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;
180 * Bionic does not have clock_nanosleep() prototype in time.h even though
181 * it provides its implementation.
183 extern int clock_nanosleep(clockid_t clock_id, int flags,
184 const struct timespec *request,
185 struct timespec *remain);
188 static int sbc_get_presets(struct audio_preset *preset, size_t *len);
189 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
191 static int sbc_cleanup(void *codec_data);
192 static int sbc_get_config(void *codec_data, struct audio_input_config *config);
193 static size_t sbc_get_buffer_size(void *codec_data);
194 static size_t sbc_get_mediapacket_duration(void *codec_data);
195 static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
196 size_t len, struct media_packet *mp,
197 size_t mp_data_len, size_t *written);
198 static bool sbc_update_qos(void *codec_data, uint8_t op);
200 #define QOS_POLICY_DEFAULT 0x00
201 #define QOS_POLICY_DECREASE 0x01
206 int (*get_presets) (struct audio_preset *preset, size_t *len);
208 int (*init) (struct audio_preset *preset, uint16_t mtu,
210 int (*cleanup) (void *codec_data);
211 int (*get_config) (void *codec_data,
212 struct audio_input_config *config);
213 size_t (*get_buffer_size) (void *codec_data);
214 size_t (*get_mediapacket_duration) (void *codec_data);
215 ssize_t (*encode_mediapacket) (void *codec_data, const uint8_t *buffer,
216 size_t len, struct media_packet *mp,
217 size_t mp_data_len, size_t *written);
218 bool (*update_qos) (void *codec_data, uint8_t op);
221 static const struct audio_codec audio_codecs[] = {
223 .type = A2DP_CODEC_SBC,
225 .get_presets = sbc_get_presets,
227 .init = sbc_codec_init,
228 .cleanup = sbc_cleanup,
229 .get_config = sbc_get_config,
230 .get_buffer_size = sbc_get_buffer_size,
231 .get_mediapacket_duration = sbc_get_mediapacket_duration,
232 .encode_mediapacket = sbc_encode_mediapacket,
233 .update_qos = sbc_update_qos,
237 #define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
239 #define MAX_AUDIO_ENDPOINTS NUM_CODECS
241 struct audio_endpoint {
243 const struct audio_codec *codec;
247 struct media_packet *mp;
252 struct timespec start;
257 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
260 AUDIO_A2DP_STATE_NONE,
261 AUDIO_A2DP_STATE_STANDBY,
262 AUDIO_A2DP_STATE_SUSPENDED,
263 AUDIO_A2DP_STATE_STARTED
266 struct a2dp_stream_out {
267 struct audio_stream_out stream;
269 struct audio_endpoint *ep;
270 enum a2dp_state_t audio_state;
271 struct audio_input_config cfg;
273 uint8_t *downmix_buf;
276 struct a2dp_audio_dev {
277 struct audio_hw_device dev;
278 struct a2dp_stream_out *out;
281 static const a2dp_sbc_t sbc_presets[] = {
283 .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
284 .channel_mode = SBC_CHANNEL_MODE_MONO |
285 SBC_CHANNEL_MODE_DUAL_CHANNEL |
286 SBC_CHANNEL_MODE_STEREO |
287 SBC_CHANNEL_MODE_JOINT_STEREO,
288 .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
289 .allocation_method = SBC_ALLOCATION_SNR |
290 SBC_ALLOCATION_LOUDNESS,
291 .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
292 SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
293 .min_bitpool = MIN_BITPOOL,
294 .max_bitpool = MAX_BITPOOL
297 .frequency = SBC_SAMPLING_FREQ_44100,
298 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
299 .subbands = SBC_SUBBANDS_8,
300 .allocation_method = SBC_ALLOCATION_LOUDNESS,
301 .block_length = SBC_BLOCK_LENGTH_16,
302 .min_bitpool = MIN_BITPOOL,
303 .max_bitpool = MAX_BITPOOL
306 .frequency = SBC_SAMPLING_FREQ_48000,
307 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
308 .subbands = SBC_SUBBANDS_8,
309 .allocation_method = SBC_ALLOCATION_LOUDNESS,
