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 "hal-utils.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);
101 const audio_codec_get_t get_codec;
104 { .get_codec = codec_aptx, .loaded = false },
105 { .get_codec = codec_sbc, .loaded = false },
108 #define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
110 #define MAX_AUDIO_ENDPOINTS NUM_CODECS
112 struct audio_endpoint {
114 const struct audio_codec *codec;
118 struct media_packet *mp;
123 struct timespec start;
128 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
131 AUDIO_A2DP_STATE_NONE,
132 AUDIO_A2DP_STATE_STANDBY,
133 AUDIO_A2DP_STATE_SUSPENDED,
134 AUDIO_A2DP_STATE_STARTED
137 struct a2dp_stream_out {
138 struct audio_stream_out stream;
140 struct audio_endpoint *ep;
141 enum a2dp_state_t audio_state;
142 struct audio_input_config cfg;
144 uint8_t *downmix_buf;
147 struct a2dp_audio_dev {
148 struct audio_hw_device dev;
149 struct a2dp_stream_out *out;
152 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
153 void *param, size_t *rsp_len, void *rsp, int *fd)
159 char cmsgbuf[CMSG_SPACE(sizeof(int))];
161 size_t s_len = sizeof(s);
163 pthread_mutex_lock(&sk_mutex);
166 error("audio: Invalid cmd socket passed to audio_ipc_cmd");
170 if (!rsp || !rsp_len) {
171 memset(&s, 0, s_len);
176 memset(&msg, 0, sizeof(msg));
177 memset(&cmd, 0, sizeof(cmd));
179 cmd.service_id = service_id;
183 iv[0].iov_base = &cmd;
184 iv[0].iov_len = sizeof(cmd);
186 iv[1].iov_base = param;
192 ret = sendmsg(audio_sk, &msg, 0);
194 error("audio: Sending command failed:%s", strerror(errno));
198 /* socket was shutdown */
200 error("audio: Command socket closed");
204 memset(&msg, 0, sizeof(msg));
205 memset(&cmd, 0, sizeof(cmd));
207 iv[0].iov_base = &cmd;
208 iv[0].iov_len = sizeof(cmd);
210 iv[1].iov_base = rsp;
211 iv[1].iov_len = *rsp_len;
217 memset(cmsgbuf, 0, sizeof(cmsgbuf));
218 msg.msg_control = cmsgbuf;
219 msg.msg_controllen = sizeof(cmsgbuf);
222 ret = recvmsg(audio_sk, &msg, 0);
224 error("audio: Receiving command response failed:%s",
229 if (ret < (ssize_t) sizeof(cmd)) {
230 error("audio: Too small response received(%zd bytes)", ret);
234 if (cmd.service_id != service_id) {
235 error("audio: Invalid service id (%u vs %u)", cmd.service_id,
240 if (ret != (ssize_t) (sizeof(cmd) + cmd.len)) {
241 error("audio: Malformed response received(%zd bytes)", ret);
245 if (cmd.opcode != opcode && cmd.opcode != AUDIO_OP_STATUS) {
246 error("audio: Invalid opcode received (%u vs %u)",
251 if (cmd.opcode == AUDIO_OP_STATUS) {
252 struct ipc_status *s = rsp;
254 if (sizeof(*s) != cmd.len) {
255 error("audio: Invalid status length");
259 if (s->code == AUDIO_STATUS_SUCCESS) {
260 error("audio: Invalid success status response");
264 pthread_mutex_unlock(&sk_mutex);
269 pthread_mutex_unlock(&sk_mutex);
271 /* Receive auxiliary data in msg */
273 struct cmsghdr *cmsg;
277 for (cmsg = CMSG_FIRSTHDR(&msg); cmsg;
278 cmsg = CMSG_NXTHDR(&msg, cmsg)) {
279 if (cmsg->cmsg_level == SOL_SOCKET
280 && cmsg->cmsg_type == SCM_RIGHTS) {
281 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(const struct audio_codec *codec,
428 struct register_state *state)
430 struct audio_endpoint *ep = state->ep;
432 /* don't even try to register more endpoints if one failed */
436 ep->id = ipc_open_cmd(codec);
440 error("Failed to register endpoint");
445 ep->codec_data = NULL;
451 static int register_endpoints(void)
453 struct register_state state;
456 state.ep = &audio_endpoints[0];
459 for (i = 0; i < NUM_CODECS; i++) {
460 const struct audio_codec *codec = audio_codecs[i].get_codec();
