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;
140 static inline void timespec_diff(struct timespec *a, struct timespec *b,
141 struct timespec *res)
143 res->tv_sec = a->tv_sec - b->tv_sec;
144 res->tv_nsec = a->tv_nsec - b->tv_nsec;
146 if (res->tv_nsec < 0) {
148 res->tv_nsec += 1000000000; /* 1sec */
152 static int sbc_get_presets(struct audio_preset *preset, size_t *len);
153 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
155 static int sbc_cleanup(void *codec_data);
156 static int sbc_get_config(void *codec_data, struct audio_input_config *config);
157 static size_t sbc_get_buffer_size(void *codec_data);
158 static size_t sbc_get_mediapacket_duration(void *codec_data);
159 static void sbc_resume(void *codec_data);
160 static ssize_t sbc_write_data(void *codec_data, const void *buffer,
161 size_t bytes, int fd);
166 int (*get_presets) (struct audio_preset *preset, size_t *len);
168 int (*init) (struct audio_preset *preset, uint16_t mtu,
170 int (*cleanup) (void *codec_data);
171 int (*get_config) (void *codec_data,
172 struct audio_input_config *config);
173 size_t (*get_buffer_size) (void *codec_data);
174 size_t (*get_mediapacket_duration) (void *codec_data);
175 void (*resume) (void *codec_data);
176 ssize_t (*write_data) (void *codec_data, const void *buffer,
177 size_t bytes, int fd);
180 static const struct audio_codec audio_codecs[] = {
182 .type = A2DP_CODEC_SBC,
184 .get_presets = sbc_get_presets,
186 .init = sbc_codec_init,
187 .cleanup = sbc_cleanup,
188 .get_config = sbc_get_config,
189 .get_buffer_size = sbc_get_buffer_size,
190 .get_mediapacket_duration = sbc_get_mediapacket_duration,
191 .resume = sbc_resume,
192 .write_data = sbc_write_data,
196 #define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
198 #define MAX_AUDIO_ENDPOINTS NUM_CODECS
200 struct audio_endpoint {
202 const struct audio_codec *codec;
207 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
210 AUDIO_A2DP_STATE_NONE,
211 AUDIO_A2DP_STATE_STANDBY,
212 AUDIO_A2DP_STATE_SUSPENDED,
213 AUDIO_A2DP_STATE_STARTED
216 struct a2dp_stream_out {
217 struct audio_stream_out stream;
219 struct audio_endpoint *ep;
220 enum a2dp_state_t audio_state;
221 struct audio_input_config cfg;
224 struct a2dp_audio_dev {
225 struct audio_hw_device dev;
226 struct a2dp_stream_out *out;
229 static const a2dp_sbc_t sbc_presets[] = {
231 .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
232 .channel_mode = SBC_CHANNEL_MODE_DUAL_CHANNEL |
233 SBC_CHANNEL_MODE_STEREO |
234 SBC_CHANNEL_MODE_JOINT_STEREO,
235 .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
236 .allocation_method = SBC_ALLOCATION_SNR |
237 SBC_ALLOCATION_LOUDNESS,
238 .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
239 SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
240 .min_bitpool = MIN_BITPOOL,
241 .max_bitpool = MAX_BITPOOL
244 .frequency = SBC_SAMPLING_FREQ_44100,
245 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
246 .subbands = SBC_SUBBANDS_8,
247 .allocation_method = SBC_ALLOCATION_LOUDNESS,
248 .block_length = SBC_BLOCK_LENGTH_16,
249 .min_bitpool = MIN_BITPOOL,
250 .max_bitpool = MAX_BITPOOL
253 .frequency = SBC_SAMPLING_FREQ_48000,
254 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
255 .subbands = SBC_SUBBANDS_8,
256 .allocation_method = SBC_ALLOCATION_LOUDNESS,
257 .block_length = SBC_BLOCK_LENGTH_16,
258 .min_bitpool = MIN_BITPOOL,
259 .max_bitpool = MAX_BITPOOL
263 static int sbc_get_presets(struct audio_preset *preset, size_t *len)
268 uint8_t *ptr = (uint8_t *) preset;
269 size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
271 count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
273 for (i = 0; i < count; i++) {
274 preset = (struct audio_preset *) ptr;
276 if (new_len + preset_size > *len)