310 .block_length = SBC_BLOCK_LENGTH_16,
311 .min_bitpool = MIN_BITPOOL,
312 .max_bitpool = MAX_BITPOOL
316 static int sbc_get_presets(struct audio_preset *preset, size_t *len)
321 uint8_t *ptr = (uint8_t *) preset;
322 size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
324 count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
326 for (i = 0; i < count; i++) {
327 preset = (struct audio_preset *) ptr;
329 if (new_len + preset_size > *len)
332 preset->len = sizeof(a2dp_sbc_t);
333 memcpy(preset->data, &sbc_presets[i], preset->len);
335 new_len += preset_size;
344 static int sbc_freq2int(uint8_t freq)
347 case SBC_SAMPLING_FREQ_16000:
349 case SBC_SAMPLING_FREQ_32000:
351 case SBC_SAMPLING_FREQ_44100:
353 case SBC_SAMPLING_FREQ_48000:
360 static const char *sbc_mode2str(uint8_t mode)
363 case SBC_CHANNEL_MODE_MONO:
365 case SBC_CHANNEL_MODE_DUAL_CHANNEL:
366 return "DualChannel";
367 case SBC_CHANNEL_MODE_STEREO:
369 case SBC_CHANNEL_MODE_JOINT_STEREO:
370 return "JointStereo";
376 static int sbc_blocks2int(uint8_t blocks)
379 case SBC_BLOCK_LENGTH_4:
381 case SBC_BLOCK_LENGTH_8:
383 case SBC_BLOCK_LENGTH_12:
385 case SBC_BLOCK_LENGTH_16:
392 static int sbc_subbands2int(uint8_t subbands)
404 static const char *sbc_allocation2str(uint8_t allocation)
406 switch (allocation) {
407 case SBC_ALLOCATION_SNR:
409 case SBC_ALLOCATION_LOUDNESS:
416 static void sbc_init_encoder(struct sbc_data *sbc_data)
418 a2dp_sbc_t *in = &sbc_data->sbc;
419 sbc_t *out = &sbc_data->enc;
421 sbc_init_a2dp(out, 0L, in, sizeof(*in));
423 out->endian = SBC_LE;
424 out->bitpool = in->max_bitpool;
426 DBG("frequency=%d channel_mode=%s block_length=%d subbands=%d "
427 "allocation=%s bitpool=%d-%d",
428 sbc_freq2int(in->frequency),
429 sbc_mode2str(in->channel_mode),
430 sbc_blocks2int(in->block_length),
431 sbc_subbands2int(in->subbands),
432 sbc_allocation2str(in->allocation_method),
433 in->min_bitpool, in->max_bitpool);
436 static void sbc_codec_calculate(struct sbc_data *sbc_data)
439 size_t out_frame_len;
442 in_frame_len = sbc_get_codesize(&sbc_data->enc);
443 out_frame_len = sbc_get_frame_length(&sbc_data->enc);
444 num_frames = sbc_data->payload_len / out_frame_len;
446 sbc_data->in_frame_len = in_frame_len;
447 sbc_data->in_buf_size = num_frames * in_frame_len;
449 sbc_data->out_frame_len = out_frame_len;
451 sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
452 sbc_data->frames_per_packet = num_frames;
454 DBG("in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
455 in_frame_len, out_frame_len, num_frames);
458 static int sbc_codec_init(struct audio_preset *preset, uint16_t payload_len,
461 struct sbc_data *sbc_data;
463 if (preset->len != sizeof(a2dp_sbc_t)) {
464 error("SBC: preset size mismatch");
465 return AUDIO_STATUS_FAILED;
468 sbc_data = calloc(sizeof(struct sbc_data), 1);
470 return AUDIO_STATUS_FAILED;
472 memcpy(&sbc_data->sbc, preset->data, preset->len);
474 sbc_init_encoder(sbc_data);
476 sbc_data->payload_len = payload_len;
478 sbc_codec_calculate(sbc_data);
480 *codec_data = sbc_data;
482 return AUDIO_STATUS_SUCCESS;
485 static int sbc_cleanup(void *codec_data)
487 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
489 sbc_finish(&sbc_data->enc);
492 return AUDIO_STATUS_SUCCESS;
495 static int sbc_get_config(void *codec_data, struct audio_input_config *config)
497 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
499 switch (sbc_data->sbc.frequency) {
500 case SBC_SAMPLING_FREQ_16000:
501 config->rate = 16000;
503 case SBC_SAMPLING_FREQ_32000:
504 config->rate = 32000;
506 case SBC_SAMPLING_FREQ_44100:
507 config->rate = 44100;
509 case SBC_SAMPLING_FREQ_48000:
