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>
33 #include "audio-msg.h"
34 #include "ipc-common.h"
37 #include "hal-audio.h"
38 #include "src/shared/util.h"
39 #include "src/shared/queue.h"
41 #define FIXED_A2DP_PLAYBACK_LATENCY_MS 25
43 #define FIXED_BUFFER_SIZE (20 * 512)
45 #define MAX_DELAY 100000 /* 100ms */
47 static const uint8_t a2dp_src_uuid[] = {
48 0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
49 0x80, 0x00, 0x00, 0x80, 0x5f, 0x9b, 0x34, 0xfb };
51 static int listen_sk = -1;
52 static int audio_sk = -1;
54 static pthread_t ipc_th = 0;
55 static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
57 static void timespec_add(struct timespec *base, uint64_t time_us,
60 res->tv_sec = base->tv_sec + time_us / 1000000;
61 res->tv_nsec = base->tv_nsec + (time_us % 1000000) * 1000;
63 if (res->tv_nsec >= 1000000000) {
65 res->tv_nsec -= 1000000000;
69 static void timespec_diff(struct timespec *a, struct timespec *b,
72 res->tv_sec = a->tv_sec - b->tv_sec;
73 res->tv_nsec = a->tv_nsec - b->tv_nsec;
75 if (res->tv_nsec < 0) {
77 res->tv_nsec += 1000000000; /* 1sec */
81 static uint64_t timespec_diff_us(struct timespec *a, struct timespec *b)
85 timespec_diff(a, b, &res);
87 return res.tv_sec * 1000000ll + res.tv_nsec / 1000ll;
92 * Bionic does not have clock_nanosleep() prototype in time.h even though
93 * it provides its implementation.
95 extern int clock_nanosleep(clockid_t clock_id, int flags,
96 const struct timespec *request,
97 struct timespec *remain);
100 static const audio_codec_get_t audio_codecs[] = {
105 #define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
107 #define MAX_AUDIO_ENDPOINTS NUM_CODECS
109 static struct queue *loaded_codecs;
111 struct audio_endpoint {
113 const struct audio_codec *codec;
117 struct media_packet *mp;
122 struct timespec start;
127 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
130 AUDIO_A2DP_STATE_NONE,
131 AUDIO_A2DP_STATE_STANDBY,
132 AUDIO_A2DP_STATE_SUSPENDED,
133 AUDIO_A2DP_STATE_STARTED
136 struct a2dp_stream_out {
137 struct audio_stream_out stream;
139 struct audio_endpoint *ep;
140 enum a2dp_state_t audio_state;
141 struct audio_input_config cfg;
143 uint8_t *downmix_buf;
146 struct a2dp_audio_dev {
147 struct audio_hw_device dev;
148 struct a2dp_stream_out *out;
151 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
152 void *param, size_t *rsp_len, void *rsp, int *fd)
158 char cmsgbuf[CMSG_SPACE(sizeof(int))];
160 size_t s_len = sizeof(s);
162 pthread_mutex_lock(&sk_mutex);
165 error("audio: Invalid cmd socket passed to audio_ipc_cmd");
169 if (!rsp || !rsp_len) {
170 memset(&s, 0, s_len);
175 memset(&msg, 0, sizeof(msg));
176 memset(&cmd, 0, sizeof(cmd));
178 cmd.service_id = service_id;
182 iv[0].iov_base = &cmd;
183 iv[0].iov_len = sizeof(cmd);
185 iv[1].iov_base = param;
191 ret = sendmsg(audio_sk, &msg, 0);
193 error("audio: Sending command failed:%s", strerror(errno));
197 /* socket was shutdown */
199 error("audio: Command socket closed");
203 memset(&msg, 0, sizeof(msg));
204 memset(&cmd, 0, sizeof(cmd));
206 iv[0].iov_base = &cmd;
207 iv[0].iov_len = sizeof(cmd);
209 iv[1].iov_base = rsp;
210 iv[1].iov_len = *rsp_len;
216 memset(cmsgbuf, 0, sizeof(cmsgbuf));
217 msg.msg_control = cmsgbuf;
218 msg.msg_controllen = sizeof(cmsgbuf);
221 ret = recvmsg(audio_sk, &msg, 0);
223 error("audio: Receiving command response failed:%s",
228 if (ret < (ssize_t) sizeof(cmd)) {
229 error("audio: Too small response received(%zd bytes)", ret);
233 if (cmd.service_id != service_id) {
234 error("audio: Invalid service id (%u vs %u)", cmd.service_id,