462 if (!audio_codecs[i].loaded)
465 register_endpoint(codec, &state);
468 return state.error ? AUDIO_STATUS_FAILED : AUDIO_STATUS_SUCCESS;
471 static void unregister_endpoints(void)
475 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++) {
476 struct audio_endpoint *ep = &audio_endpoints[i];
479 ipc_close_cmd(ep->id);
480 memset(ep, 0, sizeof(*ep));
485 static bool open_endpoint(struct audio_endpoint **epp,
486 struct audio_input_config *cfg)
488 struct audio_preset *preset;
489 struct audio_endpoint *ep = *epp;
490 const struct audio_codec *codec;
492 uint16_t payload_len;
500 if (ipc_open_stream_cmd(&ep_id, &mtu, &fd, &preset) !=
501 AUDIO_STATUS_SUCCESS)
504 DBG("ep_id=%d mtu=%u", ep_id, mtu);
506 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++)
507 if (audio_endpoints[i].id == ep_id) {
508 ep = &audio_endpoints[i];
513 error("Cound not find opened endpoint");
520 if (ep->codec->use_rtp)
521 payload_len -= sizeof(struct rtp_header);
526 codec->init(preset, payload_len, &ep->codec_data);
527 codec->get_config(ep->codec_data, cfg);
529 ep->mp = calloc(mtu, 1);
533 if (ep->codec->use_rtp) {
534 struct media_packet_rtp *mp_rtp =
535 (struct media_packet_rtp *) ep->mp;
537 mp_rtp->hdr.pt = 0x60;
538 mp_rtp->hdr.ssrc = htonl(1);
541 ep->mp_data_len = payload_len;
554 static void close_endpoint(struct audio_endpoint *ep)
556 ipc_close_stream_cmd(ep->id);
564 ep->codec->cleanup(ep->codec_data);
565 ep->codec_data = NULL;
568 static bool resume_endpoint(struct audio_endpoint *ep)
570 if (ipc_resume_stream_cmd(ep->id) != AUDIO_STATUS_SUCCESS)
576 ep->codec->update_qos(ep->codec_data, QOS_POLICY_DEFAULT);
581 static void downmix_to_mono(struct a2dp_stream_out *out, const uint8_t *buffer,
584 const int16_t *input = (const void *) buffer;
585 int16_t *output = (void *) out->downmix_buf;
588 /* PCM 16bit stereo */
589 frames = bytes / (2 * sizeof(int16_t));
591 for (i = 0; i < frames; i++) {
592 int16_t l = get_le16(&input[i * 2]);
593 int16_t r = get_le16(&input[i * 2 + 1]);
595 put_le16((l + r) / 2, &output[i]);
599 static bool wait_for_endpoint(struct audio_endpoint *ep, bool *writable)
604 struct pollfd pollfd;
607 pollfd.events = POLLOUT;
610 ret = poll(&pollfd, 1, 500);
613 *writable = !!(pollfd.revents & POLLOUT);
617 if (errno != EINTR) {
619 error("poll failed (%d)", ret);
627 static bool write_to_endpoint(struct audio_endpoint *ep, size_t bytes)
629 struct media_packet *mp = (struct media_packet *) ep->mp;
633 ret = write(ep->fd, mp, bytes);
639 * this should not happen so let's issue warning, but do not
640 * fail, we can try to write next packet
642 if (errno == EAGAIN) {
644 warn("write failed (%d)", ret);
648 if (errno != EINTR) {
650 error("write failed (%d)", ret);
658 static bool write_data(struct a2dp_stream_out *out, const void *buffer,
661 struct audio_endpoint *ep = out->ep;
662 struct media_packet *mp = (struct media_packet *) ep->mp;
663 struct media_packet_rtp *mp_rtp = (struct media_packet_rtp *) ep->mp;
664 size_t free_space = ep->mp_data_len;
667 while (consumed < bytes) {
672 struct timespec current;
673 uint64_t audio_sent, audio_passed;
674 bool do_write = false;
677 * prepare media packet in advance so we don't waste time after
680 if (ep->codec->use_rtp) {
681 mp_rtp->hdr.sequence_number = htons(ep->seq++);
682 mp_rtp->hdr.timestamp = htonl(ep->samples);
684 read = ep->codec->encode_mediapacket(ep->codec_data,
686 bytes - consumed, mp,
687 free_space, &written);
690 * not much we can do here, let's just ignore remaining
