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 static const uint8_t a2dp_src_uuid[] = {
49 0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
50 0x80, 0x00, 0x00, 0x80, 0x5f, 0x9b, 0x34, 0xfb };
52 static int listen_sk = -1;
53 static int audio_sk = -1;
55 static pthread_t ipc_th = 0;
56 static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
58 #if __BYTE_ORDER == __LITTLE_ENDIAN
69 uint16_t sequence_number;
73 } __attribute__ ((packed));
76 unsigned frame_count:4;
78 unsigned is_last_fragment:1;
79 unsigned is_first_fragment:1;
80 unsigned is_fragmented:1;
81 } __attribute__ ((packed));
83 #elif __BYTE_ORDER == __BIG_ENDIAN
94 uint16_t sequence_number;
98 } __attribute__ ((packed));
101 unsigned is_fragmented:1;
102 unsigned is_first_fragment:1;
103 unsigned is_last_fragment:1;
105 unsigned frame_count:4;
106 } __attribute__ ((packed));
109 #error "Unknown byte order"
112 struct media_packet {
113 struct rtp_header hdr;
114 struct rtp_payload payload;
118 struct audio_input_config {
121 audio_format_t format;
132 size_t out_frame_len;
134 unsigned frame_duration;
135 unsigned frames_per_packet;
138 static inline void timespec_diff(struct timespec *a, struct timespec *b,
139 struct timespec *res)
141 res->tv_sec = a->tv_sec - b->tv_sec;
142 res->tv_nsec = a->tv_nsec - b->tv_nsec;
144 if (res->tv_nsec < 0) {
146 res->tv_nsec += 1000000000; /* 1sec */
150 static int sbc_get_presets(struct audio_preset *preset, size_t *len);
151 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
153 static int sbc_cleanup(void *codec_data);
154 static int sbc_get_config(void *codec_data, struct audio_input_config *config);
155 static size_t sbc_get_buffer_size(void *codec_data);
156 static size_t sbc_get_mediapacket_duration(void *codec_data);
157 static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
158 size_t len, struct media_packet *mp,
159 size_t mp_data_len, size_t *written);
164 int (*get_presets) (struct audio_preset *preset, size_t *len);
166 int (*init) (struct audio_preset *preset, uint16_t mtu,
168 int (*cleanup) (void *codec_data);
169 int (*get_config) (void *codec_data,
170 struct audio_input_config *config);
171 size_t (*get_buffer_size) (void *codec_data);
172 size_t (*get_mediapacket_duration) (void *codec_data);
173 ssize_t (*encode_mediapacket) (void *codec_data, const uint8_t *buffer,
174 size_t len, struct media_packet *mp,
175 size_t mp_data_len, size_t *written);
178 static const struct audio_codec audio_codecs[] = {
180 .type = A2DP_CODEC_SBC,
182 .get_presets = sbc_get_presets,
184 .init = sbc_codec_init,
185 .cleanup = sbc_cleanup,
186 .get_config = sbc_get_config,
187 .get_buffer_size = sbc_get_buffer_size,
188 .get_mediapacket_duration = sbc_get_mediapacket_duration,
189 .encode_mediapacket = sbc_encode_mediapacket,
193 #define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
195 #define MAX_AUDIO_ENDPOINTS NUM_CODECS
197 struct audio_endpoint {
199 const struct audio_codec *codec;
203 struct media_packet *mp;
208 struct timespec start;
211 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
214 AUDIO_A2DP_STATE_NONE,
215 AUDIO_A2DP_STATE_STANDBY,
216 AUDIO_A2DP_STATE_SUSPENDED,
217 AUDIO_A2DP_STATE_STARTED