279 preset->len = sizeof(a2dp_sbc_t);
280 memcpy(preset->data, &sbc_presets[i], preset->len);
282 new_len += preset_size;
291 static void sbc_init_encoder(struct sbc_data *sbc_data)
293 a2dp_sbc_t *in = &sbc_data->sbc;
294 sbc_t *out = &sbc_data->enc;
296 sbc_init_a2dp(out, 0L, in, sizeof(*in));
298 out->endian = SBC_LE;
299 out->bitpool = in->max_bitpool;
302 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
305 struct sbc_data *sbc_data;
306 size_t hdr_len = sizeof(struct media_packet);
308 size_t out_frame_len;
311 if (preset->len != sizeof(a2dp_sbc_t)) {
312 error("SBC: preset size mismatch");
313 return AUDIO_STATUS_FAILED;
316 sbc_data = calloc(sizeof(struct sbc_data), 1);
318 return AUDIO_STATUS_FAILED;
320 memcpy(&sbc_data->sbc, preset->data, preset->len);
322 sbc_init_encoder(sbc_data);
324 in_frame_len = sbc_get_codesize(&sbc_data->enc);
325 out_frame_len = sbc_get_frame_length(&sbc_data->enc);
326 num_frames = (mtu - hdr_len) / out_frame_len;
328 sbc_data->in_frame_len = in_frame_len;
329 sbc_data->in_buf_size = num_frames * in_frame_len;
331 sbc_data->out_buf_size = hdr_len + num_frames * out_frame_len;
332 sbc_data->out_buf = calloc(1, sbc_data->out_buf_size);
334 sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
335 sbc_data->frames_per_packet = num_frames;
337 DBG("mtu=%u in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
338 mtu, in_frame_len, out_frame_len, num_frames);
340 *codec_data = sbc_data;
342 return AUDIO_STATUS_SUCCESS;
345 static int sbc_cleanup(void *codec_data)
347 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
349 sbc_finish(&sbc_data->enc);
350 free(sbc_data->out_buf);
353 return AUDIO_STATUS_SUCCESS;
356 static int sbc_get_config(void *codec_data, struct audio_input_config *config)
358 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
360 switch (sbc_data->sbc.frequency) {
361 case SBC_SAMPLING_FREQ_16000:
362 config->rate = 16000;
364 case SBC_SAMPLING_FREQ_32000:
365 config->rate = 32000;
367 case SBC_SAMPLING_FREQ_44100:
368 config->rate = 44100;
370 case SBC_SAMPLING_FREQ_48000:
371 config->rate = 48000;
374 return AUDIO_STATUS_FAILED;
376 config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
377 AUDIO_CHANNEL_OUT_MONO :
378 AUDIO_CHANNEL_OUT_STEREO;
379 config->format = AUDIO_FORMAT_PCM_16_BIT;
381 return AUDIO_STATUS_SUCCESS;
384 static size_t sbc_get_buffer_size(void *codec_data)
386 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
388 return sbc_data->in_buf_size;
391 static size_t sbc_get_mediapacket_duration(void *codec_data)
393 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
395 return sbc_data->frame_duration * sbc_data->frames_per_packet;
398 static void sbc_resume(void *codec_data)
400 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
404 clock_gettime(CLOCK_MONOTONIC, &sbc_data->start);
406 sbc_data->frames_sent = 0;
409 static int write_media_packet(int fd, struct sbc_data *sbc_data,
410 struct media_packet *mp, size_t data_len)
413 struct timespec diff;
414 unsigned expected_frames;
418 ret = write(fd, mp, sizeof(*mp) + data_len);
426 sbc_data->frames_sent += mp->payload.frame_count;
428 clock_gettime(CLOCK_MONOTONIC, &cur);
429 timespec_diff(&cur, &sbc_data->start, &diff);
430 expected_frames = (diff.tv_sec * 1000000 + diff.tv_nsec / 1000) /
431 sbc_data->frame_duration;
433 /* AudioFlinger does not seem to provide any *working*
434 * API to provide data in some interval and will just
435 * send another buffer as soon as we process current
436 * one. To prevent overflowing L2CAP socket, we need to
437 * introduce some artificial delay here base on how many
438 * audio frames were sent so far, i.e. if we're not
439 * lagging behind audio stream, we can sleep for
440 * duration of single media packet.