696 /* calculate where are we and where we should be */
697 clock_gettime(CLOCK_MONOTONIC, ¤t);
699 memcpy(&ep->start, ¤t, sizeof(ep->start));
700 audio_sent = ep->samples * 1000000ll / out->cfg.rate;
701 audio_passed = timespec_diff_us(¤t, &ep->start);
704 * if we're ahead of stream then wait for next write point,
705 * if we're lagging more than 100ms then stop writing and just
706 * skip data until we're back in sync
708 if (audio_sent > audio_passed) {
709 struct timespec anchor;
713 timespec_add(&ep->start, audio_sent, &anchor);
716 ret = clock_nanosleep(CLOCK_MONOTONIC,
717 TIMER_ABSTIME, &anchor,
724 error("clock_nanosleep failed (%d)",
729 } else if (!ep->resync) {
730 uint64_t diff = audio_passed - audio_sent;
732 if (diff > MAX_DELAY) {
733 warn("lag is %jums, resyncing", diff / 1000);
735 ep->codec->update_qos(ep->codec_data,
736 QOS_POLICY_DECREASE);
741 /* we send data only in case codec encoded some data, i.e. some
742 * codecs do internal buffering and output data only if full
743 * frame can be encoded
744 * in resync mode we'll just drop mediapackets
746 if (written > 0 && !ep->resync) {
747 /* wait some time for socket to be ready for write,
748 * but we'll just skip writing data if timeout occurs
750 if (!wait_for_endpoint(ep, &do_write))
754 if (ep->codec->use_rtp)
755 written += sizeof(struct rtp_header);
757 if (!write_to_endpoint(ep, written))
763 * AudioFlinger provides 16bit PCM, so sample size is 2 bytes
764 * multiplied by number of channels. Number of channels is
765 * simply number of bits set in channels mask.
767 samples = read / (2 * popcount(out->cfg.channels));
768 ep->samples += samples;
775 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
778 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
779 const void *in_buf = buffer;
780 size_t in_len = bytes;
782 /* just return in case we're closing */
783 if (out->audio_state == AUDIO_A2DP_STATE_NONE)
786 /* We can auto-start only from standby */
787 if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
788 DBG("stream in standby, auto-start");
790 if (!resume_endpoint(out->ep))
793 out->audio_state = AUDIO_A2DP_STATE_STARTED;
796 if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
797 error("audio: stream not started");
801 if (out->ep->fd < 0) {
802 error("audio: no transport socket");
807 * currently Android audioflinger is not able to provide mono stream on
808 * A2DP output so down mixing needs to be done in hal-audio plugin.
811 * AudioFlinger::PlaybackThread::readOutputParameters()
812 * frameworks/av/services/audioflinger/Threads.cpp:1631
814 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
815 if (!out->downmix_buf) {
816 error("audio: downmix buffer not initialized");
820 downmix_to_mono(out, buffer, bytes);
822 in_buf = out->downmix_buf;
826 if (!write_data(out, in_buf, in_len))
832 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
834 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
838 return out->cfg.rate;
841 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
843 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
847 if (rate != out->cfg.rate) {
848 warn("audio: cannot set sample rate to %d", rate);
855 static size_t out_get_buffer_size(const struct audio_stream *stream)
860 * We should return proper buffer size calculated by codec (so each
861 * input buffer is encoded into single media packed) but this does not
862 * work well with AudioFlinger and causes problems. For this reason we
863 * use magic value here and out_write code takes care of splitting
864 * input buffer into multiple media packets.