220 struct a2dp_stream_out {
221 struct audio_stream_out stream;
223 struct audio_endpoint *ep;
224 enum a2dp_state_t audio_state;
225 struct audio_input_config cfg;
227 uint8_t *downmix_buf;
230 struct a2dp_audio_dev {
231 struct audio_hw_device dev;
232 struct a2dp_stream_out *out;
235 static const a2dp_sbc_t sbc_presets[] = {
237 .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
238 .channel_mode = SBC_CHANNEL_MODE_MONO |
239 SBC_CHANNEL_MODE_DUAL_CHANNEL |
240 SBC_CHANNEL_MODE_STEREO |
241 SBC_CHANNEL_MODE_JOINT_STEREO,
242 .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
243 .allocation_method = SBC_ALLOCATION_SNR |
244 SBC_ALLOCATION_LOUDNESS,
245 .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
246 SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
247 .min_bitpool = MIN_BITPOOL,
248 .max_bitpool = MAX_BITPOOL
251 .frequency = SBC_SAMPLING_FREQ_44100,
252 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
253 .subbands = SBC_SUBBANDS_8,
254 .allocation_method = SBC_ALLOCATION_LOUDNESS,
255 .block_length = SBC_BLOCK_LENGTH_16,
256 .min_bitpool = MIN_BITPOOL,
257 .max_bitpool = MAX_BITPOOL
260 .frequency = SBC_SAMPLING_FREQ_48000,
261 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
262 .subbands = SBC_SUBBANDS_8,
263 .allocation_method = SBC_ALLOCATION_LOUDNESS,
264 .block_length = SBC_BLOCK_LENGTH_16,
265 .min_bitpool = MIN_BITPOOL,
266 .max_bitpool = MAX_BITPOOL
270 static int sbc_get_presets(struct audio_preset *preset, size_t *len)
275 uint8_t *ptr = (uint8_t *) preset;
276 size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
278 count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
280 for (i = 0; i < count; i++) {
281 preset = (struct audio_preset *) ptr;
283 if (new_len + preset_size > *len)
286 preset->len = sizeof(a2dp_sbc_t);
287 memcpy(preset->data, &sbc_presets[i], preset->len);
289 new_len += preset_size;
298 static int sbc_freq2int(uint8_t freq)
301 case SBC_SAMPLING_FREQ_16000:
303 case SBC_SAMPLING_FREQ_32000:
305 case SBC_SAMPLING_FREQ_44100:
307 case SBC_SAMPLING_FREQ_48000:
314 static const char *sbc_mode2str(uint8_t mode)
317 case SBC_CHANNEL_MODE_MONO:
319 case SBC_CHANNEL_MODE_DUAL_CHANNEL:
320 return "DualChannel";
321 case SBC_CHANNEL_MODE_STEREO:
323 case SBC_CHANNEL_MODE_JOINT_STEREO:
324 return "JointStereo";
330 static int sbc_blocks2int(uint8_t blocks)
333 case SBC_BLOCK_LENGTH_4:
335 case SBC_BLOCK_LENGTH_8:
337 case SBC_BLOCK_LENGTH_12:
339 case SBC_BLOCK_LENGTH_16:
346 static int sbc_subbands2int(uint8_t subbands)
358 static const char *sbc_allocation2str(uint8_t allocation)
360 switch (allocation) {
361 case SBC_ALLOCATION_SNR:
363 case SBC_ALLOCATION_LOUDNESS:
370 static void sbc_init_encoder(struct sbc_data *sbc_data)
372 a2dp_sbc_t *in = &sbc_data->sbc;
373 sbc_t *out = &sbc_data->enc;
375 sbc_init_a2dp(out, 0L, in, sizeof(*in));
377 out->endian = SBC_LE;
378 out->bitpool = in->max_bitpool;
380 DBG("frequency=%d channel_mode=%s block_length=%d subbands=%d "
381 "allocation=%s bitpool=%d-%d",
382 sbc_freq2int(in->frequency),
383 sbc_mode2str(in->channel_mode),
384 sbc_blocks2int(in->block_length),
385 sbc_subbands2int(in->subbands),
386 sbc_allocation2str(in->allocation_method),