442 if (sbc_data->frames_sent >= expected_frames)
443 usleep(sbc_data->frame_duration *
444 mp->payload.frame_count);
449 static ssize_t sbc_write_data(void *codec_data, const void *buffer,
450 size_t bytes, int fd)
452 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
455 struct media_packet *mp = (struct media_packet *) sbc_data->out_buf;
456 size_t free_space = sbc_data->out_buf_size - sizeof(*mp);
461 mp->hdr.sequence_number = htons(sbc_data->seq++);
462 mp->hdr.ssrc = htonl(1);
463 mp->payload.frame_count = 0;
465 while (bytes - consumed >= sbc_data->in_frame_len) {
468 ret = sbc_encode(&sbc_data->enc, buffer + consumed,
469 sbc_data->in_frame_len,
470 mp->data + encoded, free_space,
474 error("SBC: failed to encode block (%d)", ret);
478 mp->payload.frame_count++;
482 free_space -= written;
484 /* write data if we either filled media packed or encoded all
487 if (mp->payload.frame_count == sbc_data->frames_per_packet ||
489 mp->payload.frame_count ==
490 MAX_FRAMES_IN_PAYLOAD) {
491 ret = write_media_packet(fd, sbc_data, mp, encoded);
496 free_space = sbc_data->out_buf_size - sizeof(*mp);
497 mp->payload.frame_count = 0;
501 if (consumed != bytes) {
502 /* we should encode all input data
503 * if we did not, something went wrong but we can't really
504 * handle this so this is just sanity check
506 error("SBC: failed to encode complete input buffer");
509 /* we always assume that all data was processed and sent */
513 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
514 void *param, size_t *rsp_len, void *rsp, int *fd)
520 char cmsgbuf[CMSG_SPACE(sizeof(int))];
522 size_t s_len = sizeof(s);
524 pthread_mutex_lock(&sk_mutex);
527 error("audio: Invalid cmd socket passed to audio_ipc_cmd");
531 if (!rsp || !rsp_len) {
532 memset(&s, 0, s_len);
537 memset(&msg, 0, sizeof(msg));
538 memset(&cmd, 0, sizeof(cmd));
540 cmd.service_id = service_id;
544 iv[0].iov_base = &cmd;
545 iv[0].iov_len = sizeof(cmd);
547 iv[1].iov_base = param;
553 ret = sendmsg(audio_sk, &msg, 0);
555 error("audio: Sending command failed:%s", strerror(errno));
559 /* socket was shutdown */
561 error("audio: Command socket closed");
565 memset(&msg, 0, sizeof(msg));
566 memset(&cmd, 0, sizeof(cmd));
568 iv[0].iov_base = &cmd;
569 iv[0].iov_len = sizeof(cmd);
571 iv[1].iov_base = rsp;
572 iv[1].iov_len = *rsp_len;
578 memset(cmsgbuf, 0, sizeof(cmsgbuf));
579 msg.msg_control = cmsgbuf;
580 msg.msg_controllen = sizeof(cmsgbuf);
583 ret = recvmsg(audio_sk, &msg, 0);
585 error("audio: Receiving command response failed:%s",
590 if (ret < (ssize_t) sizeof(cmd)) {
591 error("audio: Too small response received(%zd bytes)", ret);
595 if (cmd.service_id != service_id) {
596 error("audio: Invalid service id (%u vs %u)", cmd.service_id,
601 if (ret != (ssize_t) (sizeof(cmd) + cmd.len)) {
602 error("audio: Malformed response received(%zd bytes)", ret);
606 if (cmd.opcode != opcode && cmd.opcode != AUDIO_OP_STATUS) {
607 error("audio: Invalid opcode received (%u vs %u)",
612 if (cmd.opcode == AUDIO_OP_STATUS) {
613 struct hal_status *s = rsp;
615 if (sizeof(*s) != cmd.len) {
616 error("audio: Invalid status length");
620 if (s->code == AUDIO_STATUS_SUCCESS) {
621 error("audio: Invalid success status response");
625 pthread_mutex_unlock(&sk_mutex);
630 pthread_mutex_unlock(&sk_mutex);
632 /* Receive auxiliary data in msg */
