipc_open_cmd(codec);
886 return AUDIO_STATUS_FAILED;
889 ep->codec_data = NULL;
893 return AUDIO_STATUS_SUCCESS;
896 static void unregister_endpoints(void)
900 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++) {
901 struct audio_endpoint *ep = &audio_endpoints[i];
904 ipc_close_cmd(ep->id);
905 memset(ep, 0, sizeof(*ep));
910 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
913 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
915 /* just return in case we're closing */
916 if (out->audio_state == AUDIO_A2DP_STATE_NONE)
919 /* We can auto-start only from standby */
920 if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
921 DBG("stream in standby, auto-start");
923 if (ipc_resume_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
926 out->ep->codec->resume(out->ep->codec_data);
928 out->audio_state = AUDIO_A2DP_STATE_STARTED;
931 if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
932 error("audio: stream not started");
936 if (out->ep->fd < 0) {
937 error("audio: no transport socket");
941 return out->ep->codec->write_data(out->ep->codec_data, buffer,
945 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
947 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
951 return out->cfg.rate;
954 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
956 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
960 if (rate != out->cfg.rate) {
961 warn("audio: cannot set sample rate to %d", rate);
968 static size_t out_get_buffer_size(const struct audio_stream *stream)
972 /* We should return proper buffer size calculated by codec (so each
973 * input buffer is encoded into single media packed) but this does not
974 * work well with AudioFlinger and causes problems. For this reason we
975 * use magic value here and out_write code takes care of splitting
976 * input buffer into multiple media packets.
981 static uint32_t out_get_channels(const struct audio_stream *stream)
983 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
987 return out->cfg.channels;
990 static audio_format_t out_get_format(const struct audio_stream *stream)
992 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
996 return out->cfg.format;
999 static int out_set_format(struct audio_stream *stream, audio_format_t format)
1005 static int out_standby(struct audio_stream *stream)
1007 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1011 if (out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1012 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1014 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1020 static int out_dump(const struct audio_stream *stream, int fd)
1026 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
1028 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1032 bool enter_suspend = false;
1033 bool exit_suspend = false;
1037 str = strdup(kvpairs);
1038 kvpair = strtok_r(str, ";", &saveptr);
1040 for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
1043 keyval = strchr(kvpair, '=');
1050 if (!strcmp(kvpair, "closing")) {
1051 if (!strcmp(keyval, "true"))
1052 out->audio_state = AUDIO_A2DP_STATE_NONE;
1053 } else if (!strcmp(kvpair, "A2dpSuspended")) {
1054 if (!strcmp(keyval, "true"))
1055 enter_suspend = true;
1057 exit_suspend = true;
1063 if (enter_suspend && out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1064 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1066 out->audio_state = AUDIO_A2DP_STATE_SUSPENDED;
1069 if (exit_suspend && out->audio_state == AUDIO_A2DP_STATE_SUSPENDED)
1070 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1075 static char *out_get_parameters(const struct audio_stream *stream,
1082 static uint32_t out_get_latency(const struct audio_stream_out *stream)
1084 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1085 struct audio_endpoint *ep = out->ep;
1086 size_t pkt_duration;
1090 pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
1092 return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
1095 static int out_set_volume(struct audio_stream_out *stream, float left,
1099 /* volume controlled in audioflinger mixer (digital) */
1103 static int out_get_render_position(const struct audio_stream_out *stream,
1104 uint32_t *dsp_frames)
1110 static int out_add_audio_effect(const struct audio_stream *stream,
1111 effect_handle_t effect)
1117 static int out_remove_audio_effect(const struct audio_stream *stream,
1118 effect_handle_t effect)
1124 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1130 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1136 static size_t in_get_buffer_size(const struct audio_stream *stream)
1142 static uint32_t in_get_channels(const struct audio_stream *stream)
1148 static audio_format_t in_get_format(const struct audio_stream *stream)