239 if (ret != (ssize_t) (sizeof(cmd) + cmd.len)) {
240 error("audio: Malformed response received(%zd bytes)", ret);
244 if (cmd.opcode != opcode && cmd.opcode != AUDIO_OP_STATUS) {
245 error("audio: Invalid opcode received (%u vs %u)",
250 if (cmd.opcode == AUDIO_OP_STATUS) {
251 struct ipc_status *s = rsp;
253 if (sizeof(*s) != cmd.len) {
254 error("audio: Invalid status length");
258 if (s->code == AUDIO_STATUS_SUCCESS) {
259 error("audio: Invalid success status response");
263 pthread_mutex_unlock(&sk_mutex);
268 pthread_mutex_unlock(&sk_mutex);
270 /* Receive auxiliary data in msg */
272 struct cmsghdr *cmsg;
276 for (cmsg = CMSG_FIRSTHDR(&msg); cmsg;
277 cmsg = CMSG_NXTHDR(&msg, cmsg)) {
278 if (cmsg->cmsg_level == SOL_SOCKET
279 && cmsg->cmsg_type == SCM_RIGHTS) {
280 memcpy(fd, CMSG_DATA(cmsg), sizeof(int));
292 return AUDIO_STATUS_SUCCESS;
295 /* Some serious issue happen on IPC - recover */
296 shutdown(audio_sk, SHUT_RDWR);
297 pthread_mutex_unlock(&sk_mutex);
299 return AUDIO_STATUS_FAILED;
302 static int ipc_open_cmd(const struct audio_codec *codec)
304 uint8_t buf[BLUEZ_AUDIO_MTU];
305 struct audio_cmd_open *cmd = (struct audio_cmd_open *) buf;
306 struct audio_rsp_open rsp;
307 size_t cmd_len = sizeof(buf) - sizeof(*cmd);
308 size_t rsp_len = sizeof(rsp);
313 memcpy(cmd->uuid, a2dp_src_uuid, sizeof(a2dp_src_uuid));
315 cmd->codec = codec->type;
316 cmd->presets = codec->get_presets(cmd->preset, &cmd_len);
318 cmd_len += sizeof(*cmd);
320 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN, cmd_len, cmd,
321 &rsp_len, &rsp, NULL);
323 if (result != AUDIO_STATUS_SUCCESS)
329 static int ipc_close_cmd(uint8_t endpoint_id)
331 struct audio_cmd_close cmd;
336 cmd.id = endpoint_id;
338 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE,
339 sizeof(cmd), &cmd, NULL, NULL, NULL);
344 static int ipc_open_stream_cmd(uint8_t *endpoint_id, uint16_t *mtu, int *fd,
345 struct audio_preset **caps)
347 char buf[BLUEZ_AUDIO_MTU];
348 struct audio_cmd_open_stream cmd;
349 struct audio_rsp_open_stream *rsp =
350 (struct audio_rsp_open_stream *) &buf;
351 size_t rsp_len = sizeof(buf);
357 return AUDIO_STATUS_FAILED;
359 cmd.id = *endpoint_id;
361 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN_STREAM,
362 sizeof(cmd), &cmd, &rsp_len, rsp, fd);
363 if (result == AUDIO_STATUS_SUCCESS) {
364 size_t buf_len = sizeof(struct audio_preset) +
366 *endpoint_id = rsp->id;
368 *caps = malloc(buf_len);
369 memcpy(*caps, &rsp->preset, buf_len);
377 static int ipc_close_stream_cmd(uint8_t endpoint_id)
379 struct audio_cmd_close_stream cmd;
384 cmd.id = endpoint_id;
386 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE_STREAM,
387 sizeof(cmd), &cmd, NULL, NULL, NULL);
392 static int ipc_resume_stream_cmd(uint8_t endpoint_id)
394 struct audio_cmd_resume_stream cmd;
399 cmd.id = endpoint_id;
401 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_RESUME_STREAM,
402 sizeof(cmd), &cmd, NULL, NULL, NULL);
407 static int ipc_suspend_stream_cmd(uint8_t endpoint_id)
409 struct audio_cmd_suspend_stream cmd;
414 cmd.id = endpoint_id;
416 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_SUSPEND_STREAM,
417 sizeof(cmd), &cmd, NULL, NULL, NULL);
422 struct register_state {
423 struct audio_endpoint *ep;
427 static void register_endpoint(void *data, void *user_data)
429 struct audio_codec *codec = data;
430 struct register_state *state = user_data;
431 struct audio_endpoint *ep = state->ep;
433 /* don't even try to register more endpoints if one failed */
437 ep->id = ipc_open_cmd(codec);
441 error("Failed to register endpoint");
446 ep->codec_data = NULL;
452 static int register_endpoints(void)