634 struct cmsghdr *cmsg;
638 for (cmsg = CMSG_FIRSTHDR(&msg); cmsg;
639 cmsg = CMSG_NXTHDR(&msg, cmsg)) {
640 if (cmsg->cmsg_level == SOL_SOCKET
641 && cmsg->cmsg_type == SCM_RIGHTS) {
642 memcpy(fd, CMSG_DATA(cmsg), sizeof(int));
654 return AUDIO_STATUS_SUCCESS;
657 /* Some serious issue happen on IPC - recover */
658 shutdown(audio_sk, SHUT_RDWR);
659 pthread_mutex_unlock(&sk_mutex);
661 return AUDIO_STATUS_FAILED;
664 static int ipc_open_cmd(const struct audio_codec *codec)
666 uint8_t buf[BLUEZ_AUDIO_MTU];
667 struct audio_cmd_open *cmd = (struct audio_cmd_open *) buf;
668 struct audio_rsp_open rsp;
669 size_t cmd_len = sizeof(buf) - sizeof(*cmd);
670 size_t rsp_len = sizeof(rsp);
675 memcpy(cmd->uuid, a2dp_src_uuid, sizeof(a2dp_src_uuid));
677 cmd->codec = codec->type;
678 cmd->presets = codec->get_presets(cmd->preset, &cmd_len);
680 cmd_len += sizeof(*cmd);
682 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN, cmd_len, cmd,
683 &rsp_len, &rsp, NULL);
685 if (result != AUDIO_STATUS_SUCCESS)
691 static int ipc_close_cmd(uint8_t endpoint_id)
693 struct audio_cmd_close cmd;
698 cmd.id = endpoint_id;
700 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE,
701 sizeof(cmd), &cmd, NULL, NULL, NULL);
706 static int ipc_open_stream_cmd(uint8_t endpoint_id, uint16_t *mtu, int *fd,
707 struct audio_preset **caps)
709 char buf[BLUEZ_AUDIO_MTU];
710 struct audio_cmd_open_stream cmd;
711 struct audio_rsp_open_stream *rsp =
712 (struct audio_rsp_open_stream *) &buf;
713 size_t rsp_len = sizeof(buf);
719 return AUDIO_STATUS_FAILED;
721 cmd.id = endpoint_id;
723 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN_STREAM,
724 sizeof(cmd), &cmd, &rsp_len, rsp, fd);
725 if (result == AUDIO_STATUS_SUCCESS) {
726 size_t buf_len = sizeof(struct audio_preset) +
729 *caps = malloc(buf_len);
730 memcpy(*caps, &rsp->preset, buf_len);
738 static int ipc_close_stream_cmd(uint8_t endpoint_id)
740 struct audio_cmd_close_stream cmd;
745 cmd.id = endpoint_id;
747 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE_STREAM,
748 sizeof(cmd), &cmd, NULL, NULL, NULL);
753 static int ipc_resume_stream_cmd(uint8_t endpoint_id)
755 struct audio_cmd_resume_stream cmd;
760 cmd.id = endpoint_id;
762 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_RESUME_STREAM,
763 sizeof(cmd), &cmd, NULL, NULL, NULL);
768 static int ipc_suspend_stream_cmd(uint8_t endpoint_id)
770 struct audio_cmd_suspend_stream cmd;
775 cmd.id = endpoint_id;
777 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_SUSPEND_STREAM,
778 sizeof(cmd), &cmd, NULL, NULL, NULL);
783 static int register_endpoints(void)
785 struct audio_endpoint *ep = &audio_endpoints[0];
788 for (i = 0; i < NUM_CODECS; i++, ep++) {
789 const struct audio_codec *codec = &audio_codecs[i];
791 ep->id = ipc_open_cmd(codec);
794 return AUDIO_STATUS_FAILED;
797 ep->codec_data = NULL;
801 return AUDIO_STATUS_SUCCESS;
804 static void unregister_endpoints(void)
808 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++) {
809 struct audio_endpoint *ep = &audio_endpoints[i];
812 ipc_close_cmd(ep->id);
813 memset(ep, 0, sizeof(*ep));
818 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
821 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
823 /* We can auto-start only from standby */
824 if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
825 DBG("stream in standby, auto-start");