387 in->min_bitpool, in->max_bitpool);
390 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
393 struct sbc_data *sbc_data;
394 size_t hdr_len = sizeof(struct media_packet);
396 size_t out_frame_len;
399 if (preset->len != sizeof(a2dp_sbc_t)) {
400 error("SBC: preset size mismatch");
401 return AUDIO_STATUS_FAILED;
404 sbc_data = calloc(sizeof(struct sbc_data), 1);
406 return AUDIO_STATUS_FAILED;
408 memcpy(&sbc_data->sbc, preset->data, preset->len);
410 sbc_init_encoder(sbc_data);
412 in_frame_len = sbc_get_codesize(&sbc_data->enc);
413 out_frame_len = sbc_get_frame_length(&sbc_data->enc);
414 num_frames = (mtu - hdr_len) / out_frame_len;
416 sbc_data->in_frame_len = in_frame_len;
417 sbc_data->in_buf_size = num_frames * in_frame_len;
419 sbc_data->out_frame_len = out_frame_len;
421 sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
422 sbc_data->frames_per_packet = num_frames;
424 DBG("mtu=%u in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
425 mtu, in_frame_len, out_frame_len, num_frames);
427 *codec_data = sbc_data;
429 return AUDIO_STATUS_SUCCESS;
432 static int sbc_cleanup(void *codec_data)
434 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
436 sbc_finish(&sbc_data->enc);
439 return AUDIO_STATUS_SUCCESS;
442 static int sbc_get_config(void *codec_data, struct audio_input_config *config)
444 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
446 switch (sbc_data->sbc.frequency) {
447 case SBC_SAMPLING_FREQ_16000:
448 config->rate = 16000;
450 case SBC_SAMPLING_FREQ_32000:
451 config->rate = 32000;
453 case SBC_SAMPLING_FREQ_44100:
454 config->rate = 44100;
456 case SBC_SAMPLING_FREQ_48000:
457 config->rate = 48000;
460 return AUDIO_STATUS_FAILED;
462 config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
463 AUDIO_CHANNEL_OUT_MONO :
464 AUDIO_CHANNEL_OUT_STEREO;
465 config->format = AUDIO_FORMAT_PCM_16_BIT;
467 return AUDIO_STATUS_SUCCESS;
470 static size_t sbc_get_buffer_size(void *codec_data)
472 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
474 return sbc_data->in_buf_size;
477 static size_t sbc_get_mediapacket_duration(void *codec_data)
479 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
481 return sbc_data->frame_duration * sbc_data->frames_per_packet;
484 static int write_media_packet(struct a2dp_stream_out *out, size_t mp_data_len,
485 uint32_t input_samples)
487 struct audio_endpoint *ep = out->ep;
488 struct media_packet *mp = ep->mp;
490 struct timespec diff;
491 uint32_t expected_samples;
495 ret = write(ep->fd, mp, sizeof(*mp) + mp_data_len);
503 clock_gettime(CLOCK_MONOTONIC, &cur);
504 timespec_diff(&cur, &ep->start, &diff);
505 expected_samples = (diff.tv_sec * 1000000ll + diff.tv_nsec / 1000ll) *
506 out->cfg.rate / 1000000ll;
508 /* AudioFlinger does not seem to provide any *working*
509 * API to provide data in some interval and will just
510 * send another buffer as soon as we process current
511 * one. To prevent overflowing L2CAP socket, we need to
512 * introduce some artificial delay here base on how many
513 * audio frames were sent so far, i.e. if we're not
514 * lagging behind audio stream, we can sleep for
515 * duration of single media packet.
