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"
36 #include "ipc-common.h"
39 #include "../profiles/audio/a2dp-codecs.h"
40 #include "../src/shared/util.h"
42 #define FIXED_A2DP_PLAYBACK_LATENCY_MS 25
44 #define FIXED_BUFFER_SIZE (20 * 512)
46 #define MAX_FRAMES_IN_PAYLOAD 15
48 #define MAX_DELAY 100000 /* 100ms */
50 static const uint8_t a2dp_src_uuid[] = {
51 0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
52 0x80, 0x00, 0x00, 0x80, 0x5f, 0x9b, 0x34, 0xfb };
54 static int listen_sk = -1;
55 static int audio_sk = -1;
57 static pthread_t ipc_th = 0;
58 static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
60 #if __BYTE_ORDER == __LITTLE_ENDIAN
71 uint16_t sequence_number;
75 } __attribute__ ((packed));
78 unsigned frame_count:4;
80 unsigned is_last_fragment:1;
81 unsigned is_first_fragment:1;
82 unsigned is_fragmented:1;
83 } __attribute__ ((packed));
85 #elif __BYTE_ORDER == __BIG_ENDIAN
96 uint16_t sequence_number;
100 } __attribute__ ((packed));
103 unsigned is_fragmented:1;
104 unsigned is_first_fragment:1;
105 unsigned is_last_fragment:1;
107 unsigned frame_count:4;
108 } __attribute__ ((packed));
111 #error "Unknown byte order"
114 struct media_packet {
115 struct rtp_header hdr;
116 struct rtp_payload payload;
120 struct audio_input_config {
123 audio_format_t format;
134 size_t out_frame_len;
136 unsigned frame_duration;
137 unsigned frames_per_packet;
140 static void timespec_add(struct timespec *base, uint64_t time_us,
141 struct timespec *res)
143 res->tv_sec = base->tv_sec + time_us / 1000000;
144 res->tv_nsec = base->tv_nsec + (time_us % 1000000) * 1000;
146 if (res->tv_nsec >= 1000000000) {
148 res->tv_nsec -= 1000000000;
152 static void timespec_diff(struct timespec *a, struct timespec *b,
153 struct timespec *res)
155 res->tv_sec = a->tv_sec - b->tv_sec;
156 res->tv_nsec = a->tv_nsec - b->tv_nsec;
158 if (res->tv_nsec < 0) {
160 res->tv_nsec += 1000000000; /* 1sec */
164 static uint64_t timespec_diff_us(struct timespec *a, struct timespec *b)
168 timespec_diff(a, b, &res);
170 return res.tv_sec * 1000000ll + res.tv_nsec / 1000ll;
174 /* Bionic does not have clock_nanosleep() prototype in time.h even though
175 * it provides its implementation.
177 extern int clock_nanosleep(clockid_t clock_id, int flags,
178 const struct timespec *request,
179 struct timespec *remain);
182 static int sbc_get_presets(struct audio_preset *preset, size_t *len);
183 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
185 static int sbc_cleanup(void *codec_data);
186 static int sbc_get_config(void *codec_data, struct audio_input_config *config);
187 static size_t sbc_get_buffer_size(void *codec_data);
188 static size_t sbc_get_mediapacket_duration(void *codec_data);
189 static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
190 size_t len, struct media_packet *mp,
191 size_t mp_data_len, size_t *written);
196 int (*get_presets) (struct audio_preset *preset, size_t *len);
198 int (*init) (struct audio_preset *preset, uint16_t mtu,
200 int (*cleanup) (void *codec_data);
201 int (*get_config) (void *codec_data,
202 struct audio_input_config *config);
203 size_t (*get_buffer_size) (void *codec_data);
204 size_t (*get_mediapacket_duration) (void *codec_data);
205 ssize_t (*encode_mediapacket) (void *codec_data, const uint8_t *buffer,
206 size_t len, struct media_packet *mp,
207 size_t mp_data_len, size_t *written);
210 static const struct audio_codec audio_codecs[] = {
212 .type = A2DP_CODEC_SBC,
214 .get_presets = sbc_get_presets,
216 .init = sbc_codec_init,
217 .cleanup = sbc_cleanup,
218 .get_config = sbc_get_config,
219 .get_buffer_size = sbc_get_buffer_size,
220 .get_mediapacket_duration = sbc_get_mediapacket_duration,
221 .encode_mediapacket = sbc_encode_mediapacket,
225 #define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