866 return FIXED_BUFFER_SIZE;
869 static uint32_t out_get_channels(const struct audio_stream *stream)
874 * AudioFlinger can only provide stereo stream, so we return it here and
875 * later we'll downmix this to mono in case codec requires it
878 return AUDIO_CHANNEL_OUT_STEREO;
881 static audio_format_t out_get_format(const struct audio_stream *stream)
883 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
887 return out->cfg.format;
890 static int out_set_format(struct audio_stream *stream, audio_format_t format)
896 static int out_standby(struct audio_stream *stream)
898 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
902 if (out->audio_state == AUDIO_A2DP_STATE_STARTED) {
903 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
905 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
911 static int out_dump(const struct audio_stream *stream, int fd)
917 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
919 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
923 bool enter_suspend = false;
924 bool exit_suspend = false;
928 str = strdup(kvpairs);
932 kvpair = strtok_r(str, ";", &saveptr);
934 for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
937 keyval = strchr(kvpair, '=');
944 if (!strcmp(kvpair, "closing")) {
945 if (!strcmp(keyval, "true"))
946 out->audio_state = AUDIO_A2DP_STATE_NONE;
947 } else if (!strcmp(kvpair, "A2dpSuspended")) {
948 if (!strcmp(keyval, "true"))
949 enter_suspend = true;
957 if (enter_suspend && out->audio_state == AUDIO_A2DP_STATE_STARTED) {
958 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
960 out->audio_state = AUDIO_A2DP_STATE_SUSPENDED;
963 if (exit_suspend && out->audio_state == AUDIO_A2DP_STATE_SUSPENDED)
964 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
969 static char *out_get_parameters(const struct audio_stream *stream,
976 static uint32_t out_get_latency(const struct audio_stream_out *stream)
978 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
979 struct audio_endpoint *ep = out->ep;
984 pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
986 return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
989 static int out_set_volume(struct audio_stream_out *stream, float left,
993 /* volume controlled in audioflinger mixer (digital) */
997 static int out_get_render_position(const struct audio_stream_out *stream,
998 uint32_t *dsp_frames)
1004 static int out_add_audio_effect(const struct audio_stream *stream,
1005 effect_handle_t effect)
1011 static int out_remove_audio_effect(const struct audio_stream *stream,
1012 effect_handle_t effect)
1018 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1024 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1030 static size_t in_get_buffer_size(const struct audio_stream *stream)
1036 static uint32_t in_get_channels(const struct audio_stream *stream)
1042 static audio_format_t in_get_format(const struct audio_stream *stream)
1048 static int in_set_format(struct audio_stream *stream, audio_format_t format)
1054 static int in_standby(struct audio_stream *stream)
1060 static int in_dump(const struct audio_stream *stream, int fd)
1066 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1072 static char *in_get_parameters(const struct audio_stream *stream,
1079 static int in_set_gain(struct audio_stream_in *stream, float gain)
1085 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1092 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1098 static int in_add_audio_effect(const struct audio_stream *stream,
1099 effect_handle_t effect)
1105 static int in_remove_audio_effect(const struct audio_stream *stream,
1106 effect_handle_t effect)
1112 static int audio_open_output_stream_real(struct audio_hw_device *dev,
1113 audio_io_handle_t handle,