827 if (ipc_resume_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
830 out->ep->codec->resume(out->ep->codec_data);
832 out->audio_state = AUDIO_A2DP_STATE_STARTED;
835 if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
836 error("audio: stream not started");
840 if (out->ep->fd < 0) {
841 error("audio: no transport socket");
845 return out->ep->codec->write_data(out->ep->codec_data, buffer,
849 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
851 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
855 return out->cfg.rate;
858 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
860 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
864 if (rate != out->cfg.rate) {
865 warn("audio: cannot set sample rate to %d", rate);
872 static size_t out_get_buffer_size(const struct audio_stream *stream)
876 /* We should return proper buffer size calculated by codec (so each
877 * input buffer is encoded into single media packed) but this does not
878 * work well with AudioFlinger and causes problems. For this reason we
879 * use magic value here and out_write code takes care of splitting
880 * input buffer into multiple media packets.
885 static uint32_t out_get_channels(const struct audio_stream *stream)
887 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
891 return out->cfg.channels;
894 static audio_format_t out_get_format(const struct audio_stream *stream)
896 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
900 return out->cfg.format;
903 static int out_set_format(struct audio_stream *stream, audio_format_t format)
909 static int out_standby(struct audio_stream *stream)
911 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
915 if (out->audio_state == AUDIO_A2DP_STATE_STARTED) {
916 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
918 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
924 static int out_dump(const struct audio_stream *stream, int fd)
930 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
932 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
936 bool enter_suspend = false;
937 bool exit_suspend = false;
941 str = strdup(kvpairs);
942 kvpair = strtok_r(str, ";", &saveptr);
944 for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
947 keyval = strchr(kvpair, '=');
954 if (!strcmp(kvpair, "closing")) {
955 if (!strcmp(keyval, "true"))
956 out->audio_state = AUDIO_A2DP_STATE_NONE;
957 } else if (!strcmp(kvpair, "A2dpSuspended")) {
958 if (!strcmp(keyval, "true"))
959 enter_suspend = true;
967 if (enter_suspend && out->audio_state == AUDIO_A2DP_STATE_STARTED) {
968 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
970 out->audio_state = AUDIO_A2DP_STATE_SUSPENDED;
973 if (exit_suspend && out->audio_state == AUDIO_A2DP_STATE_SUSPENDED)
974 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
979 static char *out_get_parameters(const struct audio_stream *stream,
986 static uint32_t out_get_latency(const struct audio_stream_out *stream)
988 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
989 struct audio_endpoint *ep = out->ep;
994 pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
996 return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
999 static int out_set_volume(struct audio_stream_out *stream, float left,
1003 /* volume controlled in audioflinger mixer (digital) */