227 #define MAX_AUDIO_ENDPOINTS NUM_CODECS
229 struct audio_endpoint {
231 const struct audio_codec *codec;
235 struct media_packet *mp;
240 struct timespec start;
245 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
248 AUDIO_A2DP_STATE_NONE,
249 AUDIO_A2DP_STATE_STANDBY,
250 AUDIO_A2DP_STATE_SUSPENDED,
251 AUDIO_A2DP_STATE_STARTED
254 struct a2dp_stream_out {
255 struct audio_stream_out stream;
257 struct audio_endpoint *ep;
258 enum a2dp_state_t audio_state;
259 struct audio_input_config cfg;
261 uint8_t *downmix_buf;
264 struct a2dp_audio_dev {
265 struct audio_hw_device dev;
266 struct a2dp_stream_out *out;
269 static const a2dp_sbc_t sbc_presets[] = {
271 .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
272 .channel_mode = SBC_CHANNEL_MODE_MONO |
273 SBC_CHANNEL_MODE_DUAL_CHANNEL |
274 SBC_CHANNEL_MODE_STEREO |
275 SBC_CHANNEL_MODE_JOINT_STEREO,
276 .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
277 .allocation_method = SBC_ALLOCATION_SNR |
278 SBC_ALLOCATION_LOUDNESS,
279 .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
280 SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
281 .min_bitpool = MIN_BITPOOL,
282 .max_bitpool = MAX_BITPOOL
285 .frequency = SBC_SAMPLING_FREQ_44100,
286 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
287 .subbands = SBC_SUBBANDS_8,
288 .allocation_method = SBC_ALLOCATION_LOUDNESS,
289 .block_length = SBC_BLOCK_LENGTH_16,
290 .min_bitpool = MIN_BITPOOL,
291 .max_bitpool = MAX_BITPOOL
294 .frequency = SBC_SAMPLING_FREQ_48000,
295 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
296 .subbands = SBC_SUBBANDS_8,
297 .allocation_method = SBC_ALLOCATION_LOUDNESS,
298 .block_length = SBC_BLOCK_LENGTH_16,
299 .min_bitpool = MIN_BITPOOL,
300 .max_bitpool = MAX_BITPOOL
304 static int sbc_get_presets(struct audio_preset *preset, size_t *len)
309 uint8_t *ptr = (uint8_t *) preset;
310 size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
312 count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
314 for (i = 0; i < count; i++) {
315 preset = (struct audio_preset *) ptr;
317 if (new_len + preset_size > *len)
320 preset->len = sizeof(a2dp_sbc_t);
321 memcpy(preset->data, &sbc_presets[i], preset->len);
323 new_len += preset_size;
332 static int sbc_freq2int(uint8_t freq)
335 case SBC_SAMPLING_FREQ_16000:
337 case SBC_SAMPLING_FREQ_32000:
339 case SBC_SAMPLING_FREQ_44100:
341 case SBC_SAMPLING_FREQ_48000:
348 static const char *sbc_mode2str(uint8_t mode)
351 case SBC_CHANNEL_MODE_MONO:
353 case SBC_CHANNEL_MODE_DUAL_CHANNEL:
354 return "DualChannel";
355 case SBC_CHANNEL_MODE_STEREO:
357 case SBC_CHANNEL_MODE_JOINT_STEREO:
358 return "JointStereo";
364 static int sbc_blocks2int(uint8_t blocks)
367 case SBC_BLOCK_LENGTH_4:
369 case SBC_BLOCK_LENGTH_8:
371 case SBC_BLOCK_LENGTH_12:
373 case SBC_BLOCK_LENGTH_16:
380 static int sbc_subbands2int(uint8_t subbands)
392 static const char *sbc_allocation2str(uint8_t allocation)
394 switch (allocation) {
395 case SBC_ALLOCATION_SNR:
397 case SBC_ALLOCATION_LOUDNESS:
404 static void sbc_init_encoder(struct sbc_data *sbc_data)
406 a2dp_sbc_t *in = &sbc_data->sbc;
407 sbc_t *out = &sbc_data->enc;
409 sbc_init_a2dp(out, 0L, in, sizeof(*in));
411 out->endian = SBC_LE;
412 out->bitpool = in->max_bitpool;
414 DBG("frequency=%d channel_mode=%s block_length=%d subbands=%d "
415 "allocation=%s bitpool=%d-%d",
416 sbc_freq2int(in->frequency),
417 sbc_mode2str(in->channel_mode),
418 sbc_blocks2int(in->block_length),
419 sbc_subbands2int(in->subbands),
420 sbc_allocation2str(in->allocation_method),
421 in->min_bitpool, in->max_bitpool);
424 static int sbc_codec_init(struct audio_preset *preset, uint16_t payload_len,
427 struct sbc_data *sbc_data;
429 size_t out_frame_len;
432 if (preset->len != sizeof(a2dp_sbc_t)) {
433 error("SBC: preset size mismatch");