454 struct register_state state;
456 state.ep = &audio_endpoints[0];
459 queue_foreach(loaded_codecs, register_endpoint, &state);
461 return state.error ? AUDIO_STATUS_FAILED : AUDIO_STATUS_SUCCESS;
464 static void unregister_endpoints(void)
468 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++) {
469 struct audio_endpoint *ep = &audio_endpoints[i];
472 ipc_close_cmd(ep->id);
473 memset(ep, 0, sizeof(*ep));
478 static bool open_endpoint(struct audio_endpoint **epp,
479 struct audio_input_config *cfg)
481 struct audio_preset *preset;
482 struct audio_endpoint *ep = *epp;
483 const struct audio_codec *codec;
485 uint16_t payload_len;
493 if (ipc_open_stream_cmd(&ep_id, &mtu, &fd, &preset) !=
494 AUDIO_STATUS_SUCCESS)
497 DBG("ep_id=%d mtu=%u", ep_id, mtu);
499 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++)
500 if (audio_endpoints[i].id == ep_id) {
501 ep = &audio_endpoints[i];
506 error("Cound not find opened endpoint");
513 if (ep->codec->use_rtp)
514 payload_len -= sizeof(struct rtp_header);
519 codec->init(preset, payload_len, &ep->codec_data);
520 codec->get_config(ep->codec_data, cfg);
522 ep->mp = calloc(mtu, 1);
526 if (ep->codec->use_rtp) {
527 struct media_packet_rtp *mp_rtp =
528 (struct media_packet_rtp *) ep->mp;
530 mp_rtp->hdr.pt = 0x60;
531 mp_rtp->hdr.ssrc = htonl(1);
534 ep->mp_data_len = payload_len;
547 static void close_endpoint(struct audio_endpoint *ep)
549 ipc_close_stream_cmd(ep->id);
557 ep->codec->cleanup(ep->codec_data);
558 ep->codec_data = NULL;
561 static bool resume_endpoint(struct audio_endpoint *ep)
563 if (ipc_resume_stream_cmd(ep->id) != AUDIO_STATUS_SUCCESS)
569 ep->codec->update_qos(ep->codec_data, QOS_POLICY_DEFAULT);
574 static void downmix_to_mono(struct a2dp_stream_out *out, const uint8_t *buffer,
577 const int16_t *input = (const void *) buffer;
578 int16_t *output = (void *) out->downmix_buf;
581 /* PCM 16bit stereo */
582 frames = bytes / (2 * sizeof(int16_t));
584 for (i = 0; i < frames; i++) {
585 int16_t l = le16_to_cpu(get_unaligned(&input[i * 2]));
586 int16_t r = le16_to_cpu(get_unaligned(&input[i * 2 + 1]));
588 put_unaligned(cpu_to_le16((l + r) / 2), &output[i]);
592 static bool wait_for_endpoint(struct audio_endpoint *ep, bool *writable)
597 struct pollfd pollfd;
600 pollfd.events = POLLOUT;
603 ret = poll(&pollfd, 1, 500);
606 *writable = !!(pollfd.revents & POLLOUT);
610 if (errno != EINTR) {
612 error("poll failed (%d)", ret);
620 static bool write_to_endpoint(struct audio_endpoint *ep, size_t bytes)
622 struct media_packet *mp = (struct media_packet *) ep->mp;
626 ret = write(ep->fd, mp, bytes);
632 * this should not happen so let's issue warning, but do not
633 * fail, we can try to write next packet
635 if (errno == EAGAIN) {
637 warn("write failed (%d)", ret);
641 if (errno != EINTR) {
643 error("write failed (%d)", ret);
651 static bool write_data(struct a2dp_stream_out *out, const void *buffer,
654 struct audio_endpoint *ep = out->ep;
655 struct media_packet *mp = (struct media_packet *) ep->mp;
656 struct media_packet_rtp *mp_rtp = (struct media_packet_rtp *) ep->mp;
657 size_t free_space = ep->mp_data_len;
660 while (consumed < bytes) {
665 struct timespec current;
666 uint64_t audio_sent, audio_passed;
667 bool do_write = false;
670 * prepare media packet in advance so we don't waste time after
673 if (ep->codec->use_rtp) {
674 mp_rtp->hdr.sequence_number = htons(ep->seq++);
675 mp_rtp->hdr.timestamp = htonl(ep->samples);
677 read = ep->codec->encode_mediapacket(ep->codec_data,
679 bytes - consumed, mp,
680 free_space, &written);
683 * not much we can do here, let's just ignore remaining
