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 #define SBC_QUALITY_MIN_BITPOOL 33
51 #define SBC_QUALITY_STEP 5
53 static const uint8_t a2dp_src_uuid[] = {
54 0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
55 0x80, 0x00, 0x00, 0x80, 0x5f, 0x9b, 0x34, 0xfb };
57 static int listen_sk = -1;
58 static int audio_sk = -1;
60 static pthread_t ipc_th = 0;
61 static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
63 #if __BYTE_ORDER == __LITTLE_ENDIAN
74 uint16_t sequence_number;
78 } __attribute__ ((packed));
81 unsigned frame_count:4;
83 unsigned is_last_fragment:1;
84 unsigned is_first_fragment:1;
85 unsigned is_fragmented:1;
86 } __attribute__ ((packed));
88 #elif __BYTE_ORDER == __BIG_ENDIAN
99 uint16_t sequence_number;
103 } __attribute__ ((packed));
106 unsigned is_fragmented:1;
107 unsigned is_first_fragment:1;
108 unsigned is_last_fragment:1;
110 unsigned frame_count:4;
111 } __attribute__ ((packed));
114 #error "Unknown byte order"
117 struct media_packet {
118 struct rtp_header hdr;
119 struct rtp_payload payload;
123 struct audio_input_config {
126 audio_format_t format;
134 uint16_t payload_len;
139 size_t out_frame_len;
141 unsigned frame_duration;
142 unsigned frames_per_packet;
145 static void timespec_add(struct timespec *base, uint64_t time_us,
146 struct timespec *res)
148 res->tv_sec = base->tv_sec + time_us / 1000000;
149 res->tv_nsec = base->tv_nsec + (time_us % 1000000) * 1000;
151 if (res->tv_nsec >= 1000000000) {
153 res->tv_nsec -= 1000000000;
157 static void timespec_diff(struct timespec *a, struct timespec *b,
158 struct timespec *res)
160 res->tv_sec = a->tv_sec - b->tv_sec;
161 res->tv_nsec = a->tv_nsec - b->tv_nsec;
163 if (res->tv_nsec < 0) {
165 res->tv_nsec += 1000000000; /* 1sec */
169 static uint64_t timespec_diff_us(struct timespec *a, struct timespec *b)
173 timespec_diff(a, b, &res);
175 return res.tv_sec * 1000000ll + res.tv_nsec / 1000ll;
179 /* Bionic does not have clock_nanosleep() prototype in time.h even though
180 * it provides its implementation.
182 extern int clock_nanosleep(clockid_t clock_id, int flags,
183 const struct timespec *request,
184 struct timespec *remain);
187 static int sbc_get_presets(struct audio_preset *preset, size_t *len);
188 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
190 static int sbc_cleanup(void *codec_data);
191 static int sbc_get_config(void *codec_data, struct audio_input_config *config);
192 static size_t sbc_get_buffer_size(void *codec_data);
193 static size_t sbc_get_mediapacket_duration(void *codec_data);
194 static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
195 size_t len, struct media_packet *mp,
196 size_t mp_data_len, size_t *written);
197 static bool sbc_update_qos(void *codec_data, uint8_t op);
199 #define QOS_POLICY_DEFAULT 0x00
200 #define QOS_POLICY_DECREASE 0x01
205 int (*get_presets) (struct audio_preset *preset, size_t *len);
207 int (*init) (struct audio_preset *preset, uint16_t mtu,
209 int (*cleanup) (void *codec_data);
210 int (*get_config) (void *codec_data,
211 struct audio_input_config *config);
212 size_t (*get_buffer_size) (void *codec_data);
213 size_t (*get_mediapacket_duration) (void *codec_data);
214 ssize_t (*encode_mediapacket) (void *codec_data, const uint8_t *buffer,
215 size_t len, struct media_packet *mp,
216 size_t mp_data_len, size_t *written);
217 bool (*update_qos) (void *codec_data, uint8_t op);
220 static const struct audio_codec audio_codecs[] = {
222 .type = A2DP_CODEC_SBC,
224 .get_presets = sbc_get_presets,
226 .init = sbc_codec_init,
227 .cleanup = sbc_cleanup,
228 .get_config = sbc_get_config,
229 .get_buffer_size = sbc_get_buffer_size,
230 .get_mediapacket_duration = sbc_get_mediapacket_duration,
231 .encode_mediapacket = sbc_encode_mediapacket,
232 .update_qos = sbc_update_qos,
236 #define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
238 #define MAX_AUDIO_ENDPOINTS NUM_CODECS
240 struct audio_endpoint {
242 const struct audio_codec *codec;
246 struct media_packet *mp;
251 struct timespec start;