434 return AUDIO_STATUS_FAILED;
437 sbc_data = calloc(sizeof(struct sbc_data), 1);
439 return AUDIO_STATUS_FAILED;
441 memcpy(&sbc_data->sbc, preset->data, preset->len);
443 sbc_init_encoder(sbc_data);
445 in_frame_len = sbc_get_codesize(&sbc_data->enc);
446 out_frame_len = sbc_get_frame_length(&sbc_data->enc);
447 num_frames = payload_len / out_frame_len;
449 sbc_data->in_frame_len = in_frame_len;
450 sbc_data->in_buf_size = num_frames * in_frame_len;
452 sbc_data->out_frame_len = out_frame_len;
454 sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
455 sbc_data->frames_per_packet = num_frames;
457 DBG("in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
458 in_frame_len, out_frame_len, num_frames);
460 *codec_data = sbc_data;
462 return AUDIO_STATUS_SUCCESS;
465 static int sbc_cleanup(void *codec_data)
467 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
469 sbc_finish(&sbc_data->enc);
472 return AUDIO_STATUS_SUCCESS;
475 static int sbc_get_config(void *codec_data, struct audio_input_config *config)
477 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
479 switch (sbc_data->sbc.frequency) {
480 case SBC_SAMPLING_FREQ_16000:
481 config->rate = 16000;
483 case SBC_SAMPLING_FREQ_32000:
484 config->rate = 32000;
486 case SBC_SAMPLING_FREQ_44100:
487 config->rate = 44100;
489 case SBC_SAMPLING_FREQ_48000:
490 config->rate = 48000;
493 return AUDIO_STATUS_FAILED;
495 config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
496 AUDIO_CHANNEL_OUT_MONO :
497 AUDIO_CHANNEL_OUT_STEREO;
498 config->format = AUDIO_FORMAT_PCM_16_BIT;
500 return AUDIO_STATUS_SUCCESS;
503 static size_t sbc_get_buffer_size(void *codec_data)
505 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
507 return sbc_data->in_buf_size;
510 static size_t sbc_get_mediapacket_duration(void *codec_data)
512 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
514 return sbc_data->frame_duration * sbc_data->frames_per_packet;
517 static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
518 size_t len, struct media_packet *mp,
519 size_t mp_data_len, size_t *written)
521 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
524 uint8_t frame_count = 0;
526 while (len - consumed >= sbc_data->in_frame_len &&
527 mp_data_len - encoded >= sbc_data->out_frame_len &&
528 frame_count < MAX_FRAMES_IN_PAYLOAD) {
532 read = sbc_encode(&sbc_data->enc, buffer + consumed,
533 sbc_data->in_frame_len, mp->data + encoded,
534 mp_data_len - encoded, &written);
537 error("SBC: failed to encode block at frame %d (%zd)",
548 mp->payload.frame_count = frame_count;
553 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
554 void *param, size_t *rsp_len, void *rsp, int *fd)
560 char cmsgbuf[CMSG_SPACE(sizeof(int))];
562 size_t s_len = sizeof(s);
564 pthread_mutex_lock(&sk_mutex);
567 error("audio: Invalid cmd socket passed to audio_ipc_cmd");
571 if (!rsp || !rsp_len) {
572 memset(&s, 0, s_len);
577 memset(&msg, 0, sizeof(msg));
578 memset(&cmd, 0, sizeof(cmd));
580 cmd.service_id = service_id;
584 iv[0].iov_base = &cmd;
585 iv[0].iov_len = sizeof(cmd);
587 iv[1].iov_base = param;
593 ret = sendmsg(audio_sk, &msg, 0);
595 error("audio: Sending command failed:%s", strerror(errno));
599 /* socket was shutdown */
601 error("audio: Command socket closed");
605 memset(&msg, 0, sizeof(msg));
606 memset(&cmd, 0, sizeof(cmd));
608 iv[0].iov_base = &cmd;
609 iv[0].iov_len = sizeof(cmd);
611 iv[1].iov_base = rsp;
612 iv[1].iov_len = *rsp_len;
618 memset(cmsgbuf, 0, sizeof(cmsgbuf));
619 msg.msg_control = cmsgbuf;
620 msg.msg_controllen = sizeof(cmsgbuf);
623 ret = recvmsg(audio_sk, &msg, 0);
625 error("audio: Receiving command response failed:%s",
630 if (ret < (ssize_t) sizeof(cmd)) {
631 error("audio: Too small response received(%zd bytes)", ret);
635 if (cmd.service_id != service_id) {