510 config->rate = 48000;
513 return AUDIO_STATUS_FAILED;
515 config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
516 AUDIO_CHANNEL_OUT_MONO :
517 AUDIO_CHANNEL_OUT_STEREO;
518 config->format = AUDIO_FORMAT_PCM_16_BIT;
520 return AUDIO_STATUS_SUCCESS;
523 static size_t sbc_get_buffer_size(void *codec_data)
525 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
527 return sbc_data->in_buf_size;
530 static size_t sbc_get_mediapacket_duration(void *codec_data)
532 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
534 return sbc_data->frame_duration * sbc_data->frames_per_packet;
537 static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
538 size_t len, struct media_packet *mp,
539 size_t mp_data_len, size_t *written)
541 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
544 uint8_t frame_count = 0;
546 while (len - consumed >= sbc_data->in_frame_len &&
547 mp_data_len - encoded >= sbc_data->out_frame_len &&
548 frame_count < MAX_FRAMES_IN_PAYLOAD) {
552 read = sbc_encode(&sbc_data->enc, buffer + consumed,
553 sbc_data->in_frame_len, mp->data + encoded,
554 mp_data_len - encoded, &written);
557 error("SBC: failed to encode block at frame %d (%zd)",
568 mp->payload.frame_count = frame_count;
573 static bool sbc_update_qos(void *codec_data, uint8_t op)
575 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
576 uint8_t curr_bitpool = sbc_data->enc.bitpool;
577 uint8_t new_bitpool = curr_bitpool;
580 case QOS_POLICY_DEFAULT:
581 new_bitpool = sbc_data->sbc.max_bitpool;
584 case QOS_POLICY_DECREASE:
585 if (curr_bitpool > SBC_QUALITY_MIN_BITPOOL) {
586 new_bitpool = curr_bitpool - SBC_QUALITY_STEP;
587 if (new_bitpool < SBC_QUALITY_MIN_BITPOOL)
588 new_bitpool = SBC_QUALITY_MIN_BITPOOL;
593 if (new_bitpool == curr_bitpool)
596 sbc_data->enc.bitpool = new_bitpool;
598 sbc_codec_calculate(sbc_data);
600 info("SBC: bitpool changed: %d -> %d", curr_bitpool, new_bitpool);
605 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
606 void *param, size_t *rsp_len, void *rsp, int *fd)
612 char cmsgbuf[CMSG_SPACE(sizeof(int))];
614 size_t s_len = sizeof(s);
616 pthread_mutex_lock(&sk_mutex);
619 error("audio: Invalid cmd socket passed to audio_ipc_cmd");
623 if (!rsp || !rsp_len) {
624 memset(&s, 0, s_len);
629 memset(&msg, 0, sizeof(msg));
630 memset(&cmd, 0, sizeof(cmd));
632 cmd.service_id = service_id;
636 iv[0].iov_base = &cmd;
637 iv[0].iov_len = sizeof(cmd);
639 iv[1].iov_base = param;
645 ret = sendmsg(audio_sk, &msg, 0);
647 error("audio: Sending command failed:%s", strerror(errno));
651 /* socket was shutdown */
653 error("audio: Command socket closed");
657 memset(&msg, 0, sizeof(msg));
658 memset(&cmd, 0, sizeof(cmd));
660 iv[0].iov_base = &cmd;
661 iv[0].iov_len = sizeof(cmd);
663 iv[1].iov_base = rsp;
664 iv[1].iov_len = *rsp_len;
670 memset(cmsgbuf, 0, sizeof(cmsgbuf));
671 msg.msg_control = cmsgbuf;
672 msg.msg_controllen = sizeof(cmsgbuf);
675 ret = recvmsg(audio_sk, &msg, 0);
677 error("audio: Receiving command response failed:%s",
682 if (ret < (ssize_t) sizeof(cmd)) {
683 error("audio: Too small response received(%zd bytes)", ret);
687 if (cmd.service_id != service_id) {
688 error("audio: Invalid service id (%u vs %u)", cmd.service_id,
693 if (ret != (ssize_t) (sizeof(cmd) + cmd.len)) {
694 error("audio: Malformed response received(%zd bytes)", ret);
698 if (cmd.opcode != opcode && cmd.opcode != AUDIO_OP_STATUS) {
699 error("audio: Invalid opcode received (%u vs %u)",
704 if (cmd.opcode == AUDIO_OP_STATUS) {
705 struct ipc_status *s = rsp;
707 if (sizeof(*s) != cmd.len) {
708 error("audio: Invalid status length");
712 if (s->code == AUDIO_STATUS_SUCCESS) {