256 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
259 AUDIO_A2DP_STATE_NONE,
260 AUDIO_A2DP_STATE_STANDBY,
261 AUDIO_A2DP_STATE_SUSPENDED,
262 AUDIO_A2DP_STATE_STARTED
265 struct a2dp_stream_out {
266 struct audio_stream_out stream;
268 struct audio_endpoint *ep;
269 enum a2dp_state_t audio_state;
270 struct audio_input_config cfg;
272 uint8_t *downmix_buf;
275 struct a2dp_audio_dev {
276 struct audio_hw_device dev;
277 struct a2dp_stream_out *out;
280 static const a2dp_sbc_t sbc_presets[] = {
282 .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
283 .channel_mode = SBC_CHANNEL_MODE_MONO |
284 SBC_CHANNEL_MODE_DUAL_CHANNEL |
285 SBC_CHANNEL_MODE_STEREO |
286 SBC_CHANNEL_MODE_JOINT_STEREO,
287 .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
288 .allocation_method = SBC_ALLOCATION_SNR |
289 SBC_ALLOCATION_LOUDNESS,
290 .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
291 SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
292 .min_bitpool = MIN_BITPOOL,
293 .max_bitpool = MAX_BITPOOL
296 .frequency = SBC_SAMPLING_FREQ_44100,
297 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
298 .subbands = SBC_SUBBANDS_8,
299 .allocation_method = SBC_ALLOCATION_LOUDNESS,
300 .block_length = SBC_BLOCK_LENGTH_16,
301 .min_bitpool = MIN_BITPOOL,
302 .max_bitpool = MAX_BITPOOL
305 .frequency = SBC_SAMPLING_FREQ_48000,
306 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
307 .subbands = SBC_SUBBANDS_8,
308 .allocation_method = SBC_ALLOCATION_LOUDNESS,
309 .block_length = SBC_BLOCK_LENGTH_16,
310 .min_bitpool = MIN_BITPOOL,
311 .max_bitpool = MAX_BITPOOL
315 static int sbc_get_presets(struct audio_preset *preset, size_t *len)
320 uint8_t *ptr = (uint8_t *) preset;
321 size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
323 count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
325 for (i = 0; i < count; i++) {
326 preset = (struct audio_preset *) ptr;
328 if (new_len + preset_size > *len)
331 preset->len = sizeof(a2dp_sbc_t);
332 memcpy(preset->data, &sbc_presets[i], preset->len);
334 new_len += preset_size;
343 static int sbc_freq2int(uint8_t freq)
346 case SBC_SAMPLING_FREQ_16000:
348 case SBC_SAMPLING_FREQ_32000:
350 case SBC_SAMPLING_FREQ_44100:
352 case SBC_SAMPLING_FREQ_48000:
359 static const char *sbc_mode2str(uint8_t mode)
362 case SBC_CHANNEL_MODE_MONO:
364 case SBC_CHANNEL_MODE_DUAL_CHANNEL:
365 return "DualChannel";
366 case SBC_CHANNEL_MODE_STEREO:
368 case SBC_CHANNEL_MODE_JOINT_STEREO:
369 return "JointStereo";
375 static int sbc_blocks2int(uint8_t blocks)
378 case SBC_BLOCK_LENGTH_4:
380 case SBC_BLOCK_LENGTH_8:
382 case SBC_BLOCK_LENGTH_12:
384 case SBC_BLOCK_LENGTH_16:
391 static int sbc_subbands2int(uint8_t subbands)
403 static const char *sbc_allocation2str(uint8_t allocation)
405 switch (allocation) {
406 case SBC_ALLOCATION_SNR:
408 case SBC_ALLOCATION_LOUDNESS:
415 static void sbc_init_encoder(struct sbc_data *sbc_data)
417 a2dp_sbc_t *in = &sbc_data->sbc;
418 sbc_t *out = &sbc_data->enc;
420 sbc_init_a2dp(out, 0L, in, sizeof(*in));
422 out->endian = SBC_LE;
423 out->bitpool = in->max_bitpool;
425 DBG("frequency=%d channel_mode=%s block_length=%d subbands=%d "
426 "allocation=%s bitpool=%d-%d",
427 sbc_freq2int(in->frequency),
428 sbc_mode2str(in->channel_mode),
429 sbc_blocks2int(in->block_length),
430 sbc_subbands2int(in->subbands),
431 sbc_allocation2str(in->allocation_method),
432 in->min_bitpool, in->max_bitpool);
435 static void sbc_codec_calculate(struct sbc_data *sbc_data)
438 size_t out_frame_len;
441 in_frame_len = sbc_get_codesize(&sbc_data->enc);
442 out_frame_len = sbc_get_frame_length(&sbc_data->enc);
443 num_frames = sbc_data->payload_len / out_frame_len;
445 sbc_data->in_frame_len = in_frame_len;
446 sbc_data->in_buf_size = num_frames * in_frame_len;
448 sbc_data->out_frame_len = out_frame_len;
450 sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
451 sbc_data->frames_per_packet = num_frames;
453 DBG("in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