636 error("audio: Invalid service id (%u vs %u)", cmd.service_id,
641 if (ret != (ssize_t) (sizeof(cmd) + cmd.len)) {
642 error("audio: Malformed response received(%zd bytes)", ret);
646 if (cmd.opcode != opcode && cmd.opcode != AUDIO_OP_STATUS) {
647 error("audio: Invalid opcode received (%u vs %u)",
652 if (cmd.opcode == AUDIO_OP_STATUS) {
653 struct ipc_status *s = rsp;
655 if (sizeof(*s) != cmd.len) {
656 error("audio: Invalid status length");
660 if (s->code == AUDIO_STATUS_SUCCESS) {
661 error("audio: Invalid success status response");
665 pthread_mutex_unlock(&sk_mutex);
670 pthread_mutex_unlock(&sk_mutex);
672 /* Receive auxiliary data in msg */
674 struct cmsghdr *cmsg;
678 for (cmsg = CMSG_FIRSTHDR(&msg); cmsg;
679 cmsg = CMSG_NXTHDR(&msg, cmsg)) {
680 if (cmsg->cmsg_level == SOL_SOCKET
681 && cmsg->cmsg_type == SCM_RIGHTS) {
682 memcpy(fd, CMSG_DATA(cmsg), sizeof(int));
694 return AUDIO_STATUS_SUCCESS;
697 /* Some serious issue happen on IPC - recover */
698 shutdown(audio_sk, SHUT_RDWR);
699 pthread_mutex_unlock(&sk_mutex);
701 return AUDIO_STATUS_FAILED;
704 static int ipc_open_cmd(const struct audio_codec *codec)
706 uint8_t buf[BLUEZ_AUDIO_MTU];
707 struct audio_cmd_open *cmd = (struct audio_cmd_open *) buf;
708 struct audio_rsp_open rsp;
709 size_t cmd_len = sizeof(buf) - sizeof(*cmd);
710 size_t rsp_len = sizeof(rsp);
715 memcpy(cmd->uuid, a2dp_src_uuid, sizeof(a2dp_src_uuid));
717 cmd->codec = codec->type;
718 cmd->presets = codec->get_presets(cmd->preset, &cmd_len);
720 cmd_len += sizeof(*cmd);
722 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN, cmd_len, cmd,
723 &rsp_len, &rsp, NULL);
725 if (result != AUDIO_STATUS_SUCCESS)
731 static int ipc_close_cmd(uint8_t endpoint_id)
733 struct audio_cmd_close cmd;
738 cmd.id = endpoint_id;
740 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE,
741 sizeof(cmd), &cmd, NULL, NULL, NULL);
746 static int ipc_open_stream_cmd(uint8_t endpoint_id, uint16_t *mtu, int *fd,
747 struct audio_preset **caps)
749 char buf[BLUEZ_AUDIO_MTU];
750 struct audio_cmd_open_stream cmd;
751 struct audio_rsp_open_stream *rsp =
752 (struct audio_rsp_open_stream *) &buf;
753 size_t rsp_len = sizeof(buf);
759 return AUDIO_STATUS_FAILED;
761 cmd.id = endpoint_id;
763 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN_STREAM,
764 sizeof(cmd), &cmd, &rsp_len, rsp, fd);
765 if (result == AUDIO_STATUS_SUCCESS) {
766 size_t buf_len = sizeof(struct audio_preset) +
769 *caps = malloc(buf_len);
770 memcpy(*caps, &rsp->preset, buf_len);
778 static int ipc_close_stream_cmd(uint8_t endpoint_id)
780 struct audio_cmd_close_stream cmd;
785 cmd.id = endpoint_id;
787 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE_STREAM,
788 sizeof(cmd), &cmd, NULL, NULL, NULL);
793 static int ipc_resume_stream_cmd(uint8_t endpoint_id)
795 struct audio_cmd_resume_stream cmd;
800 cmd.id = endpoint_id;
802 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_RESUME_STREAM,
803 sizeof(cmd), &cmd, NULL, NULL, NULL);
808 static int ipc_suspend_stream_cmd(uint8_t endpoint_id)
810 struct audio_cmd_suspend_stream cmd;
815 cmd.id = endpoint_id;
817 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_SUSPEND_STREAM,
818 sizeof(cmd), &cmd, NULL, NULL, NULL);
823 static int register_endpoints(void)
825 struct audio_endpoint *ep = &audio_endpoints[0];
828 for (i = 0; i < NUM_CODECS; i++, ep++) {
829 const struct audio_codec *codec = &audio_codecs[i];
831 ep->id = ipc_open_cmd(codec);
834 return AUDIO_STATUS_FAILED;
837 ep->codec_data = NULL;
841 return AUDIO_STATUS_SUCCESS;
844 static void unregister_endpoints(void)
848 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++) {
849 struct audio_endpoint *ep = &audio_endpoints[i];
852 ipc_close_cmd(ep->id);
853 memset(ep, 0, sizeof(*ep));
858 static bool open_endpoint(struct audio_endpoint *ep,