689 /* calculate where are we and where we should be */
690 clock_gettime(CLOCK_MONOTONIC, ¤t);
692 memcpy(&ep->start, ¤t, sizeof(ep->start));
693 audio_sent = ep->samples * 1000000ll / out->cfg.rate;
694 audio_passed = timespec_diff_us(¤t, &ep->start);
697 * if we're ahead of stream then wait for next write point,
698 * if we're lagging more than 100ms then stop writing and just
699 * skip data until we're back in sync
701 if (audio_sent > audio_passed) {
702 struct timespec anchor;
706 timespec_add(&ep->start, audio_sent, &anchor);
709 ret = clock_nanosleep(CLOCK_MONOTONIC,
710 TIMER_ABSTIME, &anchor,
717 error("clock_nanosleep failed (%d)",
722 } else if (!ep->resync) {
723 uint64_t diff = audio_passed - audio_sent;
725 if (diff > MAX_DELAY) {
726 warn("lag is %jums, resyncing", diff / 1000);
728 ep->codec->update_qos(ep->codec_data,
729 QOS_POLICY_DECREASE);
734 /* we send data only in case codec encoded some data, i.e. some
735 * codecs do internal buffering and output data only if full
736 * frame can be encoded
737 * in resync mode we'll just drop mediapackets
739 if (written > 0 && !ep->resync) {
740 /* wait some time for socket to be ready for write,
741 * but we'll just skip writing data if timeout occurs
743 if (!wait_for_endpoint(ep, &do_write))
747 if (ep->codec->use_rtp)
748 written += sizeof(struct rtp_header);
750 if (!write_to_endpoint(ep, written))
756 * AudioFlinger provides 16bit PCM, so sample size is 2 bytes
757 * multiplied by number of channels. Number of channels is
758 * simply number of bits set in channels mask.
760 samples = read / (2 * popcount(out->cfg.channels));
761 ep->samples += samples;
768 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
771 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
772 const void *in_buf = buffer;
773 size_t in_len = bytes;
775 /* just return in case we're closing */
776 if (out->audio_state == AUDIO_A2DP_STATE_NONE)
779 /* We can auto-start only from standby */
780 if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
781 DBG("stream in standby, auto-start");
783 if (!resume_endpoint(out->ep))
786 out->audio_state = AUDIO_A2DP_STATE_STARTED;
789 if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
790 error("audio: stream not started");
794 if (out->ep->fd < 0) {
795 error("audio: no transport socket");
800 * currently Android audioflinger is not able to provide mono stream on
801 * A2DP output so down mixing needs to be done in hal-audio plugin.
804 * AudioFlinger::PlaybackThread::readOutputParameters()
805 * frameworks/av/services/audioflinger/Threads.cpp:1631
807 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
808 if (!out->downmix_buf) {
809 error("audio: downmix buffer not initialized");
813 downmix_to_mono(out, buffer, bytes);
815 in_buf = out->downmix_buf;
819 if (!write_data(out, in_buf, in_len))
825 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
827 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
831 return out->cfg.rate;
834 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
836 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
840 if (rate != out->cfg.rate) {
841 warn("audio: cannot set sample rate to %d", rate);
848 static size_t out_get_buffer_size(const struct audio_stream *stream)
853 * We should return proper buffer size calculated by codec (so each
854 * input buffer is encoded into single media packed) but this does not
855 * work well with AudioFlinger and causes problems. For this reason we
856 * use magic value here and out_write code takes care of splitting
857 * input buffer into multiple media packets.