454 in_frame_len, out_frame_len, num_frames);
457 static int sbc_codec_init(struct audio_preset *preset, uint16_t payload_len,
460 struct sbc_data *sbc_data;
462 if (preset->len != sizeof(a2dp_sbc_t)) {
463 error("SBC: preset size mismatch");
464 return AUDIO_STATUS_FAILED;
467 sbc_data = calloc(sizeof(struct sbc_data), 1);
469 return AUDIO_STATUS_FAILED;
471 memcpy(&sbc_data->sbc, preset->data, preset->len);
473 sbc_init_encoder(sbc_data);
475 sbc_data->payload_len = payload_len;
477 sbc_codec_calculate(sbc_data);
479 *codec_data = sbc_data;
481 return AUDIO_STATUS_SUCCESS;
484 static int sbc_cleanup(void *codec_data)
486 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
488 sbc_finish(&sbc_data->enc);
491 return AUDIO_STATUS_SUCCESS;
494 static int sbc_get_config(void *codec_data, struct audio_input_config *config)
496 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
498 switch (sbc_data->sbc.frequency) {
499 case SBC_SAMPLING_FREQ_16000:
500 config->rate = 16000;
502 case SBC_SAMPLING_FREQ_32000:
503 config->rate = 32000;
505 case SBC_SAMPLING_FREQ_44100:
506 config->rate = 44100;
508 case SBC_SAMPLING_FREQ_48000:
509 config->rate = 48000;
512 return AUDIO_STATUS_FAILED;
514 config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
515 AUDIO_CHANNEL_OUT_MONO :
516 AUDIO_CHANNEL_OUT_STEREO;
517 config->format = AUDIO_FORMAT_PCM_16_BIT;
519 return AUDIO_STATUS_SUCCESS;
522 static size_t sbc_get_buffer_size(void *codec_data)
524 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
526 return sbc_data->in_buf_size;
529 static size_t sbc_get_mediapacket_duration(void *codec_data)
531 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
533 return sbc_data->frame_duration * sbc_data->frames_per_packet;
536 static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
537 size_t len, struct media_packet *mp,
538 size_t mp_data_len, size_t *written)
540 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
543 uint8_t frame_count = 0;
545 while (len - consumed >= sbc_data->in_frame_len &&
546 mp_data_len - encoded >= sbc_data->out_frame_len &&
547 frame_count < MAX_FRAMES_IN_PAYLOAD) {
551 read = sbc_encode(&sbc_data->enc, buffer + consumed,
552 sbc_data->in_frame_len, mp->data + encoded,
553 mp_data_len - encoded, &written);
556 error("SBC: failed to encode block at frame %d (%zd)",
567 mp->payload.frame_count = frame_count;
572 static bool sbc_update_qos(void *codec_data, uint8_t op)
574 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
575 uint8_t curr_bitpool = sbc_data->enc.bitpool;
576 uint8_t new_bitpool = curr_bitpool;
579 case QOS_POLICY_DEFAULT:
580 new_bitpool = sbc_data->sbc.max_bitpool;
583 case QOS_POLICY_DECREASE:
584 if (curr_bitpool > SBC_QUALITY_MIN_BITPOOL) {
585 new_bitpool = curr_bitpool - SBC_QUALITY_STEP;
586 if (new_bitpool < SBC_QUALITY_MIN_BITPOOL)
587 new_bitpool = SBC_QUALITY_MIN_BITPOOL;
592 if (new_bitpool == curr_bitpool)
595 sbc_data->enc.bitpool = new_bitpool;
597 sbc_codec_calculate(sbc_data);
599 info("SBC: bitpool changed: %d -> %d", curr_bitpool, new_bitpool);
604 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
605 void *param, size_t *rsp_len, void *rsp, int *fd)
611 char cmsgbuf[CMSG_SPACE(sizeof(int))];
613 size_t s_len = sizeof(s);
615 pthread_mutex_lock(&sk_mutex);
618 error("audio: Invalid cmd socket passed to audio_ipc_cmd");
622 if (!rsp || !rsp_len) {
623 memset(&s, 0, s_len);
628 memset(&msg, 0, sizeof(msg));
629 memset(&cmd, 0, sizeof(cmd));
631 cmd.service_id = service_id;
635 iv[0].iov_base = &cmd;
636 iv[0].iov_len = sizeof(cmd);
638 iv[1].iov_base = param;
644 ret = sendmsg(audio_sk, &msg, 0);
646 error("audio: Sending command failed:%s", strerror(errno));
650 /* socket was shutdown */
652 error("audio: Command socket closed");
656 memset(&msg, 0, sizeof(msg));
657 memset(&cmd, 0, sizeof(cmd));
659 iv[0].iov_base = &cmd;
660 iv[0].iov_len = sizeof(cmd);
662 iv[1].iov_base = rsp;
663 iv[1].iov_len = *rsp_len;
669 memset(cmsgbuf, 0, sizeof(cmsgbuf));
670 msg.msg_control = cmsgbuf;