1114 audio_devices_t devices,
1115 audio_output_flags_t flags,
1116 struct audio_config *config,
1117 struct audio_stream_out **stream_out,
1118 const char *address)
1120 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1121 struct a2dp_stream_out *out;
1123 out = calloc(1, sizeof(struct a2dp_stream_out));
1129 out->stream.common.get_sample_rate = out_get_sample_rate;
1130 out->stream.common.set_sample_rate = out_set_sample_rate;
1131 out->stream.common.get_buffer_size = out_get_buffer_size;
1132 out->stream.common.get_channels = out_get_channels;
1133 out->stream.common.get_format = out_get_format;
1134 out->stream.common.set_format = out_set_format;
1135 out->stream.common.standby = out_standby;
1136 out->stream.common.dump = out_dump;
1137 out->stream.common.set_parameters = out_set_parameters;
1138 out->stream.common.get_parameters = out_get_parameters;
1139 out->stream.common.add_audio_effect = out_add_audio_effect;
1140 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1141 out->stream.get_latency = out_get_latency;
1142 out->stream.set_volume = out_set_volume;
1143 out->stream.write = out_write;
1144 out->stream.get_render_position = out_get_render_position;
1146 /* We want to autoselect opened endpoint */
1149 if (!open_endpoint(&out->ep, &out->cfg))
1152 DBG("rate=%d channels=%d format=%d", out->cfg.rate,
1153 out->cfg.channels, out->cfg.format);
1155 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1156 out->downmix_buf = malloc(FIXED_BUFFER_SIZE / 2);
1157 if (!out->downmix_buf)
1161 *stream_out = &out->stream;
1162 a2dp_dev->out = out;
1164 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1169 error("audio: cannot open output stream");
1175 #if ANDROID_VERSION >= PLATFORM_VER(5, 0, 0)
1176 static int audio_open_output_stream(struct audio_hw_device *dev,
1177 audio_io_handle_t handle,
1178 audio_devices_t devices,
1179 audio_output_flags_t flags,
1180 struct audio_config *config,
1181 struct audio_stream_out **stream_out,
1182 const char *address)
1184 return audio_open_output_stream_real(dev, handle, devices, flags,
1185 config, stream_out, address);
1188 static int audio_open_output_stream(struct audio_hw_device *dev,
1189 audio_io_handle_t handle,
1190 audio_devices_t devices,
1191 audio_output_flags_t flags,
1192 struct audio_config *config,
1193 struct audio_stream_out **stream_out)
1195 return audio_open_output_stream_real(dev, handle, devices, flags,
1196 config, stream_out, NULL);
1200 static void audio_close_output_stream(struct audio_hw_device *dev,
1201 struct audio_stream_out *stream)
1203 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1204 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1208 close_endpoint(a2dp_dev->out->ep);
1210 free(out->downmix_buf);
1213 a2dp_dev->out = NULL;
1216 static int audio_set_parameters(struct audio_hw_device *dev,
1217 const char *kvpairs)
1219 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1220 struct a2dp_stream_out *out = a2dp_dev->out;
1227 return out->stream.common.set_parameters((struct audio_stream *) out,
1231 static char *audio_get_parameters(const struct audio_hw_device *dev,
1238 static int audio_init_check(const struct audio_hw_device *dev)
1244 static int audio_set_voice_volume(struct audio_hw_device *dev, float volume)
1250 static int audio_set_master_volume(struct audio_hw_device *dev, float volume)
1256 static int audio_set_mode(struct audio_hw_device *dev, int mode)
1262 static int audio_set_mic_mute(struct audio_hw_device *dev, bool state)
1268 static int audio_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1274 static size_t audio_get_input_buffer_size(const struct audio_hw_device *dev,
1275 const struct audio_config *config)