713 error("audio: Invalid success status response");
717 pthread_mutex_unlock(&sk_mutex);
722 pthread_mutex_unlock(&sk_mutex);
724 /* Receive auxiliary data in msg */
726 struct cmsghdr *cmsg;
730 for (cmsg = CMSG_FIRSTHDR(&msg); cmsg;
731 cmsg = CMSG_NXTHDR(&msg, cmsg)) {
732 if (cmsg->cmsg_level == SOL_SOCKET
733 && cmsg->cmsg_type == SCM_RIGHTS) {
734 memcpy(fd, CMSG_DATA(cmsg), sizeof(int));
746 return AUDIO_STATUS_SUCCESS;
749 /* Some serious issue happen on IPC - recover */
750 shutdown(audio_sk, SHUT_RDWR);
751 pthread_mutex_unlock(&sk_mutex);
753 return AUDIO_STATUS_FAILED;
756 static int ipc_open_cmd(const struct audio_codec *codec)
758 uint8_t buf[BLUEZ_AUDIO_MTU];
759 struct audio_cmd_open *cmd = (struct audio_cmd_open *) buf;
760 struct audio_rsp_open rsp;
761 size_t cmd_len = sizeof(buf) - sizeof(*cmd);
762 size_t rsp_len = sizeof(rsp);
767 memcpy(cmd->uuid, a2dp_src_uuid, sizeof(a2dp_src_uuid));
769 cmd->codec = codec->type;
770 cmd->presets = codec->get_presets(cmd->preset, &cmd_len);
772 cmd_len += sizeof(*cmd);
774 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN, cmd_len, cmd,
775 &rsp_len, &rsp, NULL);
777 if (result != AUDIO_STATUS_SUCCESS)
783 static int ipc_close_cmd(uint8_t endpoint_id)
785 struct audio_cmd_close cmd;
790 cmd.id = endpoint_id;
792 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE,
793 sizeof(cmd), &cmd, NULL, NULL, NULL);
798 static int ipc_open_stream_cmd(uint8_t endpoint_id, uint16_t *mtu, int *fd,
799 struct audio_preset **caps)
801 char buf[BLUEZ_AUDIO_MTU];
802 struct audio_cmd_open_stream cmd;
803 struct audio_rsp_open_stream *rsp =
804 (struct audio_rsp_open_stream *) &buf;
805 size_t rsp_len = sizeof(buf);
811 return AUDIO_STATUS_FAILED;
813 cmd.id = endpoint_id;
815 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN_STREAM,
816 sizeof(cmd), &cmd, &rsp_len, rsp, fd);
817 if (result == AUDIO_STATUS_SUCCESS) {
818 size_t buf_len = sizeof(struct audio_preset) +
821 *caps = malloc(buf_len);
822 memcpy(*caps, &rsp->preset, buf_len);
830 static int ipc_close_stream_cmd(uint8_t endpoint_id)
832 struct audio_cmd_close_stream cmd;
837 cmd.id = endpoint_id;
839 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE_STREAM,
840 sizeof(cmd), &cmd, NULL, NULL, NULL);
845 static int ipc_resume_stream_cmd(uint8_t endpoint_id)
847 struct audio_cmd_resume_stream cmd;
852 cmd.id = endpoint_id;
854 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_RESUME_STREAM,
855 sizeof(cmd), &cmd, NULL, NULL, NULL);
860 static int ipc_suspend_stream_cmd(uint8_t endpoint_id)
862 struct audio_cmd_suspend_stream cmd;
867 cmd.id = endpoint_id;
869 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_SUSPEND_STREAM,
870 sizeof(cmd), &cmd, NULL, NULL, NULL);
875 static int register_endpoints(void)
877 struct audio_endpoint *ep = &audio_endpoints[0];
880 for (i = 0; i < NUM_CODECS; i++, ep++) {
881 const struct audio_codec *codec = &audio_codecs[i];
883 ep->id = ipc_open_cmd(codec);
886 return AUDIO_STATUS_FAILED;
889 ep->codec_data = NULL;
893 return AUDIO_STATUS_SUCCESS;
896 static void unregister_endpoints(void)
900 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++) {
901 struct audio_endpoint *ep = &audio_endpoints[i];
904 ipc_close_cmd(ep->id);
905 memset(ep, 0, sizeof(*ep));
910 static bool open_endpoint(struct audio_endpoint *ep,
911 struct audio_input_config *cfg)
913 struct audio_preset *preset;
914 const struct audio_codec *codec;
916 uint16_t payload_len;
919 if (ipc_open_stream_cmd(ep->id, &mtu, &fd, &preset) !=
920 AUDIO_STATUS_SUCCESS)
925 payload_len = mtu - sizeof(*ep->mp);
930 codec->init(preset, payload_len, &ep->codec_data);