859 struct audio_input_config *cfg)
861 struct audio_preset *preset;
862 const struct audio_codec *codec;
864 uint16_t payload_len;
867 if (ipc_open_stream_cmd(ep->id, &mtu, &fd, &preset) !=
868 AUDIO_STATUS_SUCCESS)
873 payload_len = mtu - sizeof(*ep->mp);
878 codec->init(preset, payload_len, &ep->codec_data);
879 codec->get_config(ep->codec_data, cfg);
881 ep->mp = calloc(mtu, 1);
886 ep->mp->hdr.ssrc = htonl(1);
888 ep->mp_data_len = payload_len;
901 static void close_endpoint(struct audio_endpoint *ep)
903 ipc_close_stream_cmd(ep->id);
911 ep->codec->cleanup(ep->codec_data);
912 ep->codec_data = NULL;
915 static bool resume_endpoint(struct audio_endpoint *ep)
917 if (ipc_resume_stream_cmd(ep->id) != AUDIO_STATUS_SUCCESS)
926 static void downmix_to_mono(struct a2dp_stream_out *out, const uint8_t *buffer,
929 const int16_t *input = (const void *) buffer;
930 int16_t *output = (void *) out->downmix_buf;
933 for (i = 0; i < bytes / 2; i++) {
934 int16_t l = le16_to_cpu(get_unaligned(&input[i * 2]));
935 int16_t r = le16_to_cpu(get_unaligned(&input[i * 2 + 1]));
937 put_unaligned(cpu_to_le16((l + r) / 2), &output[i]);
941 static bool wait_for_endpoint(struct audio_endpoint *ep, bool *writable)
946 struct pollfd pollfd;
949 pollfd.events = POLLOUT;
952 ret = poll(&pollfd, 1, 500);
955 *writable = !!(pollfd.revents & POLLOUT);
959 if (errno != EINTR) {
961 error("poll failed (%d)", ret);
969 static bool write_to_endpoint(struct audio_endpoint *ep, size_t bytes)
971 struct media_packet *mp = (struct media_packet *) ep->mp;
975 ret = write(ep->fd, mp, sizeof(*mp) + bytes);
980 /* this should not happen so let's issue warning, but do not
981 * fail, we can try to write next packet
983 if (errno == EAGAIN) {
985 warn("write failed (%d)", ret);
989 if (errno != EINTR) {
991 error("write failed (%d)", ret);
999 static bool write_data(struct a2dp_stream_out *out, const void *buffer,
1002 struct audio_endpoint *ep = out->ep;
1003 struct media_packet *mp = (struct media_packet *) ep->mp;
1004 size_t free_space = ep->mp_data_len;
1005 size_t consumed = 0;
1007 while (consumed < bytes) {
1012 struct timespec current;
1013 uint64_t audio_sent, audio_passed;
1014 bool do_write = false;
1016 /* prepare media packet in advance so we don't waste time after
1019 mp->hdr.sequence_number = htons(ep->seq++);
1020 mp->hdr.timestamp = htonl(ep->samples);
1021 read = ep->codec->encode_mediapacket(ep->codec_data,
1023 bytes - consumed, mp,
1024 free_space, &written);
1026 /* not much we can do here, let's just ignore remaining
1032 /* calculate where are we and where we should be */
1033 clock_gettime(CLOCK_MONOTONIC, ¤t);
1035 memcpy(&ep->start, ¤t, sizeof(ep->start));
1036 audio_sent = ep->samples * 1000000ll / out->cfg.rate;
1037 audio_passed = timespec_diff_us(¤t, &ep->start);
1039 /* if we're ahead of stream then wait for next write point
1040 * if we're lagging more than 100ms then stop writing and just
1041 * skip data until we're back in sync
1043 if (audio_sent > audio_passed) {
1044 struct timespec anchor;
1048 timespec_add(&ep->start, audio_sent, &anchor);
1051 ret = clock_nanosleep(CLOCK_MONOTONIC,
1052 TIMER_ABSTIME, &anchor,
1059 error("clock_nanosleep failed (%d)",
1064 } else if (!ep->resync) {
1065 uint64_t diff = audio_passed - audio_sent;
1067 if (diff > MAX_DELAY) {
1068 warn("lag is %jums, resyncing", diff / 1000);
1073 /* in resync mode we'll just drop mediapackets */
1075 /* wait some time for socket to be ready for write,
1076 * but we'll just skip writing data if timeout occurs
1078 if (!wait_for_endpoint(ep, &do_write))
1082 if (!write_to_endpoint(ep, written))
1086 /* AudioFlinger provides 16bit PCM, so sample size is 2 bytes
1087 * multiplied by number of channels. Number of channels is
1088 * simply number of bits set in channels mask.