517 if (ep->samples >= expected_samples)
518 usleep(input_samples * 1000000 / out->cfg.rate);
523 static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
524 size_t len, struct media_packet *mp,
525 size_t mp_data_len, size_t *written)
527 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
530 uint8_t frame_count = 0;
532 while (len - consumed >= sbc_data->in_frame_len &&
533 mp_data_len - encoded >= sbc_data->out_frame_len &&
534 frame_count < MAX_FRAMES_IN_PAYLOAD) {
538 read = sbc_encode(&sbc_data->enc, buffer + consumed,
539 sbc_data->in_frame_len, mp->data + encoded,
540 mp_data_len - encoded, &written);
543 error("SBC: failed to encode block at frame %d (%zd)",
554 mp->payload.frame_count = frame_count;
559 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
560 void *param, size_t *rsp_len, void *rsp, int *fd)
566 char cmsgbuf[CMSG_SPACE(sizeof(int))];
568 size_t s_len = sizeof(s);
570 pthread_mutex_lock(&sk_mutex);
573 error("audio: Invalid cmd socket passed to audio_ipc_cmd");
577 if (!rsp || !rsp_len) {
578 memset(&s, 0, s_len);
583 memset(&msg, 0, sizeof(msg));
584 memset(&cmd, 0, sizeof(cmd));
586 cmd.service_id = service_id;
590 iv[0].iov_base = &cmd;
591 iv[0].iov_len = sizeof(cmd);
593 iv[1].iov_base = param;
599 ret = sendmsg(audio_sk, &msg, 0);
601 error("audio: Sending command failed:%s", strerror(errno));
605 /* socket was shutdown */
607 error("audio: Command socket closed");
611 memset(&msg, 0, sizeof(msg));
612 memset(&cmd, 0, sizeof(cmd));
614 iv[0].iov_base = &cmd;
615 iv[0].iov_len = sizeof(cmd);
617 iv[1].iov_base = rsp;
618 iv[1].iov_len = *rsp_len;
624 memset(cmsgbuf, 0, sizeof(cmsgbuf));
625 msg.msg_control = cmsgbuf;
626 msg.msg_controllen = sizeof(cmsgbuf);
629 ret = recvmsg(audio_sk, &msg, 0);
631 error("audio: Receiving command response failed:%s",
636 if (ret < (ssize_t) sizeof(cmd)) {
637 error("audio: Too small response received(%zd bytes)", ret);
641 if (cmd.service_id != service_id) {
642 error("audio: Invalid service id (%u vs %u)", cmd.service_id,
647 if (ret != (ssize_t) (sizeof(cmd) + cmd.len)) {
648 error("audio: Malformed response received(%zd bytes)", ret);
652 if (cmd.opcode != opcode && cmd.opcode != AUDIO_OP_STATUS) {
653 error("audio: Invalid opcode received (%u vs %u)",
658 if (cmd.opcode == AUDIO_OP_STATUS) {
659 struct ipc_status *s = rsp;
661 if (sizeof(*s) != cmd.len) {
662 error("audio: Invalid status length");
666 if (s->code == AUDIO_STATUS_SUCCESS) {
667 error("audio: Invalid success status response");
671 pthread_mutex_unlock(&sk_mutex);
676 pthread_mutex_unlock(&sk_mutex);
678 /* Receive auxiliary data in msg */
680 struct cmsghdr *cmsg;
684 for (cmsg = CMSG_FIRSTHDR(&msg); cmsg;
685 cmsg = CMSG_NXTHDR(&msg, cmsg)) {
686 if (cmsg->cmsg_level == SOL_SOCKET
687 && cmsg->cmsg_type == SCM_RIGHTS) {
688 memcpy(fd, CMSG_DATA(cmsg), sizeof(int));
700 return AUDIO_STATUS_SUCCESS;
703 /* Some serious issue happen on IPC - recover */
704 shutdown(audio_sk, SHUT_RDWR);
705 pthread_mutex_unlock(&sk_mutex);
707 return AUDIO_STATUS_FAILED;
710 static int ipc_open_cmd(const struct audio_codec *codec)
712 uint8_t buf[BLUEZ_AUDIO_MTU];
713 struct audio_cmd_open *cmd = (struct audio_cmd_open *) buf;
714 struct audio_rsp_open rsp;
715 size_t cmd_len = sizeof(buf) - sizeof(*cmd);
716 size_t rsp_len = sizeof(rsp);
721 memcpy(cmd->uuid, a2dp_src_uuid, sizeof(a2dp_src_uuid));
723 cmd->codec = codec->type;
724 cmd->presets = codec->get_presets(cmd->preset, &cmd_len);
726 cmd_len += sizeof(*cmd);
728 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN, cmd_len, cmd,
729 &rsp_len, &rsp, NULL);
731 if (result != AUDIO_STATUS_SUCCESS)
737 static int ipc_close_cmd(uint8_t endpoint_id)
739 struct audio_cmd_close cmd;
744 cmd.id = endpoint_id;