931 codec->get_config(ep->codec_data, cfg);
933 ep->mp = calloc(mtu, 1);
938 ep->mp->hdr.ssrc = htonl(1);
940 ep->mp_data_len = payload_len;
953 static void close_endpoint(struct audio_endpoint *ep)
955 ipc_close_stream_cmd(ep->id);
963 ep->codec->cleanup(ep->codec_data);
964 ep->codec_data = NULL;
967 static bool resume_endpoint(struct audio_endpoint *ep)
969 if (ipc_resume_stream_cmd(ep->id) != AUDIO_STATUS_SUCCESS)
975 ep->codec->update_qos(ep->codec_data, QOS_POLICY_DEFAULT);
980 static void downmix_to_mono(struct a2dp_stream_out *out, const uint8_t *buffer,
983 const int16_t *input = (const void *) buffer;
984 int16_t *output = (void *) out->downmix_buf;
987 for (i = 0; i < bytes / 2; i++) {
988 int16_t l = le16_to_cpu(get_unaligned(&input[i * 2]));
989 int16_t r = le16_to_cpu(get_unaligned(&input[i * 2 + 1]));
991 put_unaligned(cpu_to_le16((l + r) / 2), &output[i]);
995 static bool wait_for_endpoint(struct audio_endpoint *ep, bool *writable)
1000 struct pollfd pollfd;
1003 pollfd.events = POLLOUT;
1006 ret = poll(&pollfd, 1, 500);
1009 *writable = !!(pollfd.revents & POLLOUT);
1013 if (errno != EINTR) {
1015 error("poll failed (%d)", ret);
1023 static bool write_to_endpoint(struct audio_endpoint *ep, size_t bytes)
1025 struct media_packet *mp = (struct media_packet *) ep->mp;
1029 ret = write(ep->fd, mp, sizeof(*mp) + bytes);
1035 * this should not happen so let's issue warning, but do not
1036 * fail, we can try to write next packet
1038 if (errno == EAGAIN) {
1040 warn("write failed (%d)", ret);
1044 if (errno != EINTR) {
1046 error("write failed (%d)", ret);
1054 static bool write_data(struct a2dp_stream_out *out, const void *buffer,
1057 struct audio_endpoint *ep = out->ep;
1058 struct media_packet *mp = (struct media_packet *) ep->mp;
1059 size_t free_space = ep->mp_data_len;
1060 size_t consumed = 0;
1062 while (consumed < bytes) {
1067 struct timespec current;
1068 uint64_t audio_sent, audio_passed;
1069 bool do_write = false;
1072 * prepare media packet in advance so we don't waste time after
1075 mp->hdr.sequence_number = htons(ep->seq++);
1076 mp->hdr.timestamp = htonl(ep->samples);
1077 read = ep->codec->encode_mediapacket(ep->codec_data,
1079 bytes - consumed, mp,
1080 free_space, &written);
1083 * not much we can do here, let's just ignore remaining
1089 /* calculate where are we and where we should be */
1090 clock_gettime(CLOCK_MONOTONIC, ¤t);
1092 memcpy(&ep->start, ¤t, sizeof(ep->start));
1093 audio_sent = ep->samples * 1000000ll / out->cfg.rate;
1094 audio_passed = timespec_diff_us(¤t, &ep->start);
1097 * if we're ahead of stream then wait for next write point,
1098 * if we're lagging more than 100ms then stop writing and just
1099 * skip data until we're back in sync
1101 if (audio_sent > audio_passed) {
1102 struct timespec anchor;
1106 timespec_add(&ep->start, audio_sent, &anchor);
1109 ret = clock_nanosleep(CLOCK_MONOTONIC,
1110 TIMER_ABSTIME, &anchor,
1117 error("clock_nanosleep failed (%d)",
1122 } else if (!ep->resync) {
1123 uint64_t diff = audio_passed - audio_sent;
1125 if (diff > MAX_DELAY) {
1126 warn("lag is %jums, resyncing", diff / 1000);
1128 ep->codec->update_qos(ep->codec_data,
1129 QOS_POLICY_DECREASE);
1134 /* in resync mode we'll just drop mediapackets */
1136 /* wait some time for socket to be ready for write,
1137 * but we'll just skip writing data if timeout occurs
1139 if (!wait_for_endpoint(ep, &do_write))
1143 if (!write_to_endpoint(ep, written))
1148 * AudioFlinger provides 16bit PCM, so sample size is 2 bytes
1149 * multiplied by number of channels. Number of channels is
1150 * simply number of bits set in channels mask.