671 msg.msg_controllen = sizeof(cmsgbuf);
674 ret = recvmsg(audio_sk, &msg, 0);
676 error("audio: Receiving command response failed:%s",
681 if (ret < (ssize_t) sizeof(cmd)) {
682 error("audio: Too small response received(%zd bytes)", ret);
686 if (cmd.service_id != service_id) {
687 error("audio: Invalid service id (%u vs %u)", cmd.service_id,
692 if (ret != (ssize_t) (sizeof(cmd) + cmd.len)) {
693 error("audio: Malformed response received(%zd bytes)", ret);
697 if (cmd.opcode != opcode && cmd.opcode != AUDIO_OP_STATUS) {
698 error("audio: Invalid opcode received (%u vs %u)",
703 if (cmd.opcode == AUDIO_OP_STATUS) {
704 struct ipc_status *s = rsp;
706 if (sizeof(*s) != cmd.len) {
707 error("audio: Invalid status length");
711 if (s->code == AUDIO_STATUS_SUCCESS) {
712 error("audio: Invalid success status response");
716 pthread_mutex_unlock(&sk_mutex);
721 pthread_mutex_unlock(&sk_mutex);
723 /* Receive auxiliary data in msg */
725 struct cmsghdr *cmsg;
729 for (cmsg = CMSG_FIRSTHDR(&msg); cmsg;
730 cmsg = CMSG_NXTHDR(&msg, cmsg)) {
731 if (cmsg->cmsg_level == SOL_SOCKET
732 && cmsg->cmsg_type == SCM_RIGHTS) {
733 memcpy(fd, CMSG_DATA(cmsg), sizeof(int));
745 return AUDIO_STATUS_SUCCESS;
748 /* Some serious issue happen on IPC - recover */
749 shutdown(audio_sk, SHUT_RDWR);
750 pthread_mutex_unlock(&sk_mutex);
752 return AUDIO_STATUS_FAILED;
755 static int ipc_open_cmd(const struct audio_codec *codec)
757 uint8_t buf[BLUEZ_AUDIO_MTU];
758 struct audio_cmd_open *cmd = (struct audio_cmd_open *) buf;
759 struct audio_rsp_open rsp;
760 size_t cmd_len = sizeof(buf) - sizeof(*cmd);
761 size_t rsp_len = sizeof(rsp);
766 memcpy(cmd->uuid, a2dp_src_uuid, sizeof(a2dp_src_uuid));
768 cmd->codec = codec->type;
769 cmd->presets = codec->get_presets(cmd->preset, &cmd_len);
771 cmd_len += sizeof(*cmd);
773 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN, cmd_len, cmd,
774 &rsp_len, &rsp, NULL);
776 if (result != AUDIO_STATUS_SUCCESS)
782 static int ipc_close_cmd(uint8_t endpoint_id)
784 struct audio_cmd_close cmd;
789 cmd.id = endpoint_id;
791 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE,
792 sizeof(cmd), &cmd, NULL, NULL, NULL);
797 static int ipc_open_stream_cmd(uint8_t endpoint_id, uint16_t *mtu, int *fd,
798 struct audio_preset **caps)
800 char buf[BLUEZ_AUDIO_MTU];
801 struct audio_cmd_open_stream cmd;
802 struct audio_rsp_open_stream *rsp =
803 (struct audio_rsp_open_stream *) &buf;
804 size_t rsp_len = sizeof(buf);
810 return AUDIO_STATUS_FAILED;
812 cmd.id = endpoint_id;
814 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN_STREAM,
815 sizeof(cmd), &cmd, &rsp_len, rsp, fd);
816 if (result == AUDIO_STATUS_SUCCESS) {
817 size_t buf_len = sizeof(struct audio_preset) +
820 *caps = malloc(buf_len);
821 memcpy(*caps, &rsp->preset, buf_len);
829 static int ipc_close_stream_cmd(uint8_t endpoint_id)
831 struct audio_cmd_close_stream cmd;
836 cmd.id = endpoint_id;
838 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE_STREAM,
839 sizeof(cmd), &cmd, NULL, NULL, NULL);
844 static int ipc_resume_stream_cmd(uint8_t endpoint_id)
846 struct audio_cmd_resume_stream cmd;
851 cmd.id = endpoint_id;
853 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_RESUME_STREAM,
854 sizeof(cmd), &cmd, NULL, NULL, NULL);
859 static int ipc_suspend_stream_cmd(uint8_t endpoint_id)
861 struct audio_cmd_suspend_stream cmd;
866 cmd.id = endpoint_id;
868 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_SUSPEND_STREAM,
869 sizeof(cmd), &cmd, NULL, NULL, NULL);
874 static int register_endpoints(void)
876 struct audio_endpoint *ep = &audio_endpoints[0];
879 for (i = 0; i < NUM_CODECS; i++, ep++) {
880 const struct audio_codec *codec = &audio_codecs[i];
882 ep->id = ipc_open_cmd(codec);
885 return AUDIO_STATUS_FAILED;
888 ep->codec_data = NULL;
892 return AUDIO_STATUS_SUCCESS;
895 static void unregister_endpoints(void)
899 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++) {
900 struct audio_endpoint *ep = &audio_endpoints[i];
903 ipc_close_cmd(ep->id);
904 memset(ep, 0, sizeof(*ep));