746 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE,
747 sizeof(cmd), &cmd, NULL, NULL, NULL);
752 static int ipc_open_stream_cmd(uint8_t endpoint_id, uint16_t *mtu, int *fd,
753 struct audio_preset **caps)
755 char buf[BLUEZ_AUDIO_MTU];
756 struct audio_cmd_open_stream cmd;
757 struct audio_rsp_open_stream *rsp =
758 (struct audio_rsp_open_stream *) &buf;
759 size_t rsp_len = sizeof(buf);
765 return AUDIO_STATUS_FAILED;
767 cmd.id = endpoint_id;
769 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN_STREAM,
770 sizeof(cmd), &cmd, &rsp_len, rsp, fd);
771 if (result == AUDIO_STATUS_SUCCESS) {
772 size_t buf_len = sizeof(struct audio_preset) +
775 *caps = malloc(buf_len);
776 memcpy(*caps, &rsp->preset, buf_len);
784 static int ipc_close_stream_cmd(uint8_t endpoint_id)
786 struct audio_cmd_close_stream cmd;
791 cmd.id = endpoint_id;
793 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE_STREAM,
794 sizeof(cmd), &cmd, NULL, NULL, NULL);
799 static int ipc_resume_stream_cmd(uint8_t endpoint_id)
801 struct audio_cmd_resume_stream cmd;
806 cmd.id = endpoint_id;
808 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_RESUME_STREAM,
809 sizeof(cmd), &cmd, NULL, NULL, NULL);
814 static int ipc_suspend_stream_cmd(uint8_t endpoint_id)
816 struct audio_cmd_suspend_stream cmd;
821 cmd.id = endpoint_id;
823 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_SUSPEND_STREAM,
824 sizeof(cmd), &cmd, NULL, NULL, NULL);
829 static int register_endpoints(void)
831 struct audio_endpoint *ep = &audio_endpoints[0];
834 for (i = 0; i < NUM_CODECS; i++, ep++) {
835 const struct audio_codec *codec = &audio_codecs[i];
837 ep->id = ipc_open_cmd(codec);
840 return AUDIO_STATUS_FAILED;
843 ep->codec_data = NULL;
847 return AUDIO_STATUS_SUCCESS;
850 static void unregister_endpoints(void)
854 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++) {
855 struct audio_endpoint *ep = &audio_endpoints[i];
858 ipc_close_cmd(ep->id);
859 memset(ep, 0, sizeof(*ep));
864 static int set_blocking(int fd)
868 flags = fcntl(fd, F_GETFL, 0);
871 error("fcntl(F_GETFL): %s (%d)", strerror(-err), -err);
875 if (fcntl(fd, F_SETFL, flags & ~O_NONBLOCK) < 0) {
877 error("fcntl(F_SETFL): %s (%d)", strerror(-err), -err);
884 static bool open_endpoint(struct audio_endpoint *ep,
885 struct audio_input_config *cfg)
887 struct audio_preset *preset;
888 const struct audio_codec *codec;
892 if (ipc_open_stream_cmd(ep->id, &mtu, &fd, &preset) !=
893 AUDIO_STATUS_SUCCESS)
896 if (set_blocking(fd) < 0)
902 codec->init(preset, mtu, &ep->codec_data);
903 codec->get_config(ep->codec_data, cfg);
905 ep->mp = calloc(mtu, 1);
910 ep->mp->hdr.ssrc = htonl(1);
912 ep->mp_data_len = mtu - sizeof(*ep->mp);
925 static void close_endpoint(struct audio_endpoint *ep)
927 ipc_close_stream_cmd(ep->id);
935 ep->codec->cleanup(ep->codec_data);
936 ep->codec_data = NULL;
939 static void downmix_to_mono(struct a2dp_stream_out *out, const uint8_t *buffer,
942 const int16_t *input = (const void *) buffer;
943 int16_t *output = (void *) out->downmix_buf;
946 for (i = 0; i < bytes / 2; i++) {
947 int16_t l = le16_to_cpu(get_unaligned(&input[i * 2]));
948 int16_t r = le16_to_cpu(get_unaligned(&input[i * 2 + 1]));
950 put_unaligned(cpu_to_le16((l + r) / 2), &output[i]);
954 static bool write_data(struct a2dp_stream_out *out, const void *buffer,
957 struct audio_endpoint *ep = out->ep;
958 struct media_packet *mp = (struct media_packet *) ep->mp;
959 size_t free_space = ep->mp_data_len;
962 while (consumed < bytes) {
968 read = ep->codec->encode_mediapacket(ep->codec_data,
970 bytes - consumed, mp,
971 free_space, &written);
973 /* This is non-fatal and we can just assume buffer was processed
974 * properly and wait for next one.