859 return FIXED_BUFFER_SIZE;
862 static uint32_t out_get_channels(const struct audio_stream *stream)
867 * AudioFlinger can only provide stereo stream, so we return it here and
868 * later we'll downmix this to mono in case codec requires it
871 return AUDIO_CHANNEL_OUT_STEREO;
874 static audio_format_t out_get_format(const struct audio_stream *stream)
876 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
880 return out->cfg.format;
883 static int out_set_format(struct audio_stream *stream, audio_format_t format)
889 static int out_standby(struct audio_stream *stream)
891 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
895 if (out->audio_state == AUDIO_A2DP_STATE_STARTED) {
896 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
898 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
904 static int out_dump(const struct audio_stream *stream, int fd)
910 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
912 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
916 bool enter_suspend = false;
917 bool exit_suspend = false;
921 str = strdup(kvpairs);
925 kvpair = strtok_r(str, ";", &saveptr);
927 for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
930 keyval = strchr(kvpair, '=');
937 if (!strcmp(kvpair, "closing")) {
938 if (!strcmp(keyval, "true"))
939 out->audio_state = AUDIO_A2DP_STATE_NONE;
940 } else if (!strcmp(kvpair, "A2dpSuspended")) {
941 if (!strcmp(keyval, "true"))
942 enter_suspend = true;
950 if (enter_suspend && out->audio_state == AUDIO_A2DP_STATE_STARTED) {
951 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
953 out->audio_state = AUDIO_A2DP_STATE_SUSPENDED;
956 if (exit_suspend && out->audio_state == AUDIO_A2DP_STATE_SUSPENDED)
957 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
962 static char *out_get_parameters(const struct audio_stream *stream,
969 static uint32_t out_get_latency(const struct audio_stream_out *stream)
971 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
972 struct audio_endpoint *ep = out->ep;
977 pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
979 return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
982 static int out_set_volume(struct audio_stream_out *stream, float left,
986 /* volume controlled in audioflinger mixer (digital) */
990 static int out_get_render_position(const struct audio_stream_out *stream,
991 uint32_t *dsp_frames)
997 static int out_add_audio_effect(const struct audio_stream *stream,
998 effect_handle_t effect)
1004 static int out_remove_audio_effect(const struct audio_stream *stream,
1005 effect_handle_t effect)
1011 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1017 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1023 static size_t in_get_buffer_size(const struct audio_stream *stream)
1029 static uint32_t in_get_channels(const struct audio_stream *stream)
1035 static audio_format_t in_get_format(const struct audio_stream *stream)
1041 static int in_set_format(struct audio_stream *stream, audio_format_t format)
1047 static int in_standby(struct audio_stream *stream)
1053 static int in_dump(const struct audio_stream *stream, int fd)
1059 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1065 static char *in_get_parameters(const struct audio_stream *stream,
1072 static int in_set_gain(struct audio_stream_in *stream, float gain)
1078 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1085 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1091 static int in_add_audio_effect(const struct audio_stream *stream,
1092 effect_handle_t effect)
1098 static int in_remove_audio_effect(const struct audio_stream *stream,
1099 effect_handle_t effect)
1105 static int audio_open_output_stream(struct audio_hw_device *dev,
1106 audio_io_handle_t handle,
1107 audio_devices_t devices,
1108 audio_output_flags_t flags,
1109 struct audio_config *config,
1110 struct audio_stream_out **stream_out)
1113 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1114 struct a2dp_stream_out *out;
1116 out = calloc(1, sizeof(struct a2dp_stream_out));
1122 out->stream.common.get_sample_rate = out_get_sample_rate;