909 static bool open_endpoint(struct audio_endpoint *ep,
910 struct audio_input_config *cfg)
912 struct audio_preset *preset;
913 const struct audio_codec *codec;
915 uint16_t payload_len;
918 if (ipc_open_stream_cmd(ep->id, &mtu, &fd, &preset) !=
919 AUDIO_STATUS_SUCCESS)
924 payload_len = mtu - sizeof(*ep->mp);
929 codec->init(preset, payload_len, &ep->codec_data);
930 codec->get_config(ep->codec_data, cfg);
932 ep->mp = calloc(mtu, 1);
937 ep->mp->hdr.ssrc = htonl(1);
939 ep->mp_data_len = payload_len;
952 static void close_endpoint(struct audio_endpoint *ep)
954 ipc_close_stream_cmd(ep->id);
962 ep->codec->cleanup(ep->codec_data);
963 ep->codec_data = NULL;
966 static bool resume_endpoint(struct audio_endpoint *ep)
968 if (ipc_resume_stream_cmd(ep->id) != AUDIO_STATUS_SUCCESS)
974 ep->codec->update_qos(ep->codec_data, QOS_POLICY_DEFAULT);
979 static void downmix_to_mono(struct a2dp_stream_out *out, const uint8_t *buffer,
982 const int16_t *input = (const void *) buffer;
983 int16_t *output = (void *) out->downmix_buf;
986 for (i = 0; i < bytes / 2; i++) {
987 int16_t l = le16_to_cpu(get_unaligned(&input[i * 2]));
988 int16_t r = le16_to_cpu(get_unaligned(&input[i * 2 + 1]));
990 put_unaligned(cpu_to_le16((l + r) / 2), &output[i]);
994 static bool wait_for_endpoint(struct audio_endpoint *ep, bool *writable)
999 struct pollfd pollfd;
1002 pollfd.events = POLLOUT;
1005 ret = poll(&pollfd, 1, 500);
1008 *writable = !!(pollfd.revents & POLLOUT);
1012 if (errno != EINTR) {
1014 error("poll failed (%d)", ret);
1022 static bool write_to_endpoint(struct audio_endpoint *ep, size_t bytes)
1024 struct media_packet *mp = (struct media_packet *) ep->mp;
1028 ret = write(ep->fd, mp, sizeof(*mp) + bytes);
1033 /* this should not happen so let's issue warning, but do not
1034 * fail, we can try to write next packet
1036 if (errno == EAGAIN) {
1038 warn("write failed (%d)", ret);
1042 if (errno != EINTR) {
1044 error("write failed (%d)", ret);
1052 static bool write_data(struct a2dp_stream_out *out, const void *buffer,
1055 struct audio_endpoint *ep = out->ep;
1056 struct media_packet *mp = (struct media_packet *) ep->mp;
1057 size_t free_space = ep->mp_data_len;
1058 size_t consumed = 0;
1060 while (consumed < bytes) {
1065 struct timespec current;
1066 uint64_t audio_sent, audio_passed;
1067 bool do_write = false;
1069 /* prepare media packet in advance so we don't waste time after
1072 mp->hdr.sequence_number = htons(ep->seq++);
1073 mp->hdr.timestamp = htonl(ep->samples);
1074 read = ep->codec->encode_mediapacket(ep->codec_data,
1076 bytes - consumed, mp,
1077 free_space, &written);
1079 /* not much we can do here, let's just ignore remaining
1085 /* calculate where are we and where we should be */
1086 clock_gettime(CLOCK_MONOTONIC, ¤t);
1088 memcpy(&ep->start, ¤t, sizeof(ep->start));
1089 audio_sent = ep->samples * 1000000ll / out->cfg.rate;
1090 audio_passed = timespec_diff_us(¤t, &ep->start);
1092 /* if we're ahead of stream then wait for next write point
1093 * if we're lagging more than 100ms then stop writing and just
1094 * skip data until we're back in sync
1096 if (audio_sent > audio_passed) {
1097 struct timespec anchor;
1101 timespec_add(&ep->start, audio_sent, &anchor);
1104 ret = clock_nanosleep(CLOCK_MONOTONIC,
1105 TIMER_ABSTIME, &anchor,
1112 error("clock_nanosleep failed (%d)",
1117 } else if (!ep->resync) {
1118 uint64_t diff = audio_passed - audio_sent;
1120 if (diff > MAX_DELAY) {
1121 warn("lag is %jums, resyncing", diff / 1000);
1123 ep->codec->update_qos(ep->codec_data,
1124 QOS_POLICY_DECREASE);
1129 /* in resync mode we'll just drop mediapackets */
1131 /* wait some time for socket to be ready for write,
1132 * but we'll just skip writing data if timeout occurs
1134 if (!wait_for_endpoint(ep, &do_write))
1138 if (!write_to_endpoint(ep, written))
1142 /* AudioFlinger provides 16bit PCM, so sample size is 2 bytes
1143 * multiplied by number of channels. Number of channels is
1144 * simply number of bits set in channels mask.