1154 static int in_set_format(struct audio_stream *stream, audio_format_t format)
1160 static int in_standby(struct audio_stream *stream)
1166 static int in_dump(const struct audio_stream *stream, int fd)
1172 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1178 static char *in_get_parameters(const struct audio_stream *stream,
1185 static int in_set_gain(struct audio_stream_in *stream, float gain)
1191 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1198 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1204 static int in_add_audio_effect(const struct audio_stream *stream,
1205 effect_handle_t effect)
1211 static int in_remove_audio_effect(const struct audio_stream *stream,
1212 effect_handle_t effect)
1218 static int set_blocking(int fd)
1222 flags = fcntl(fd, F_GETFL, 0);
1224 error("fcntl(F_GETFL): %s (%d)", strerror(errno), errno);
1228 if (fcntl(fd, F_SETFL, flags & ~O_NONBLOCK) < 0) {
1229 error("fcntl(F_SETFL): %s (%d)", strerror(errno), errno);
1236 static int audio_open_output_stream(struct audio_hw_device *dev,
1237 audio_io_handle_t handle,
1238 audio_devices_t devices,
1239 audio_output_flags_t flags,
1240 struct audio_config *config,
1241 struct audio_stream_out **stream_out)
1244 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1245 struct a2dp_stream_out *out;
1246 struct audio_preset *preset;
1247 const struct audio_codec *codec;
1251 out = calloc(1, sizeof(struct a2dp_stream_out));
1257 out->stream.common.get_sample_rate = out_get_sample_rate;
1258 out->stream.common.set_sample_rate = out_set_sample_rate;
1259 out->stream.common.get_buffer_size = out_get_buffer_size;
1260 out->stream.common.get_channels = out_get_channels;
1261 out->stream.common.get_format = out_get_format;
1262 out->stream.common.set_format = out_set_format;
1263 out->stream.common.standby = out_standby;
1264 out->stream.common.dump = out_dump;
1265 out->stream.common.set_parameters = out_set_parameters;
1266 out->stream.common.get_parameters = out_get_parameters;
1267 out->stream.common.add_audio_effect = out_add_audio_effect;
1268 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1269 out->stream.get_latency = out_get_latency;
1270 out->stream.set_volume = out_set_volume;
1271 out->stream.write = out_write;
1272 out->stream.get_render_position = out_get_render_position;
1274 /* TODO: for now we always use endpoint 0 */
1275 out->ep = &audio_endpoints[0];
1277 if (ipc_open_stream_cmd(out->ep->id, &mtu, &fd, &preset) !=
1278 AUDIO_STATUS_SUCCESS)
1281 if (!preset || fd < 0)
1284 if (set_blocking(fd) < 0) {
1290 codec = out->ep->codec;
1292 codec->init(preset, mtu, &out->ep->codec_data);
1293 codec->get_config(out->ep->codec_data, &out->cfg);
1295 DBG("rate=%d channels=%d format=%d", out->cfg.rate,
1296 out->cfg.channels, out->cfg.format);
1300 *stream_out = &out->stream;
1301 a2dp_dev->out = out;
1303 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1308 error("audio: cannot open output stream");
1314 static void audio_close_output_stream(struct audio_hw_device *dev,
1315 struct audio_stream_out *stream)
1317 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1318 struct audio_endpoint *ep = a2dp_dev->out->ep;
1322 ipc_close_stream_cmd(ep->id);
1328 ep->codec->cleanup(ep->codec_data);
1329 ep->codec_data = NULL;
1332 a2dp_dev->out = NULL;
1335 static int audio_set_parameters(struct audio_hw_device *dev,
1336 const char *kvpairs)
1338 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1339 struct a2dp_stream_out *out = a2dp_dev->out;
1346 return out->stream.common.set_parameters((struct audio_stream *) out,
1350 static char *audio_get_parameters(const struct audio_hw_device *dev,
1357 static int audio_init_check(const struct audio_hw_device *dev)
1363 static int audio_set_voice_volume(struct audio_hw_device *dev, float volume)
1369 static int audio_set_master_volume(struct audio_hw_device *dev, float volume)
1375 static int audio_set_mode(struct audio_hw_device *dev, int mode)
1381 static int audio_set_mic_mute(struct audio_hw_device *dev, bool state)
1387 static int audio_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1393 static size_t audio_get_input_buffer_size(const struct audio_hw_device *dev,
1394 const struct audio_config *config)
1400 static int audio_open_input_stream(struct audio_hw_device *dev,
1401 audio_io_handle_t handle,
1402 audio_devices_t devices,
1403 struct audio_config *config,
1404 struct audio_stream_in **stream_in)
1406 struct audio_stream_in *in;
1410 in = calloc(1, sizeof(struct audio_stream_in));
1414 in->common.get_sample_rate = in_get_sample_rate;
1415 in->common.set_sample_rate = in_set_sample_rate;
1416 in->common.get_buffer_size = in_get_buffer_size;
1417 in->common.get_channels = in_get_channels;