1123 out->stream.common.set_sample_rate = out_set_sample_rate;
1124 out->stream.common.get_buffer_size = out_get_buffer_size;
1125 out->stream.common.get_channels = out_get_channels;
1126 out->stream.common.get_format = out_get_format;
1127 out->stream.common.set_format = out_set_format;
1128 out->stream.common.standby = out_standby;
1129 out->stream.common.dump = out_dump;
1130 out->stream.common.set_parameters = out_set_parameters;
1131 out->stream.common.get_parameters = out_get_parameters;
1132 out->stream.common.add_audio_effect = out_add_audio_effect;
1133 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1134 out->stream.get_latency = out_get_latency;
1135 out->stream.set_volume = out_set_volume;
1136 out->stream.write = out_write;
1137 out->stream.get_render_position = out_get_render_position;
1139 /* We want to autoselect opened endpoint */
1142 if (!open_endpoint(&out->ep, &out->cfg))
1145 DBG("rate=%d channels=%d format=%d", out->cfg.rate,
1146 out->cfg.channels, out->cfg.format);
1148 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1149 out->downmix_buf = malloc(FIXED_BUFFER_SIZE / 2);
1150 if (!out->downmix_buf)
1154 *stream_out = &out->stream;
1155 a2dp_dev->out = out;
1157 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1162 error("audio: cannot open output stream");
1168 static void audio_close_output_stream(struct audio_hw_device *dev,
1169 struct audio_stream_out *stream)
1171 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1172 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1176 close_endpoint(a2dp_dev->out->ep);
1178 free(out->downmix_buf);
1181 a2dp_dev->out = NULL;
1184 static int audio_set_parameters(struct audio_hw_device *dev,
1185 const char *kvpairs)
1187 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1188 struct a2dp_stream_out *out = a2dp_dev->out;
1195 return out->stream.common.set_parameters((struct audio_stream *) out,
1199 static char *audio_get_parameters(const struct audio_hw_device *dev,
1206 static int audio_init_check(const struct audio_hw_device *dev)
1212 static int audio_set_voice_volume(struct audio_hw_device *dev, float volume)
1218 static int audio_set_master_volume(struct audio_hw_device *dev, float volume)
1224 static int audio_set_mode(struct audio_hw_device *dev, int mode)
1230 static int audio_set_mic_mute(struct audio_hw_device *dev, bool state)
1236 static int audio_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1242 static size_t audio_get_input_buffer_size(const struct audio_hw_device *dev,
1243 const struct audio_config *config)
1249 static int audio_open_input_stream(struct audio_hw_device *dev,
1250 audio_io_handle_t handle,
1251 audio_devices_t devices,
1252 struct audio_config *config,
1253 struct audio_stream_in **stream_in)
1255 struct audio_stream_in *in;
1259 in = calloc(1, sizeof(struct audio_stream_in));
1263 in->common.get_sample_rate = in_get_sample_rate;
1264 in->common.set_sample_rate = in_set_sample_rate;
1265 in->common.get_buffer_size = in_get_buffer_size;
1266 in->common.get_channels = in_get_channels;
1267 in->common.get_format = in_get_format;
1268 in->common.set_format = in_set_format;
1269 in->common.standby = in_standby;
1270 in->common.dump = in_dump;
1271 in->common.set_parameters = in_set_parameters;
1272 in->common.get_parameters = in_get_parameters;
1273 in->common.add_audio_effect = in_add_audio_effect;
1274 in->common.remove_audio_effect = in_remove_audio_effect;
1275 in->set_gain = in_set_gain;
1277 in->get_input_frames_lost = in_get_input_frames_lost;
1284 static void audio_close_input_stream(struct audio_hw_device *dev,
1285 struct audio_stream_in *stream_in)
1291 static int audio_dump(const audio_hw_device_t *device, int fd)
1297 static void unload_codec(void *data)
1299 struct audio_codec *codec = data;
1305 static int audio_close(hw_device_t *device)
1307 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
1311 unregister_endpoints();
1313 queue_destroy(loaded_codecs, unload_codec);
1314 loaded_codecs = NULL;
1316 shutdown(listen_sk, SHUT_RDWR);
1317 shutdown(audio_sk, SHUT_RDWR);
1319 pthread_join(ipc_th, NULL);