1146 samples = read / (2 * popcount(out->cfg.channels));
1147 ep->samples += samples;
1154 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
1157 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1158 const void *in_buf = buffer;
1159 size_t in_len = bytes;
1161 /* just return in case we're closing */
1162 if (out->audio_state == AUDIO_A2DP_STATE_NONE)
1165 /* We can auto-start only from standby */
1166 if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
1167 DBG("stream in standby, auto-start");
1169 if (!resume_endpoint(out->ep))
1172 out->audio_state = AUDIO_A2DP_STATE_STARTED;
1175 if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
1176 error("audio: stream not started");
1180 if (out->ep->fd < 0) {
1181 error("audio: no transport socket");
1185 /* currently Android audioflinger is not able to provide mono stream on
1186 * A2DP output so down mixing needs to be done in hal-audio plugin.
1189 * AudioFlinger::PlaybackThread::readOutputParameters()
1190 * frameworks/av/services/audioflinger/Threads.cpp:1631
1192 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1193 if (!out->downmix_buf) {
1194 error("audio: downmix buffer not initialized");
1198 downmix_to_mono(out, buffer, bytes);
1200 in_buf = out->downmix_buf;
1204 if (!write_data(out, in_buf, in_len))
1210 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
1212 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1216 return out->cfg.rate;
1219 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1221 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1225 if (rate != out->cfg.rate) {
1226 warn("audio: cannot set sample rate to %d", rate);
1233 static size_t out_get_buffer_size(const struct audio_stream *stream)
1237 /* We should return proper buffer size calculated by codec (so each
1238 * input buffer is encoded into single media packed) but this does not
1239 * work well with AudioFlinger and causes problems. For this reason we
1240 * use magic value here and out_write code takes care of splitting
1241 * input buffer into multiple media packets.
1243 return FIXED_BUFFER_SIZE;
1246 static uint32_t out_get_channels(const struct audio_stream *stream)
1250 /* AudioFlinger can only provide stereo stream, so we return it here and
1251 * later we'll downmix this to mono in case codec requires it
1254 return AUDIO_CHANNEL_OUT_STEREO;
1257 static audio_format_t out_get_format(const struct audio_stream *stream)
1259 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1263 return out->cfg.format;
1266 static int out_set_format(struct audio_stream *stream, audio_format_t format)
1272 static int out_standby(struct audio_stream *stream)
1274 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1278 if (out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1279 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1281 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1287 static int out_dump(const struct audio_stream *stream, int fd)
1293 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
1295 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1299 bool enter_suspend = false;
1300 bool exit_suspend = false;
1304 str = strdup(kvpairs);
1308 kvpair = strtok_r(str, ";", &saveptr);
1310 for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
1313 keyval = strchr(kvpair, '=');
1320 if (!strcmp(kvpair, "closing")) {
1321 if (!strcmp(keyval, "true"))
1322 out->audio_state = AUDIO_A2DP_STATE_NONE;
1323 } else if (!strcmp(kvpair, "A2dpSuspended")) {
1324 if (!strcmp(keyval, "true"))
1325 enter_suspend = true;
1327 exit_suspend = true;
1333 if (enter_suspend && out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1334 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1336 out->audio_state = AUDIO_A2DP_STATE_SUSPENDED;
1339 if (exit_suspend && out->audio_state == AUDIO_A2DP_STATE_SUSPENDED)
1340 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1345 static char *out_get_parameters(const struct audio_stream *stream,
1352 static uint32_t out_get_latency(const struct audio_stream_out *stream)
1354 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1355 struct audio_endpoint *ep = out->ep;
1356 size_t pkt_duration;
1360 pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
1362 return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
1365 static int out_set_volume(struct audio_stream_out *stream, float left,
1369 /* volume controlled in audioflinger mixer (digital) */
1373 static int out_get_render_position(const struct audio_stream_out *stream,
1374 uint32_t *dsp_frames)
1380 static int out_add_audio_effect(const struct audio_stream *stream,
1381 effect_handle_t effect)
1387 static int out_remove_audio_effect(const struct audio_stream *stream,