1152 samples = read / (2 * popcount(out->cfg.channels));
1153 ep->samples += samples;
1160 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
1163 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1164 const void *in_buf = buffer;
1165 size_t in_len = bytes;
1167 /* just return in case we're closing */
1168 if (out->audio_state == AUDIO_A2DP_STATE_NONE)
1171 /* We can auto-start only from standby */
1172 if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
1173 DBG("stream in standby, auto-start");
1175 if (!resume_endpoint(out->ep))
1178 out->audio_state = AUDIO_A2DP_STATE_STARTED;
1181 if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
1182 error("audio: stream not started");
1186 if (out->ep->fd < 0) {
1187 error("audio: no transport socket");
1192 * currently Android audioflinger is not able to provide mono stream on
1193 * A2DP output so down mixing needs to be done in hal-audio plugin.
1196 * AudioFlinger::PlaybackThread::readOutputParameters()
1197 * frameworks/av/services/audioflinger/Threads.cpp:1631
1199 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1200 if (!out->downmix_buf) {
1201 error("audio: downmix buffer not initialized");
1205 downmix_to_mono(out, buffer, bytes);
1207 in_buf = out->downmix_buf;
1211 if (!write_data(out, in_buf, in_len))
1217 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
1219 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1223 return out->cfg.rate;
1226 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1228 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1232 if (rate != out->cfg.rate) {
1233 warn("audio: cannot set sample rate to %d", rate);
1240 static size_t out_get_buffer_size(const struct audio_stream *stream)
1245 * We should return proper buffer size calculated by codec (so each
1246 * input buffer is encoded into single media packed) but this does not
1247 * work well with AudioFlinger and causes problems. For this reason we
1248 * use magic value here and out_write code takes care of splitting
1249 * input buffer into multiple media packets.
1251 return FIXED_BUFFER_SIZE;
1254 static uint32_t out_get_channels(const struct audio_stream *stream)
1259 * AudioFlinger can only provide stereo stream, so we return it here and
1260 * later we'll downmix this to mono in case codec requires it
1263 return AUDIO_CHANNEL_OUT_STEREO;
1266 static audio_format_t out_get_format(const struct audio_stream *stream)
1268 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1272 return out->cfg.format;
1275 static int out_set_format(struct audio_stream *stream, audio_format_t format)
1281 static int out_standby(struct audio_stream *stream)
1283 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1287 if (out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1288 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1290 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1296 static int out_dump(const struct audio_stream *stream, int fd)
1302 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
1304 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1308 bool enter_suspend = false;
1309 bool exit_suspend = false;
1313 str = strdup(kvpairs);
1317 kvpair = strtok_r(str, ";", &saveptr);
1319 for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
1322 keyval = strchr(kvpair, '=');
1329 if (!strcmp(kvpair, "closing")) {
1330 if (!strcmp(keyval, "true"))
1331 out->audio_state = AUDIO_A2DP_STATE_NONE;
1332 } else if (!strcmp(kvpair, "A2dpSuspended")) {
1333 if (!strcmp(keyval, "true"))
1334 enter_suspend = true;
1336 exit_suspend = true;
1342 if (enter_suspend && out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1343 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1345 out->audio_state = AUDIO_A2DP_STATE_SUSPENDED;
1348 if (exit_suspend && out->audio_state == AUDIO_A2DP_STATE_SUSPENDED)
1349 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1354 static char *out_get_parameters(const struct audio_stream *stream,
1361 static uint32_t out_get_latency(const struct audio_stream_out *stream)
1363 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1364 struct audio_endpoint *ep = out->ep;
1365 size_t pkt_duration;
1369 pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
1371 return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
1374 static int out_set_volume(struct audio_stream_out *stream, float left,
1378 /* volume controlled in audioflinger mixer (digital) */
1382 static int out_get_render_position(const struct audio_stream_out *stream,
1383 uint32_t *dsp_frames)
1389 static int out_add_audio_effect(const struct audio_stream *stream,
1390 effect_handle_t effect)
1396 static int out_remove_audio_effect(const struct audio_stream *stream,