1090 samples = read / (2 * popcount(out->cfg.channels));
1091 ep->samples += samples;
1098 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
1101 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1102 const void *in_buf = buffer;
1103 size_t in_len = bytes;
1105 /* just return in case we're closing */
1106 if (out->audio_state == AUDIO_A2DP_STATE_NONE)
1109 /* We can auto-start only from standby */
1110 if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
1111 DBG("stream in standby, auto-start");
1113 if (!resume_endpoint(out->ep))
1116 out->audio_state = AUDIO_A2DP_STATE_STARTED;
1119 if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
1120 error("audio: stream not started");
1124 if (out->ep->fd < 0) {
1125 error("audio: no transport socket");
1129 /* currently Android audioflinger is not able to provide mono stream on
1130 * A2DP output so down mixing needs to be done in hal-audio plugin.
1133 * AudioFlinger::PlaybackThread::readOutputParameters()
1134 * frameworks/av/services/audioflinger/Threads.cpp:1631
1136 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1137 if (!out->downmix_buf) {
1138 error("audio: downmix buffer not initialized");
1142 downmix_to_mono(out, buffer, bytes);
1144 in_buf = out->downmix_buf;
1148 if (!write_data(out, in_buf, in_len))
1154 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
1156 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1160 return out->cfg.rate;
1163 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1165 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1169 if (rate != out->cfg.rate) {
1170 warn("audio: cannot set sample rate to %d", rate);
1177 static size_t out_get_buffer_size(const struct audio_stream *stream)
1181 /* We should return proper buffer size calculated by codec (so each
1182 * input buffer is encoded into single media packed) but this does not
1183 * work well with AudioFlinger and causes problems. For this reason we
1184 * use magic value here and out_write code takes care of splitting
1185 * input buffer into multiple media packets.
1187 return FIXED_BUFFER_SIZE;
1190 static uint32_t out_get_channels(const struct audio_stream *stream)
1194 /* AudioFlinger can only provide stereo stream, so we return it here and
1195 * later we'll downmix this to mono in case codec requires it
1198 return AUDIO_CHANNEL_OUT_STEREO;
1201 static audio_format_t out_get_format(const struct audio_stream *stream)
1203 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1207 return out->cfg.format;
1210 static int out_set_format(struct audio_stream *stream, audio_format_t format)
1216 static int out_standby(struct audio_stream *stream)
1218 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1222 if (out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1223 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1225 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1231 static int out_dump(const struct audio_stream *stream, int fd)
1237 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
1239 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1243 bool enter_suspend = false;
1244 bool exit_suspend = false;
1248 str = strdup(kvpairs);
1252 kvpair = strtok_r(str, ";", &saveptr);
1254 for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
1257 keyval = strchr(kvpair, '=');
1264 if (!strcmp(kvpair, "closing")) {
1265 if (!strcmp(keyval, "true"))
1266 out->audio_state = AUDIO_A2DP_STATE_NONE;
1267 } else if (!strcmp(kvpair, "A2dpSuspended")) {
1268 if (!strcmp(keyval, "true"))
1269 enter_suspend = true;
1271 exit_suspend = true;
1277 if (enter_suspend && out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1278 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1280 out->audio_state = AUDIO_A2DP_STATE_SUSPENDED;
1283 if (exit_suspend && out->audio_state == AUDIO_A2DP_STATE_SUSPENDED)
1284 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1289 static char *out_get_parameters(const struct audio_stream *stream,
1296 static uint32_t out_get_latency(const struct audio_stream_out *stream)
1298 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1299 struct audio_endpoint *ep = out->ep;
1300 size_t pkt_duration;
1304 pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
1306 return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
1309 static int out_set_volume(struct audio_stream_out *stream, float left,
1313 /* volume controlled in audioflinger mixer (digital) */
1317 static int out_get_render_position(const struct audio_stream_out *stream,
1318 uint32_t *dsp_frames)
1324 static int out_add_audio_effect(const struct audio_stream *stream,
1325 effect_handle_t effect)
1331 static int out_remove_audio_effect(const struct audio_stream *stream,