1388 effect_handle_t effect)
1394 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1400 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1406 static size_t in_get_buffer_size(const struct audio_stream *stream)
1412 static uint32_t in_get_channels(const struct audio_stream *stream)
1418 static audio_format_t in_get_format(const struct audio_stream *stream)
1424 static int in_set_format(struct audio_stream *stream, audio_format_t format)
1430 static int in_standby(struct audio_stream *stream)
1436 static int in_dump(const struct audio_stream *stream, int fd)
1442 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1448 static char *in_get_parameters(const struct audio_stream *stream,
1455 static int in_set_gain(struct audio_stream_in *stream, float gain)
1461 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1468 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1474 static int in_add_audio_effect(const struct audio_stream *stream,
1475 effect_handle_t effect)
1481 static int in_remove_audio_effect(const struct audio_stream *stream,
1482 effect_handle_t effect)
1488 static int audio_open_output_stream(struct audio_hw_device *dev,
1489 audio_io_handle_t handle,
1490 audio_devices_t devices,
1491 audio_output_flags_t flags,
1492 struct audio_config *config,
1493 struct audio_stream_out **stream_out)
1496 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1497 struct a2dp_stream_out *out;
1499 out = calloc(1, sizeof(struct a2dp_stream_out));
1505 out->stream.common.get_sample_rate = out_get_sample_rate;
1506 out->stream.common.set_sample_rate = out_set_sample_rate;
1507 out->stream.common.get_buffer_size = out_get_buffer_size;
1508 out->stream.common.get_channels = out_get_channels;
1509 out->stream.common.get_format = out_get_format;
1510 out->stream.common.set_format = out_set_format;
1511 out->stream.common.standby = out_standby;
1512 out->stream.common.dump = out_dump;
1513 out->stream.common.set_parameters = out_set_parameters;
1514 out->stream.common.get_parameters = out_get_parameters;
1515 out->stream.common.add_audio_effect = out_add_audio_effect;
1516 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1517 out->stream.get_latency = out_get_latency;
1518 out->stream.set_volume = out_set_volume;
1519 out->stream.write = out_write;
1520 out->stream.get_render_position = out_get_render_position;
1522 /* TODO: for now we always use endpoint 0 */
1523 out->ep = &audio_endpoints[0];
1525 if (!open_endpoint(out->ep, &out->cfg))
1528 DBG("rate=%d channels=%d format=%d", out->cfg.rate,
1529 out->cfg.channels, out->cfg.format);
1531 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1532 out->downmix_buf = malloc(FIXED_BUFFER_SIZE / 2);
1533 if (!out->downmix_buf)
1537 *stream_out = &out->stream;
1538 a2dp_dev->out = out;
1540 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1545 error("audio: cannot open output stream");
1551 static void audio_close_output_stream(struct audio_hw_device *dev,
1552 struct audio_stream_out *stream)
1554 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1555 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1559 close_endpoint(a2dp_dev->out->ep);
1561 free(out->downmix_buf);
1564 a2dp_dev->out = NULL;
1567 static int audio_set_parameters(struct audio_hw_device *dev,
1568 const char *kvpairs)
1570 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1571 struct a2dp_stream_out *out = a2dp_dev->out;
1578 return out->stream.common.set_parameters((struct audio_stream *) out,
1582 static char *audio_get_parameters(const struct audio_hw_device *dev,
1589 static int audio_init_check(const struct audio_hw_device *dev)
1595 static int audio_set_voice_volume(struct audio_hw_device *dev, float volume)
1601 static int audio_set_master_volume(struct audio_hw_device *dev, float volume)
1607 static int audio_set_mode(struct audio_hw_device *dev, int mode)
1613 static int audio_set_mic_mute(struct audio_hw_device *dev, bool state)
1619 static int audio_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1625 static size_t audio_get_input_buffer_size(const struct audio_hw_device *dev,
1626 const struct audio_config *config)
1632 static int audio_open_input_stream(struct audio_hw_device *dev,
1633 audio_io_handle_t handle,
1634 audio_devices_t devices,
1635 struct audio_config *config,
1636 struct audio_stream_in **stream_in)
1638 struct audio_stream_in *in;
1642 in = calloc(1, sizeof(struct audio_stream_in));
1646 in->common.get_sample_rate = in_get_sample_rate;
1647 in->common.set_sample_rate = in_set_sample_rate;
1648 in->common.get_buffer_size = in_get_buffer_size;
1649 in->common.get_channels = in_get_channels;