1281 static int audio_open_input_stream_real(struct audio_hw_device *dev,
1282 audio_io_handle_t handle,
1283 audio_devices_t devices,
1284 struct audio_config *config,
1285 struct audio_stream_in **stream_in,
1286 audio_input_flags_t flags,
1287 const char *address,
1288 audio_source_t source)
1290 struct audio_stream_in *in;
1294 in = calloc(1, sizeof(struct audio_stream_in));
1298 in->common.get_sample_rate = in_get_sample_rate;
1299 in->common.set_sample_rate = in_set_sample_rate;
1300 in->common.get_buffer_size = in_get_buffer_size;
1301 in->common.get_channels = in_get_channels;
1302 in->common.get_format = in_get_format;
1303 in->common.set_format = in_set_format;
1304 in->common.standby = in_standby;
1305 in->common.dump = in_dump;
1306 in->common.set_parameters = in_set_parameters;
1307 in->common.get_parameters = in_get_parameters;
1308 in->common.add_audio_effect = in_add_audio_effect;
1309 in->common.remove_audio_effect = in_remove_audio_effect;
1310 in->set_gain = in_set_gain;
1312 in->get_input_frames_lost = in_get_input_frames_lost;
1319 #if ANDROID_VERSION >= PLATFORM_VER(5, 0, 0)
1320 static int audio_open_input_stream(struct audio_hw_device *dev,
1321 audio_io_handle_t handle,
1322 audio_devices_t devices,
1323 struct audio_config *config,
1324 struct audio_stream_in **stream_in,
1325 audio_input_flags_t flags,
1326 const char *address,
1327 audio_source_t source)
1329 return audio_open_input_stream_real(dev, handle, devices, config,
1330 stream_in, flags, address,
1334 static int audio_open_input_stream(struct audio_hw_device *dev,
1335 audio_io_handle_t handle,
1336 audio_devices_t devices,
1337 struct audio_config *config,
1338 struct audio_stream_in **stream_in)
1340 return audio_open_input_stream_real(dev, handle, devices, config,
1341 stream_in, 0, NULL, 0);
1345 static void audio_close_input_stream(struct audio_hw_device *dev,
1346 struct audio_stream_in *stream_in)
1352 static int audio_dump(const audio_hw_device_t *device, int fd)
1358 #if ANDROID_VERSION >= PLATFORM_VER(5, 0, 0)
1359 static int set_master_mute(struct audio_hw_device *dev, bool mute)
1365 static int get_master_mute(struct audio_hw_device *dev, bool *mute)
1371 static int create_audio_patch(struct audio_hw_device *dev,
1372 unsigned int num_sources,
1373 const struct audio_port_config *sources,
1374 unsigned int num_sinks,
1375 const struct audio_port_config *sinks,
1376 audio_patch_handle_t *handle)
1382 static int release_audio_patch(struct audio_hw_device *dev,
1383 audio_patch_handle_t handle)
1389 static int get_audio_port(struct audio_hw_device *dev, struct audio_port *port)
1395 static int set_audio_port_config(struct audio_hw_device *dev,
1396 const struct audio_port_config *config)
1403 static int audio_close(hw_device_t *device)
1405 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
1410 unregister_endpoints();
1412 for (i = 0; i < NUM_CODECS; i++) {
1413 const struct audio_codec *codec = audio_codecs[i].get_codec();
1415 if (!audio_codecs[i].loaded)
1421 audio_codecs[i].loaded = false;
1424 shutdown(listen_sk, SHUT_RDWR);
1425 shutdown(audio_sk, SHUT_RDWR);
1427 pthread_join(ipc_th, NULL);
1436 static void *ipc_handler(void *data)
1445 DBG("Waiting for connection ...");
1447 sk = accept(listen_sk, NULL, NULL);
1454 if (err != ECONNABORTED && err != EINVAL)
1455 error("audio: Failed to accept socket: %d (%s)",
1456 err, strerror(err));
1461 pthread_mutex_lock(&sk_mutex);
1463 pthread_mutex_unlock(&sk_mutex);
1465 DBG("Audio IPC: Connected");
1467 if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
1468 error("audio: Failed to register endpoints");
1470 unregister_endpoints();
1472 pthread_mutex_lock(&sk_mutex);
1473 shutdown(audio_sk, SHUT_RDWR);
1476 pthread_mutex_unlock(&sk_mutex);