1397 effect_handle_t effect)
1403 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1409 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1415 static size_t in_get_buffer_size(const struct audio_stream *stream)
1421 static uint32_t in_get_channels(const struct audio_stream *stream)
1427 static audio_format_t in_get_format(const struct audio_stream *stream)
1433 static int in_set_format(struct audio_stream *stream, audio_format_t format)
1439 static int in_standby(struct audio_stream *stream)
1445 static int in_dump(const struct audio_stream *stream, int fd)
1451 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1457 static char *in_get_parameters(const struct audio_stream *stream,
1464 static int in_set_gain(struct audio_stream_in *stream, float gain)
1470 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1477 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1483 static int in_add_audio_effect(const struct audio_stream *stream,
1484 effect_handle_t effect)
1490 static int in_remove_audio_effect(const struct audio_stream *stream,
1491 effect_handle_t effect)
1497 static int audio_open_output_stream(struct audio_hw_device *dev,
1498 audio_io_handle_t handle,
1499 audio_devices_t devices,
1500 audio_output_flags_t flags,
1501 struct audio_config *config,
1502 struct audio_stream_out **stream_out)
1505 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1506 struct a2dp_stream_out *out;
1508 out = calloc(1, sizeof(struct a2dp_stream_out));
1514 out->stream.common.get_sample_rate = out_get_sample_rate;
1515 out->stream.common.set_sample_rate = out_set_sample_rate;
1516 out->stream.common.get_buffer_size = out_get_buffer_size;
1517 out->stream.common.get_channels = out_get_channels;
1518 out->stream.common.get_format = out_get_format;
1519 out->stream.common.set_format = out_set_format;
1520 out->stream.common.standby = out_standby;
1521 out->stream.common.dump = out_dump;
1522 out->stream.common.set_parameters = out_set_parameters;
1523 out->stream.common.get_parameters = out_get_parameters;
1524 out->stream.common.add_audio_effect = out_add_audio_effect;
1525 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1526 out->stream.get_latency = out_get_latency;
1527 out->stream.set_volume = out_set_volume;
1528 out->stream.write = out_write;
1529 out->stream.get_render_position = out_get_render_position;
1531 /* TODO: for now we always use endpoint 0 */
1532 out->ep = &audio_endpoints[0];
1534 if (!open_endpoint(out->ep, &out->cfg))
1537 DBG("rate=%d channels=%d format=%d", out->cfg.rate,
1538 out->cfg.channels, out->cfg.format);
1540 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1541 out->downmix_buf = malloc(FIXED_BUFFER_SIZE / 2);
1542 if (!out->downmix_buf)
1546 *stream_out = &out->stream;
1547 a2dp_dev->out = out;
1549 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1554 error("audio: cannot open output stream");
1560 static void audio_close_output_stream(struct audio_hw_device *dev,
1561 struct audio_stream_out *stream)
1563 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1564 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1568 close_endpoint(a2dp_dev->out->ep);
1570 free(out->downmix_buf);
1573 a2dp_dev->out = NULL;
1576 static int audio_set_parameters(struct audio_hw_device *dev,
1577 const char *kvpairs)
1579 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1580 struct a2dp_stream_out *out = a2dp_dev->out;
1587 return out->stream.common.set_parameters((struct audio_stream *) out,
1591 static char *audio_get_parameters(const struct audio_hw_device *dev,
1598 static int audio_init_check(const struct audio_hw_device *dev)
1604 static int audio_set_voice_volume(struct audio_hw_device *dev, float volume)
1610 static int audio_set_master_volume(struct audio_hw_device *dev, float volume)
1616 static int audio_set_mode(struct audio_hw_device *dev, int mode)
1622 static int audio_set_mic_mute(struct audio_hw_device *dev, bool state)
1628 static int audio_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1634 static size_t audio_get_input_buffer_size(const struct audio_hw_device *dev,
1635 const struct audio_config *config)
1641 static int audio_open_input_stream(struct audio_hw_device *dev,
1642 audio_io_handle_t handle,
1643 audio_devices_t devices,
1644 struct audio_config *config,
1645 struct audio_stream_in **stream_in)
1647 struct audio_stream_in *in;
1651 in = calloc(1, sizeof(struct audio_stream_in));
1655 in->common.get_sample_rate = in_get_sample_rate;
1656 in->common.set_sample_rate = in_set_sample_rate;
1657 in->common.get_buffer_size = in_get_buffer_size;
1658 in->common.get_channels = in_get_channels;