1007 static int out_get_render_position(const struct audio_stream_out *stream,
1008 uint32_t *dsp_frames)
1014 static int out_add_audio_effect(const struct audio_stream *stream,
1015 effect_handle_t effect)
1021 static int out_remove_audio_effect(const struct audio_stream *stream,
1022 effect_handle_t effect)
1028 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1034 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1040 static size_t in_get_buffer_size(const struct audio_stream *stream)
1046 static uint32_t in_get_channels(const struct audio_stream *stream)
1052 static audio_format_t in_get_format(const struct audio_stream *stream)
1058 static int in_set_format(struct audio_stream *stream, audio_format_t format)
1064 static int in_standby(struct audio_stream *stream)
1070 static int in_dump(const struct audio_stream *stream, int fd)
1076 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1082 static char *in_get_parameters(const struct audio_stream *stream,
1089 static int in_set_gain(struct audio_stream_in *stream, float gain)
1095 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1102 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1108 static int in_add_audio_effect(const struct audio_stream *stream,
1109 effect_handle_t effect)
1115 static int in_remove_audio_effect(const struct audio_stream *stream,
1116 effect_handle_t effect)
1122 static int set_blocking(int fd)
1126 flags = fcntl(fd, F_GETFL, 0);
1128 error("fcntl(F_GETFL): %s (%d)", strerror(errno), errno);
1132 if (fcntl(fd, F_SETFL, flags & ~O_NONBLOCK) < 0) {
1133 error("fcntl(F_SETFL): %s (%d)", strerror(errno), errno);
1140 static int audio_open_output_stream(struct audio_hw_device *dev,
1141 audio_io_handle_t handle,
1142 audio_devices_t devices,
1143 audio_output_flags_t flags,
1144 struct audio_config *config,
1145 struct audio_stream_out **stream_out)
1148 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1149 struct a2dp_stream_out *out;
1150 struct audio_preset *preset;
1151 const struct audio_codec *codec;
1155 out = calloc(1, sizeof(struct a2dp_stream_out));
1161 out->stream.common.get_sample_rate = out_get_sample_rate;
1162 out->stream.common.set_sample_rate = out_set_sample_rate;
1163 out->stream.common.get_buffer_size = out_get_buffer_size;
1164 out->stream.common.get_channels = out_get_channels;
1165 out->stream.common.get_format = out_get_format;
1166 out->stream.common.set_format = out_set_format;
1167 out->stream.common.standby = out_standby;
1168 out->stream.common.dump = out_dump;
1169 out->stream.common.set_parameters = out_set_parameters;
1170 out->stream.common.get_parameters = out_get_parameters;
1171 out->stream.common.add_audio_effect = out_add_audio_effect;
1172 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1173 out->stream.get_latency = out_get_latency;
1174 out->stream.set_volume = out_set_volume;
1175 out->stream.write = out_write;
1176 out->stream.get_render_position = out_get_render_position;
1178 /* TODO: for now we always use endpoint 0 */
1179 out->ep = &audio_endpoints[0];
1181 if (ipc_open_stream_cmd(out->ep->id, &mtu, &fd, &preset) !=
1182 AUDIO_STATUS_SUCCESS)
1185 if (!preset || fd < 0)
1188 if (set_blocking(fd) < 0) {
1194 codec = out->ep->codec;
1196 codec->init(preset, mtu, &out->ep->codec_data);
1197 codec->get_config(out->ep->codec_data, &out->cfg);
1199 DBG("rate=%d channels=%d format=%d", out->cfg.rate,