981 mp->hdr.sequence_number = htons(ep->seq++);
982 mp->hdr.timestamp = htonl(ep->samples);
984 /* AudioFlinger provides 16bit PCM, so sample size is 2 bytes
985 * multipled by number of channels. Number of channels is simply
986 * number of bits set in channels mask.
988 samples = read / (2 * popcount(out->cfg.channels));
989 ep->samples += samples;
991 ret = write_media_packet(out, written, samples);
999 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
1002 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1003 const void *in_buf = buffer;
1004 size_t in_len = bytes;
1006 /* just return in case we're closing */
1007 if (out->audio_state == AUDIO_A2DP_STATE_NONE)
1010 /* We can auto-start only from standby */
1011 if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
1012 DBG("stream in standby, auto-start");
1014 if (ipc_resume_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1017 clock_gettime(CLOCK_MONOTONIC, &out->ep->start);
1018 out->ep->samples = 0;
1020 out->audio_state = AUDIO_A2DP_STATE_STARTED;
1023 if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
1024 error("audio: stream not started");
1028 if (out->ep->fd < 0) {
1029 error("audio: no transport socket");
1033 /* currently Android audioflinger is not able to provide mono stream on
1034 * A2DP output so down mixing needs to be done in hal-audio plugin.
1037 * AudioFlinger::PlaybackThread::readOutputParameters()
1038 * frameworks/av/services/audioflinger/Threads.cpp:1631
1040 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1041 if (!out->downmix_buf) {
1042 error("audio: downmix buffer not initialized");
1046 downmix_to_mono(out, buffer, bytes);
1048 in_buf = out->downmix_buf;
1052 if (!write_data(out, in_buf, in_len))
1058 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
1060 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1064 return out->cfg.rate;
1067 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1069 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1073 if (rate != out->cfg.rate) {
1074 warn("audio: cannot set sample rate to %d", rate);
1081 static size_t out_get_buffer_size(const struct audio_stream *stream)
1085 /* We should return proper buffer size calculated by codec (so each
1086 * input buffer is encoded into single media packed) but this does not
1087 * work well with AudioFlinger and causes problems. For this reason we
1088 * use magic value here and out_write code takes care of splitting
1089 * input buffer into multiple media packets.