1418 in->common.get_format = in_get_format;
1419 in->common.set_format = in_set_format;
1420 in->common.standby = in_standby;
1421 in->common.dump = in_dump;
1422 in->common.set_parameters = in_set_parameters;
1423 in->common.get_parameters = in_get_parameters;
1424 in->common.add_audio_effect = in_add_audio_effect;
1425 in->common.remove_audio_effect = in_remove_audio_effect;
1426 in->set_gain = in_set_gain;
1428 in->get_input_frames_lost = in_get_input_frames_lost;
1435 static void audio_close_input_stream(struct audio_hw_device *dev,
1436 struct audio_stream_in *stream_in)
1442 static int audio_dump(const audio_hw_device_t *device, int fd)
1448 static int audio_close(hw_device_t *device)
1450 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
1454 unregister_endpoints();
1456 shutdown(listen_sk, SHUT_RDWR);
1457 shutdown(audio_sk, SHUT_RDWR);
1459 pthread_join(ipc_th, NULL);
1468 static void *ipc_handler(void *data)
1477 DBG("Waiting for connection ...");
1479 sk = accept(listen_sk, NULL, NULL);
1486 if (err != ECONNABORTED && err != EINVAL)
1487 error("audio: Failed to accept socket: %d (%s)",
1488 err, strerror(err));
1493 pthread_mutex_lock(&sk_mutex);
1495 pthread_mutex_unlock(&sk_mutex);
1497 DBG("Audio IPC: Connected");
1499 if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
1500 error("audio: Failed to register endpoints");
1502 unregister_endpoints();
1504 pthread_mutex_lock(&sk_mutex);
1505 shutdown(audio_sk, SHUT_RDWR);
1508 pthread_mutex_unlock(&sk_mutex);
1513 memset(&pfd, 0, sizeof(pfd));
1515 pfd.events = POLLHUP | POLLERR | POLLNVAL;
1517 /* Check if socket is still alive. Empty while loop.*/
1518 while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
1520 if (pfd.revents & (POLLHUP | POLLERR | POLLNVAL)) {
1521 info("Audio HAL: Socket closed");
1523 pthread_mutex_lock(&sk_mutex);
1526 pthread_mutex_unlock(&sk_mutex);
1530 /* audio_sk is closed at this point, just cleanup endpoints states */
1531 memset(audio_endpoints, 0, sizeof(audio_endpoints));
1533 info("Closing Audio IPC thread");
1537 static int audio_ipc_init(void)
1539 struct sockaddr_un addr;
1545 sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
1548 error("audio: Failed to create socket: %d (%s)", err,
1553 memset(&addr, 0, sizeof(addr));
1554 addr.sun_family = AF_UNIX;
1556 memcpy(addr.sun_path, BLUEZ_AUDIO_SK_PATH,
1557 sizeof(BLUEZ_AUDIO_SK_PATH));
1559 if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
1561 error("audio: Failed to bind socket: %d (%s)", err,
1566 if (listen(sk, 1) < 0) {
1568 error("audio: Failed to listen on the socket: %d (%s)", err,
1575 err = pthread_create(&ipc_th, NULL, ipc_handler, NULL);
1579 error("audio: Failed to start Audio IPC thread: %d (%s)",
1580 err, strerror(err));
1591 static int audio_open(const hw_module_t *module, const char *name,
1592 hw_device_t **device)
1594 struct a2dp_audio_dev *a2dp_dev;
1599 if (strcmp(name, AUDIO_HARDWARE_INTERFACE)) {
1600 error("audio: interface %s not matching [%s]", name,
1601 AUDIO_HARDWARE_INTERFACE);
1605 err = audio_ipc_init();
1609 a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
1613 a2dp_dev->dev.common.tag = HARDWARE_DEVICE_TAG;
1614 a2dp_dev->dev.common.version = AUDIO_DEVICE_API_VERSION_CURRENT;
1615 a2dp_dev->dev.common.module = (struct hw_module_t *) module;
1616 a2dp_dev->dev.common.close = audio_close;
1618 a2dp_dev->dev.init_check = audio_init_check;
1619 a2dp_dev->dev.set_voice_volume = audio_set_voice_volume;
1620 a2dp_dev->dev.set_master_volume = audio_set_master_volume;
1621 a2dp_dev->dev.set_mode = audio_set_mode;
1622 a2dp_dev->dev.set_mic_mute = audio_set_mic_mute;
1623 a2dp_dev->dev.get_mic_mute = audio_get_mic_mute;
1624 a2dp_dev->dev.set_parameters = audio_set_parameters;
1625 a2dp_dev->dev.get_parameters = audio_get_parameters;
1626 a2dp_dev->dev.get_input_buffer_size = audio_get_input_buffer_size;
1627 a2dp_dev->dev.open_output_stream = audio_open_output_stream;
1628 a2dp_dev->dev.close_output_stream = audio_close_output_stream;
1629 a2dp_dev->dev.open_input_stream = audio_open_input_stream;
1630 a2dp_dev->dev.close_input_stream = audio_close_input_stream;
1631 a2dp_dev->dev.dump = audio_dump;
1633 /* Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
1634 * This results from the structure of following structs:a2dp_audio_dev,
1635 * audio_hw_device. We will rely on this later in the code.*/
1636 *device = &a2dp_dev->dev.common;
1641 static struct hw_module_methods_t hal_module_methods = {
1645 struct audio_module HAL_MODULE_INFO_SYM = {
1647 .tag = HARDWARE_MODULE_TAG,
1650 .id = AUDIO_HARDWARE_MODULE_ID,
1651 .name = "A2DP Bluez HW HAL",
1652 .author = "Intel Corporation",
1653 .methods = &hal_module_methods,