1200 out->cfg.channels, out->cfg.format);
1204 *stream_out = &out->stream;
1205 a2dp_dev->out = out;
1207 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1212 error("audio: cannot open output stream");
1218 static void audio_close_output_stream(struct audio_hw_device *dev,
1219 struct audio_stream_out *stream)
1221 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1222 struct audio_endpoint *ep = a2dp_dev->out->ep;
1226 ipc_close_stream_cmd(ep->id);
1232 ep->codec->cleanup(ep->codec_data);
1233 ep->codec_data = NULL;
1236 a2dp_dev->out = NULL;
1239 static int audio_set_parameters(struct audio_hw_device *dev,
1240 const char *kvpairs)
1242 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1243 struct a2dp_stream_out *out = a2dp_dev->out;
1250 return out->stream.common.set_parameters((struct audio_stream *) out,
1254 static char *audio_get_parameters(const struct audio_hw_device *dev,
1261 static int audio_init_check(const struct audio_hw_device *dev)
1267 static int audio_set_voice_volume(struct audio_hw_device *dev, float volume)
1273 static int audio_set_master_volume(struct audio_hw_device *dev, float volume)
1279 static int audio_set_mode(struct audio_hw_device *dev, int mode)
1285 static int audio_set_mic_mute(struct audio_hw_device *dev, bool state)
1291 static int audio_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1297 static size_t audio_get_input_buffer_size(const struct audio_hw_device *dev,
1298 const struct audio_config *config)
1304 static int audio_open_input_stream(struct audio_hw_device *dev,
1305 audio_io_handle_t handle,
1306 audio_devices_t devices,
1307 struct audio_config *config,
1308 struct audio_stream_in **stream_in)
1310 struct audio_stream_in *in;
1314 in = calloc(1, sizeof(struct audio_stream_in));
1318 in->common.get_sample_rate = in_get_sample_rate;
1319 in->common.set_sample_rate = in_set_sample_rate;
1320 in->common.get_buffer_size = in_get_buffer_size;
1321 in->common.get_channels = in_get_channels;
1322 in->common.get_format = in_get_format;
1323 in->common.set_format = in_set_format;
1324 in->common.standby = in_standby;
1325 in->common.dump = in_dump;
1326 in->common.set_parameters = in_set_parameters;
1327 in->common.get_parameters = in_get_parameters;
1328 in->common.add_audio_effect = in_add_audio_effect;
1329 in->common.remove_audio_effect = in_remove_audio_effect;
1330 in->set_gain = in_set_gain;
1332 in->get_input_frames_lost = in_get_input_frames_lost;
1339 static void audio_close_input_stream(struct audio_hw_device *dev,
1340 struct audio_stream_in *stream_in)
1346 static int audio_dump(const audio_hw_device_t *device, int fd)
1352 static int audio_close(hw_device_t *device)
1354 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
1358 unregister_endpoints();
1360 shutdown(listen_sk, SHUT_RDWR);
1361 shutdown(audio_sk, SHUT_RDWR);
1363 pthread_join(ipc_th, NULL);
1372 static void *ipc_handler(void *data)
1381 DBG("Waiting for connection ...");
1383 sk = accept(listen_sk, NULL, NULL);
1390 if (err != ECONNABORTED && err != EINVAL)
1391 error("audio: Failed to accept socket: %d (%s)",
1392 err, strerror(err));
1397 pthread_mutex_lock(&sk_mutex);
1399 pthread_mutex_unlock(&sk_mutex);
1401 DBG("Audio IPC: Connected");
1403 if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
1404 error("audio: Failed to register endpoints");