1481 memset(&pfd, 0, sizeof(pfd));
1483 pfd.events = POLLHUP | POLLERR | POLLNVAL;
1485 /* Check if socket is still alive. Empty while loop.*/
1486 while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
1488 info("Audio HAL: Socket closed");
1490 pthread_mutex_lock(&sk_mutex);
1493 pthread_mutex_unlock(&sk_mutex);
1496 /* audio_sk is closed at this point, just cleanup endpoints states */
1497 memset(audio_endpoints, 0, sizeof(audio_endpoints));
1499 info("Closing Audio IPC thread");
1503 static int audio_ipc_init(void)
1505 struct sockaddr_un addr;
1511 sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
1514 error("audio: Failed to create socket: %d (%s)", -err,
1519 memset(&addr, 0, sizeof(addr));
1520 addr.sun_family = AF_UNIX;
1522 memcpy(addr.sun_path, BLUEZ_AUDIO_SK_PATH,
1523 sizeof(BLUEZ_AUDIO_SK_PATH));
1525 if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
1527 error("audio: Failed to bind socket: %d (%s)", -err,
1532 if (listen(sk, 1) < 0) {
1534 error("audio: Failed to listen on the socket: %d (%s)", -err,
1541 err = pthread_create(&ipc_th, NULL, ipc_handler, NULL);
1545 error("audio: Failed to start Audio IPC thread: %d (%s)",
1546 -err, strerror(-err));
1557 static int audio_open(const hw_module_t *module, const char *name,
1558 hw_device_t **device)
1560 struct a2dp_audio_dev *a2dp_dev;
1566 if (strcmp(name, AUDIO_HARDWARE_INTERFACE)) {
1567 error("audio: interface %s not matching [%s]", name,
1568 AUDIO_HARDWARE_INTERFACE);
1572 err = audio_ipc_init();
1576 a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
1580 a2dp_dev->dev.common.tag = HARDWARE_DEVICE_TAG;
1581 a2dp_dev->dev.common.version = AUDIO_DEVICE_API_VERSION_CURRENT;
1582 a2dp_dev->dev.common.module = (struct hw_module_t *) module;
1583 a2dp_dev->dev.common.close = audio_close;
1585 a2dp_dev->dev.init_check = audio_init_check;
1586 a2dp_dev->dev.set_voice_volume = audio_set_voice_volume;
1587 a2dp_dev->dev.set_master_volume = audio_set_master_volume;
1588 a2dp_dev->dev.set_mode = audio_set_mode;
1589 a2dp_dev->dev.set_mic_mute = audio_set_mic_mute;
1590 a2dp_dev->dev.get_mic_mute = audio_get_mic_mute;
1591 a2dp_dev->dev.set_parameters = audio_set_parameters;
1592 a2dp_dev->dev.get_parameters = audio_get_parameters;
1593 a2dp_dev->dev.get_input_buffer_size = audio_get_input_buffer_size;
1594 a2dp_dev->dev.open_output_stream = audio_open_output_stream;
1595 a2dp_dev->dev.close_output_stream = audio_close_output_stream;
1596 a2dp_dev->dev.open_input_stream = audio_open_input_stream;
1597 a2dp_dev->dev.close_input_stream = audio_close_input_stream;
1598 a2dp_dev->dev.dump = audio_dump;
1599 #if ANDROID_VERSION >= PLATFORM_VER(5, 0, 0)
1600 a2dp_dev->dev.set_master_mute = set_master_mute;
1601 a2dp_dev->dev.get_master_mute = get_master_mute;
1602 a2dp_dev->dev.create_audio_patch = create_audio_patch;
1603 a2dp_dev->dev.release_audio_patch = release_audio_patch;
1604 a2dp_dev->dev.get_audio_port = get_audio_port;
1605 a2dp_dev->dev.set_audio_port_config = set_audio_port_config;
1608 for (i = 0; i < NUM_CODECS; i++) {
1609 const struct audio_codec *codec = audio_codecs[i].get_codec();
1611 if (codec->load && !codec->load())
1614 audio_codecs[i].loaded = true;
1618 * Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
1619 * This results from the structure of following structs:a2dp_audio_dev,
1620 * audio_hw_device. We will rely on this later in the code.
1622 *device = &a2dp_dev->dev.common;
1627 static struct hw_module_methods_t hal_module_methods = {
1631 struct audio_module HAL_MODULE_INFO_SYM = {
1633 .tag = HARDWARE_MODULE_TAG,
1636 .id = AUDIO_HARDWARE_MODULE_ID,
1637 .name = "A2DP Bluez HW HAL",
1638 .author = "Intel Corporation",
1639 .methods = &hal_module_methods,