1332 effect_handle_t effect)
1338 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1344 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1350 static size_t in_get_buffer_size(const struct audio_stream *stream)
1356 static uint32_t in_get_channels(const struct audio_stream *stream)
1362 static audio_format_t in_get_format(const struct audio_stream *stream)
1368 static int in_set_format(struct audio_stream *stream, audio_format_t format)
1374 static int in_standby(struct audio_stream *stream)
1380 static int in_dump(const struct audio_stream *stream, int fd)
1386 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1392 static char *in_get_parameters(const struct audio_stream *stream,
1399 static int in_set_gain(struct audio_stream_in *stream, float gain)
1405 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1412 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1418 static int in_add_audio_effect(const struct audio_stream *stream,
1419 effect_handle_t effect)
1425 static int in_remove_audio_effect(const struct audio_stream *stream,
1426 effect_handle_t effect)
1432 static int audio_open_output_stream(struct audio_hw_device *dev,
1433 audio_io_handle_t handle,
1434 audio_devices_t devices,
1435 audio_output_flags_t flags,
1436 struct audio_config *config,
1437 struct audio_stream_out **stream_out)
1440 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1441 struct a2dp_stream_out *out;
1443 out = calloc(1, sizeof(struct a2dp_stream_out));
1449 out->stream.common.get_sample_rate = out_get_sample_rate;
1450 out->stream.common.set_sample_rate = out_set_sample_rate;
1451 out->stream.common.get_buffer_size = out_get_buffer_size;
1452 out->stream.common.get_channels = out_get_channels;
1453 out->stream.common.get_format = out_get_format;
1454 out->stream.common.set_format = out_set_format;
1455 out->stream.common.standby = out_standby;
1456 out->stream.common.dump = out_dump;
1457 out->stream.common.set_parameters = out_set_parameters;
1458 out->stream.common.get_parameters = out_get_parameters;
1459 out->stream.common.add_audio_effect = out_add_audio_effect;
1460 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1461 out->stream.get_latency = out_get_latency;
1462 out->stream.set_volume = out_set_volume;
1463 out->stream.write = out_write;
1464 out->stream.get_render_position = out_get_render_position;
1466 /* TODO: for now we always use endpoint 0 */
1467 out->ep = &audio_endpoints[0];
1469 if (!open_endpoint(out->ep, &out->cfg))
1472 DBG("rate=%d channels=%d format=%d", out->cfg.rate,
1473 out->cfg.channels, out->cfg.format);
1475 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1476 out->downmix_buf = malloc(FIXED_BUFFER_SIZE / 2);
1477 if (!out->downmix_buf)
1481 *stream_out = &out->stream;
1482 a2dp_dev->out = out;
1484 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1489 error("audio: cannot open output stream");
1495 static void audio_close_output_stream(struct audio_hw_device *dev,
1496 struct audio_stream_out *stream)
1498 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1499 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1503 close_endpoint(a2dp_dev->out->ep);
1505 free(out->downmix_buf);
1508 a2dp_dev->out = NULL;
1511 static int audio_set_parameters(struct audio_hw_device *dev,
1512 const char *kvpairs)
1514 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1515 struct a2dp_stream_out *out = a2dp_dev->out;
1522 return out->stream.common.set_parameters((struct audio_stream *) out,
1526 static char *audio_get_parameters(const struct audio_hw_device *dev,
1533 static int audio_init_check(const struct audio_hw_device *dev)
1539 static int audio_set_voice_volume(struct audio_hw_device *dev, float volume)
1545 static int audio_set_master_volume(struct audio_hw_device *dev, float volume)
1551 static int audio_set_mode(struct audio_hw_device *dev, int mode)
1557 static int audio_set_mic_mute(struct audio_hw_device *dev, bool state)
1563 static int audio_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1569 static size_t audio_get_input_buffer_size(const struct audio_hw_device *dev,
1570 const struct audio_config *config)
1576 static int audio_open_input_stream(struct audio_hw_device *dev,
1577 audio_io_handle_t handle,
1578 audio_devices_t devices,
1579 struct audio_config *config,
1580 struct audio_stream_in **stream_in)
1582 struct audio_stream_in *in;
1586 in = calloc(1, sizeof(struct audio_stream_in));
1590 in->common.get_sample_rate = in_get_sample_rate;
1591 in->common.set_sample_rate = in_set_sample_rate;
1592 in->common.get_buffer_size = in_get_buffer_size;
1593 in->common.get_channels = in_get_channels;