1659 in->common.get_format = in_get_format;
1660 in->common.set_format = in_set_format;
1661 in->common.standby = in_standby;
1662 in->common.dump = in_dump;
1663 in->common.set_parameters = in_set_parameters;
1664 in->common.get_parameters = in_get_parameters;
1665 in->common.add_audio_effect = in_add_audio_effect;
1666 in->common.remove_audio_effect = in_remove_audio_effect;
1667 in->set_gain = in_set_gain;
1669 in->get_input_frames_lost = in_get_input_frames_lost;
1676 static void audio_close_input_stream(struct audio_hw_device *dev,
1677 struct audio_stream_in *stream_in)
1683 static int audio_dump(const audio_hw_device_t *device, int fd)
1689 static int audio_close(hw_device_t *device)
1691 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
1695 unregister_endpoints();
1697 shutdown(listen_sk, SHUT_RDWR);
1698 shutdown(audio_sk, SHUT_RDWR);
1700 pthread_join(ipc_th, NULL);
1709 static void *ipc_handler(void *data)
1718 DBG("Waiting for connection ...");
1720 sk = accept(listen_sk, NULL, NULL);
1727 if (err != ECONNABORTED && err != EINVAL)
1728 error("audio: Failed to accept socket: %d (%s)",
1729 err, strerror(err));
1734 pthread_mutex_lock(&sk_mutex);
1736 pthread_mutex_unlock(&sk_mutex);
1738 DBG("Audio IPC: Connected");
1740 if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
1741 error("audio: Failed to register endpoints");
1743 unregister_endpoints();
1745 pthread_mutex_lock(&sk_mutex);
1746 shutdown(audio_sk, SHUT_RDWR);
1749 pthread_mutex_unlock(&sk_mutex);
1754 memset(&pfd, 0, sizeof(pfd));
1756 pfd.events = POLLHUP | POLLERR | POLLNVAL;
1758 /* Check if socket is still alive. Empty while loop.*/
1759 while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
1761 if (pfd.revents & (POLLHUP | POLLERR | POLLNVAL)) {
1762 info("Audio HAL: Socket closed");
1764 pthread_mutex_lock(&sk_mutex);
1767 pthread_mutex_unlock(&sk_mutex);
1771 /* audio_sk is closed at this point, just cleanup endpoints states */
1772 memset(audio_endpoints, 0, sizeof(audio_endpoints));
1774 info("Closing Audio IPC thread");
1778 static int audio_ipc_init(void)
1780 struct sockaddr_un addr;
1786 sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
1789 error("audio: Failed to create socket: %d (%s)", -err,
1794 memset(&addr, 0, sizeof(addr));
1795 addr.sun_family = AF_UNIX;
1797 memcpy(addr.sun_path, BLUEZ_AUDIO_SK_PATH,
1798 sizeof(BLUEZ_AUDIO_SK_PATH));
1800 if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
1802 error("audio: Failed to bind socket: %d (%s)", -err,
1807 if (listen(sk, 1) < 0) {
1809 error("audio: Failed to listen on the socket: %d (%s)", -err,
1816 err = pthread_create(&ipc_th, NULL, ipc_handler, NULL);
1820 error("audio: Failed to start Audio IPC thread: %d (%s)",
1821 -err, strerror(-err));
1832 static int audio_open(const hw_module_t *module, const char *name,
1833 hw_device_t **device)
1835 struct a2dp_audio_dev *a2dp_dev;
1840 if (strcmp(name, AUDIO_HARDWARE_INTERFACE)) {
1841 error("audio: interface %s not matching [%s]", name,
1842 AUDIO_HARDWARE_INTERFACE);
1846 err = audio_ipc_init();
1850 a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
1854 a2dp_dev->dev.common.tag = HARDWARE_DEVICE_TAG;
1855 a2dp_dev->dev.common.version = AUDIO_DEVICE_API_VERSION_CURRENT;
1856 a2dp_dev->dev.common.module = (struct hw_module_t *) module;
1857 a2dp_dev->dev.common.close = audio_close;
1859 a2dp_dev->dev.init_check = audio_init_check;
1860 a2dp_dev->dev.set_voice_volume = audio_set_voice_volume;
1861 a2dp_dev->dev.set_master_volume = audio_set_master_volume;
1862 a2dp_dev->dev.set_mode = audio_set_mode;
1863 a2dp_dev->dev.set_mic_mute = audio_set_mic_mute;
1864 a2dp_dev->dev.get_mic_mute = audio_get_mic_mute;
1865 a2dp_dev->dev.set_parameters = audio_set_parameters;
1866 a2dp_dev->dev.get_parameters = audio_get_parameters;
1867 a2dp_dev->dev.get_input_buffer_size = audio_get_input_buffer_size;
1868 a2dp_dev->dev.open_output_stream = audio_open_output_stream;
1869 a2dp_dev->dev.close_output_stream = audio_close_output_stream;
1870 a2dp_dev->dev.open_input_stream = audio_open_input_stream;
1871 a2dp_dev->dev.close_input_stream = audio_close_input_stream;
1872 a2dp_dev->dev.dump = audio_dump;
1875 * Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
1876 * This results from the structure of following structs:a2dp_audio_dev,
1877 * audio_hw_device. We will rely on this later in the code.
1879 *device = &a2dp_dev->dev.common;
1884 static struct hw_module_methods_t hal_module_methods = {
1888 struct audio_module HAL_MODULE_INFO_SYM = {
1890 .tag = HARDWARE_MODULE_TAG,
1893 .id = AUDIO_HARDWARE_MODULE_ID,
1894 .name = "A2DP Bluez HW HAL",
1895 .author = "Intel Corporation",
1896 .methods = &hal_module_methods,