1650 in->common.get_format = in_get_format;
1651 in->common.set_format = in_set_format;
1652 in->common.standby = in_standby;
1653 in->common.dump = in_dump;
1654 in->common.set_parameters = in_set_parameters;
1655 in->common.get_parameters = in_get_parameters;
1656 in->common.add_audio_effect = in_add_audio_effect;
1657 in->common.remove_audio_effect = in_remove_audio_effect;
1658 in->set_gain = in_set_gain;
1660 in->get_input_frames_lost = in_get_input_frames_lost;
1667 static void audio_close_input_stream(struct audio_hw_device *dev,
1668 struct audio_stream_in *stream_in)
1674 static int audio_dump(const audio_hw_device_t *device, int fd)
1680 static int audio_close(hw_device_t *device)
1682 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
1686 unregister_endpoints();
1688 shutdown(listen_sk, SHUT_RDWR);
1689 shutdown(audio_sk, SHUT_RDWR);
1691 pthread_join(ipc_th, NULL);
1700 static void *ipc_handler(void *data)
1709 DBG("Waiting for connection ...");
1711 sk = accept(listen_sk, NULL, NULL);
1718 if (err != ECONNABORTED && err != EINVAL)
1719 error("audio: Failed to accept socket: %d (%s)",
1720 err, strerror(err));
1725 pthread_mutex_lock(&sk_mutex);
1727 pthread_mutex_unlock(&sk_mutex);
1729 DBG("Audio IPC: Connected");
1731 if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
1732 error("audio: Failed to register endpoints");
1734 unregister_endpoints();
1736 pthread_mutex_lock(&sk_mutex);
1737 shutdown(audio_sk, SHUT_RDWR);
1740 pthread_mutex_unlock(&sk_mutex);
1745 memset(&pfd, 0, sizeof(pfd));
1747 pfd.events = POLLHUP | POLLERR | POLLNVAL;
1749 /* Check if socket is still alive. Empty while loop.*/
1750 while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
1752 if (pfd.revents & (POLLHUP | POLLERR | POLLNVAL)) {
1753 info("Audio HAL: Socket closed");
1755 pthread_mutex_lock(&sk_mutex);
1758 pthread_mutex_unlock(&sk_mutex);
1762 /* audio_sk is closed at this point, just cleanup endpoints states */
1763 memset(audio_endpoints, 0, sizeof(audio_endpoints));
1765 info("Closing Audio IPC thread");
1769 static int audio_ipc_init(void)
1771 struct sockaddr_un addr;
1777 sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
1780 error("audio: Failed to create socket: %d (%s)", -err,
1785 memset(&addr, 0, sizeof(addr));
1786 addr.sun_family = AF_UNIX;
1788 memcpy(addr.sun_path, BLUEZ_AUDIO_SK_PATH,
1789 sizeof(BLUEZ_AUDIO_SK_PATH));
1791 if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
1793 error("audio: Failed to bind socket: %d (%s)", -err,
1798 if (listen(sk, 1) < 0) {
1800 error("audio: Failed to listen on the socket: %d (%s)", -err,
1807 err = pthread_create(&ipc_th, NULL, ipc_handler, NULL);
1811 error("audio: Failed to start Audio IPC thread: %d (%s)",
1812 -err, strerror(-err));
1823 static int audio_open(const hw_module_t *module, const char *name,
1824 hw_device_t **device)
1826 struct a2dp_audio_dev *a2dp_dev;
1831 if (strcmp(name, AUDIO_HARDWARE_INTERFACE)) {
1832 error("audio: interface %s not matching [%s]", name,
1833 AUDIO_HARDWARE_INTERFACE);
1837 err = audio_ipc_init();
1841 a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
1845 a2dp_dev->dev.common.tag = HARDWARE_DEVICE_TAG;
1846 a2dp_dev->dev.common.version = AUDIO_DEVICE_API_VERSION_CURRENT;
1847 a2dp_dev->dev.common.module = (struct hw_module_t *) module;
1848 a2dp_dev->dev.common.close = audio_close;
1850 a2dp_dev->dev.init_check = audio_init_check;
1851 a2dp_dev->dev.set_voice_volume = audio_set_voice_volume;
1852 a2dp_dev->dev.set_master_volume = audio_set_master_volume;
1853 a2dp_dev->dev.set_mode = audio_set_mode;
1854 a2dp_dev->dev.set_mic_mute = audio_set_mic_mute;
1855 a2dp_dev->dev.get_mic_mute = audio_get_mic_mute;
1856 a2dp_dev->dev.set_parameters = audio_set_parameters;
1857 a2dp_dev->dev.get_parameters = audio_get_parameters;
1858 a2dp_dev->dev.get_input_buffer_size = audio_get_input_buffer_size;
1859 a2dp_dev->dev.open_output_stream = audio_open_output_stream;
1860 a2dp_dev->dev.close_output_stream = audio_close_output_stream;
1861 a2dp_dev->dev.open_input_stream = audio_open_input_stream;
1862 a2dp_dev->dev.close_input_stream = audio_close_input_stream;
1863 a2dp_dev->dev.dump = audio_dump;
1865 /* Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
1866 * This results from the structure of following structs:a2dp_audio_dev,
1867 * audio_hw_device. We will rely on this later in the code.*/
1868 *device = &a2dp_dev->dev.common;
1873 static struct hw_module_methods_t hal_module_methods = {
1877 struct audio_module HAL_MODULE_INFO_SYM = {
1879 .tag = HARDWARE_MODULE_TAG,
1882 .id = AUDIO_HARDWARE_MODULE_ID,
1883 .name = "A2DP Bluez HW HAL",
1884 .author = "Intel Corporation",
1885 .methods = &hal_module_methods,