1406 unregister_endpoints();
1408 pthread_mutex_lock(&sk_mutex);
1409 shutdown(audio_sk, SHUT_RDWR);
1412 pthread_mutex_unlock(&sk_mutex);
1417 memset(&pfd, 0, sizeof(pfd));
1419 pfd.events = POLLHUP | POLLERR | POLLNVAL;
1421 /* Check if socket is still alive. Empty while loop.*/
1422 while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
1424 if (pfd.revents & (POLLHUP | POLLERR | POLLNVAL)) {
1425 info("Audio HAL: Socket closed");
1427 pthread_mutex_lock(&sk_mutex);
1430 pthread_mutex_unlock(&sk_mutex);
1434 /* audio_sk is closed at this point, just cleanup endpoints states */
1435 memset(audio_endpoints, 0, sizeof(audio_endpoints));
1437 info("Closing Audio IPC thread");
1441 static int audio_ipc_init(void)
1443 struct sockaddr_un addr;
1449 sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
1452 error("audio: Failed to create socket: %d (%s)", err,
1457 memset(&addr, 0, sizeof(addr));
1458 addr.sun_family = AF_UNIX;
1460 memcpy(addr.sun_path, BLUEZ_AUDIO_SK_PATH,
1461 sizeof(BLUEZ_AUDIO_SK_PATH));
1463 if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
1465 error("audio: Failed to bind socket: %d (%s)", err,
1470 if (listen(sk, 1) < 0) {
1472 error("audio: Failed to listen on the socket: %d (%s)", err,
1479 err = pthread_create(&ipc_th, NULL, ipc_handler, NULL);
1483 error("audio: Failed to start Audio IPC thread: %d (%s)",
1484 err, strerror(err));
1495 static int audio_open(const hw_module_t *module, const char *name,
1496 hw_device_t **device)
1498 struct a2dp_audio_dev *a2dp_dev;
1503 if (strcmp(name, AUDIO_HARDWARE_INTERFACE)) {
1504 error("audio: interface %s not matching [%s]", name,
1505 AUDIO_HARDWARE_INTERFACE);
1509 err = audio_ipc_init();
1513 a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
1517 a2dp_dev->dev.common.tag = HARDWARE_DEVICE_TAG;
1518 a2dp_dev->dev.common.version = AUDIO_DEVICE_API_VERSION_CURRENT;
1519 a2dp_dev->dev.common.module = (struct hw_module_t *) module;
1520 a2dp_dev->dev.common.close = audio_close;
1522 a2dp_dev->dev.init_check = audio_init_check;
1523 a2dp_dev->dev.set_voice_volume = audio_set_voice_volume;
1524 a2dp_dev->dev.set_master_volume = audio_set_master_volume;
1525 a2dp_dev->dev.set_mode = audio_set_mode;
1526 a2dp_dev->dev.set_mic_mute = audio_set_mic_mute;
1527 a2dp_dev->dev.get_mic_mute = audio_get_mic_mute;
1528 a2dp_dev->dev.set_parameters = audio_set_parameters;
1529 a2dp_dev->dev.get_parameters = audio_get_parameters;
1530 a2dp_dev->dev.get_input_buffer_size = audio_get_input_buffer_size;
1531 a2dp_dev->dev.open_output_stream = audio_open_output_stream;
1532 a2dp_dev->dev.close_output_stream = audio_close_output_stream;
1533 a2dp_dev->dev.open_input_stream = audio_open_input_stream;
1534 a2dp_dev->dev.close_input_stream = audio_close_input_stream;
1535 a2dp_dev->dev.dump = audio_dump;
1537 /* Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
1538 * This results from the structure of following structs:a2dp_audio_dev,
1539 * audio_hw_device. We will rely on this later in the code.*/
1540 *device = &a2dp_dev->dev.common;
1545 static struct hw_module_methods_t hal_module_methods = {
1549 struct audio_module HAL_MODULE_INFO_SYM = {
1551 .tag = HARDWARE_MODULE_TAG,
1554 .id = AUDIO_HARDWARE_MODULE_ID,
1555 .name = "A2DP Bluez HW HAL",
1556 .author = "Intel Corporation",
1557 .methods = &hal_module_methods,