1594 in->common.get_format = in_get_format;
1595 in->common.set_format = in_set_format;
1596 in->common.standby = in_standby;
1597 in->common.dump = in_dump;
1598 in->common.set_parameters = in_set_parameters;
1599 in->common.get_parameters = in_get_parameters;
1600 in->common.add_audio_effect = in_add_audio_effect;
1601 in->common.remove_audio_effect = in_remove_audio_effect;
1602 in->set_gain = in_set_gain;
1604 in->get_input_frames_lost = in_get_input_frames_lost;
1611 static void audio_close_input_stream(struct audio_hw_device *dev,
1612 struct audio_stream_in *stream_in)
1618 static int audio_dump(const audio_hw_device_t *device, int fd)
1624 static int audio_close(hw_device_t *device)
1626 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
1630 unregister_endpoints();
1632 shutdown(listen_sk, SHUT_RDWR);
1633 shutdown(audio_sk, SHUT_RDWR);
1635 pthread_join(ipc_th, NULL);
1644 static void *ipc_handler(void *data)
1653 DBG("Waiting for connection ...");
1655 sk = accept(listen_sk, NULL, NULL);
1662 if (err != ECONNABORTED && err != EINVAL)
1663 error("audio: Failed to accept socket: %d (%s)",
1664 err, strerror(err));
1669 pthread_mutex_lock(&sk_mutex);
1671 pthread_mutex_unlock(&sk_mutex);
1673 DBG("Audio IPC: Connected");
1675 if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
1676 error("audio: Failed to register endpoints");
1678 unregister_endpoints();
1680 pthread_mutex_lock(&sk_mutex);
1681 shutdown(audio_sk, SHUT_RDWR);
1684 pthread_mutex_unlock(&sk_mutex);
1689 memset(&pfd, 0, sizeof(pfd));
1691 pfd.events = POLLHUP | POLLERR | POLLNVAL;
1693 /* Check if socket is still alive. Empty while loop.*/
1694 while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
1696 if (pfd.revents & (POLLHUP | POLLERR | POLLNVAL)) {
1697 info("Audio HAL: Socket closed");
1699 pthread_mutex_lock(&sk_mutex);
1702 pthread_mutex_unlock(&sk_mutex);
1706 /* audio_sk is closed at this point, just cleanup endpoints states */
1707 memset(audio_endpoints, 0, sizeof(audio_endpoints));
1709 info("Closing Audio IPC thread");
1713 static int audio_ipc_init(void)
1715 struct sockaddr_un addr;
1721 sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
1724 error("audio: Failed to create socket: %d (%s)", -err,
1729 memset(&addr, 0, sizeof(addr));
1730 addr.sun_family = AF_UNIX;
1732 memcpy(addr.sun_path, BLUEZ_AUDIO_SK_PATH,
1733 sizeof(BLUEZ_AUDIO_SK_PATH));
1735 if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
1737 error("audio: Failed to bind socket: %d (%s)", -err,
1742 if (listen(sk, 1) < 0) {
1744 error("audio: Failed to listen on the socket: %d (%s)", -err,
1751 err = pthread_create(&ipc_th, NULL, ipc_handler, NULL);
1755 error("audio: Failed to start Audio IPC thread: %d (%s)",
1756 -err, strerror(-err));
1767 static int audio_open(const hw_module_t *module, const char *name,
1768 hw_device_t **device)
1770 struct a2dp_audio_dev *a2dp_dev;
1775 if (strcmp(name, AUDIO_HARDWARE_INTERFACE)) {
1776 error("audio: interface %s not matching [%s]", name,
1777 AUDIO_HARDWARE_INTERFACE);
1781 err = audio_ipc_init();
1785 a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
1789 a2dp_dev->dev.common.tag = HARDWARE_DEVICE_TAG;
1790 a2dp_dev->dev.common.version = AUDIO_DEVICE_API_VERSION_CURRENT;
1791 a2dp_dev->dev.common.module = (struct hw_module_t *) module;
1792 a2dp_dev->dev.common.close = audio_close;
1794 a2dp_dev->dev.init_check = audio_init_check;
1795 a2dp_dev->dev.set_voice_volume = audio_set_voice_volume;
1796 a2dp_dev->dev.set_master_volume = audio_set_master_volume;
1797 a2dp_dev->dev.set_mode = audio_set_mode;
1798 a2dp_dev->dev.set_mic_mute = audio_set_mic_mute;
1799 a2dp_dev->dev.get_mic_mute = audio_get_mic_mute;
1800 a2dp_dev->dev.set_parameters = audio_set_parameters;
1801 a2dp_dev->dev.get_parameters = audio_get_parameters;
1802 a2dp_dev->dev.get_input_buffer_size = audio_get_input_buffer_size;
1803 a2dp_dev->dev.open_output_stream = audio_open_output_stream;
1804 a2dp_dev->dev.close_output_stream = audio_close_output_stream;
1805 a2dp_dev->dev.open_input_stream = audio_open_input_stream;
1806 a2dp_dev->dev.close_input_stream = audio_close_input_stream;
1807 a2dp_dev->dev.dump = audio_dump;
1809 /* Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
1810 * This results from the structure of following structs:a2dp_audio_dev,
1811 * audio_hw_device. We will rely on this later in the code.*/
1812 *device = &a2dp_dev->dev.common;
1817 static struct hw_module_methods_t hal_module_methods = {
1821 struct audio_module HAL_MODULE_INFO_SYM = {
1823 .tag = HARDWARE_MODULE_TAG,
1826 .id = AUDIO_HARDWARE_MODULE_ID,
1827 .name = "A2DP Bluez HW HAL",
1828 .author = "Intel Corporation",
1829 .methods = &hal_module_methods,