1328 static void *ipc_handler(void *data)
1337 DBG("Waiting for connection ...");
1339 sk = accept(listen_sk, NULL, NULL);
1346 if (err != ECONNABORTED && err != EINVAL)
1347 error("audio: Failed to accept socket: %d (%s)",
1348 err, strerror(err));
1353 pthread_mutex_lock(&sk_mutex);
1355 pthread_mutex_unlock(&sk_mutex);
1357 DBG("Audio IPC: Connected");
1359 if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
1360 error("audio: Failed to register endpoints");
1362 unregister_endpoints();
1364 pthread_mutex_lock(&sk_mutex);
1365 shutdown(audio_sk, SHUT_RDWR);
1368 pthread_mutex_unlock(&sk_mutex);
1373 memset(&pfd, 0, sizeof(pfd));
1375 pfd.events = POLLHUP | POLLERR | POLLNVAL;
1377 /* Check if socket is still alive. Empty while loop.*/
1378 while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
1380 if (pfd.revents & (POLLHUP | POLLERR | POLLNVAL)) {
1381 info("Audio HAL: Socket closed");
1383 pthread_mutex_lock(&sk_mutex);
1386 pthread_mutex_unlock(&sk_mutex);
1390 /* audio_sk is closed at this point, just cleanup endpoints states */
1391 memset(audio_endpoints, 0, sizeof(audio_endpoints));
1393 info("Closing Audio IPC thread");
1397 static int audio_ipc_init(void)
1399 struct sockaddr_un addr;
1405 sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
1408 error("audio: Failed to create socket: %d (%s)", -err,
1413 memset(&addr, 0, sizeof(addr));
1414 addr.sun_family = AF_UNIX;
1416 memcpy(addr.sun_path, BLUEZ_AUDIO_SK_PATH,
1417 sizeof(BLUEZ_AUDIO_SK_PATH));
1419 if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
1421 error("audio: Failed to bind socket: %d (%s)", -err,
1426 if (listen(sk, 1) < 0) {
1428 error("audio: Failed to listen on the socket: %d (%s)", -err,
1435 err = pthread_create(&ipc_th, NULL, ipc_handler, NULL);
1439 error("audio: Failed to start Audio IPC thread: %d (%s)",
1440 -err, strerror(-err));
1451 static int audio_open(const hw_module_t *module, const char *name,
1452 hw_device_t **device)
1454 struct a2dp_audio_dev *a2dp_dev;
1460 if (strcmp(name, AUDIO_HARDWARE_INTERFACE)) {
1461 error("audio: interface %s not matching [%s]", name,
1462 AUDIO_HARDWARE_INTERFACE);
1466 err = audio_ipc_init();
1470 a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
1474 a2dp_dev->dev.common.tag = HARDWARE_DEVICE_TAG;
1475 a2dp_dev->dev.common.version = AUDIO_DEVICE_API_VERSION_CURRENT;
1476 a2dp_dev->dev.common.module = (struct hw_module_t *) module;
1477 a2dp_dev->dev.common.close = audio_close;
1479 a2dp_dev->dev.init_check = audio_init_check;
1480 a2dp_dev->dev.set_voice_volume = audio_set_voice_volume;
1481 a2dp_dev->dev.set_master_volume = audio_set_master_volume;
1482 a2dp_dev->dev.set_mode = audio_set_mode;
1483 a2dp_dev->dev.set_mic_mute = audio_set_mic_mute;
1484 a2dp_dev->dev.get_mic_mute = audio_get_mic_mute;
1485 a2dp_dev->dev.set_parameters = audio_set_parameters;
1486 a2dp_dev->dev.get_parameters = audio_get_parameters;
1487 a2dp_dev->dev.get_input_buffer_size = audio_get_input_buffer_size;
1488 a2dp_dev->dev.open_output_stream = audio_open_output_stream;
1489 a2dp_dev->dev.close_output_stream = audio_close_output_stream;
1490 a2dp_dev->dev.open_input_stream = audio_open_input_stream;
1491 a2dp_dev->dev.close_input_stream = audio_close_input_stream;
1492 a2dp_dev->dev.dump = audio_dump;
1494 loaded_codecs = queue_new();
1496 for (i = 0; i < NUM_CODECS; i++) {
1497 audio_codec_get_t get_codec = audio_codecs[i];
1498 const struct audio_codec *codec = get_codec();
1500 if (codec->load && !codec->load())
1503 queue_push_tail(loaded_codecs, (void *) codec);
1507 * Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
1508 * This results from the structure of following structs:a2dp_audio_dev,
1509 * audio_hw_device. We will rely on this later in the code.
1511 *device = &a2dp_dev->dev.common;
1516 static struct hw_module_methods_t hal_module_methods = {
1520 struct audio_module HAL_MODULE_INFO_SYM = {
1522 .tag = HARDWARE_MODULE_TAG,
1525 .id = AUDIO_HARDWARE_MODULE_ID,
1526 .name = "A2DP Bluez HW HAL",
1527 .author = "Intel Corporation",
1528 .methods = &hal_module_methods,