1091 return FIXED_BUFFER_SIZE;
1094 static uint32_t out_get_channels(const struct audio_stream *stream)
1098 /* AudioFlinger can only provide stereo stream, so we return it here and
1099 * later we'll downmix this to mono in case codec requires it
1102 return AUDIO_CHANNEL_OUT_STEREO;
1105 static audio_format_t out_get_format(const struct audio_stream *stream)
1107 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1111 return out->cfg.format;
1114 static int out_set_format(struct audio_stream *stream, audio_format_t format)
1120 static int out_standby(struct audio_stream *stream)
1122 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1126 if (out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1127 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1129 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1135 static int out_dump(const struct audio_stream *stream, int fd)
1141 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
1143 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1147 bool enter_suspend = false;
1148 bool exit_suspend = false;
1152 str = strdup(kvpairs);
1153 kvpair = strtok_r(str, ";", &saveptr);
1155 for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
1158 keyval = strchr(kvpair, '=');
1165 if (!strcmp(kvpair, "closing")) {
1166 if (!strcmp(keyval, "true"))
1167 out->audio_state = AUDIO_A2DP_STATE_NONE;
1168 } else if (!strcmp(kvpair, "A2dpSuspended")) {
1169 if (!strcmp(keyval, "true"))
1170 enter_suspend = true;
1172 exit_suspend = true;
1178 if (enter_suspend && out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1179 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1181 out->audio_state = AUDIO_A2DP_STATE_SUSPENDED;
1184 if (exit_suspend && out->audio_state == AUDIO_A2DP_STATE_SUSPENDED)
1185 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1190 static char *out_get_parameters(const struct audio_stream *stream,
1197 static uint32_t out_get_latency(const struct audio_stream_out *stream)
1199 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1200 struct audio_endpoint *ep = out->ep;
1201 size_t pkt_duration;
1205 pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
1207 return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
1210 static int out_set_volume(struct audio_stream_out *stream, float left,
1214 /* volume controlled in audioflinger mixer (digital) */
1218 static int out_get_render_position(const struct audio_stream_out *stream,
1219 uint32_t *dsp_frames)
1225 static int out_add_audio_effect(const struct audio_stream *stream,
1226 effect_handle_t effect)
1232 static int out_remove_audio_effect(const struct audio_stream *stream,
1233 effect_handle_t effect)
1239 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1245 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1251 static size_t in_get_buffer_size(const struct audio_stream *stream)
1257 static uint32_t in_get_channels(const struct audio_stream *stream)
1263 static audio_format_t in_get_format(const struct audio_stream *stream)
1269 static int in_set_format(struct audio_stream *stream, audio_format_t format)
1275 static int in_standby(struct audio_stream *stream)
1281 static int in_dump(const struct audio_stream *stream, int fd)
1287 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1293 static char *in_get_parameters(const struct audio_stream *stream,
1300 static int in_set_gain(struct audio_stream_in *stream, float gain)
1306 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1313 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1319 static int in_add_audio_effect(const struct audio_stream *stream,
1320 effect_handle_t effect)
1326 static int in_remove_audio_effect(const struct audio_stream *stream,
1327 effect_handle_t effect)
1333 static int audio_open_output_stream(struct audio_hw_device *dev,
1334 audio_io_handle_t handle,
1335 audio_devices_t devices,
1336 audio_output_flags_t flags,
1337 struct audio_config *config,
1338 struct audio_stream_out **stream_out)
1341 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1342 struct a2dp_stream_out *out;
1344 out = calloc(1, sizeof(struct a2dp_stream_out));
1350 out->stream.common.get_sample_rate = out_get_sample_rate;
1351 out->stream.common.set_sample_rate = out_set_sample_rate;
1352 out->stream.common.get_buffer_size = out_get_buffer_size;
1353 out->stream.common.get_channels = out_get_channels;
1354 out->stream.common.get_format = out_get_format;
1355 out->stream.common.set_format = out_set_format;
1356 out->stream.common.standby = out_standby;
1357 out->stream.common.dump = out_dump;
1358 out->stream.common.set_parameters = out_set_parameters;
1359 out->stream.common.get_parameters = out_get_parameters;
1360 out->stream.common.add_audio_effect = out_add_audio_effect;
1361 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1362 out->stream.get_latency = out_get_latency;
1363 out->stream.set_volume = out_set_volume;
1364 out->stream.write = out_write;
1365 out->stream.get_render_position = out_get_render_position;
1367 /* TODO: for now we always use endpoint 0 */
1368 out->ep = &audio_endpoints[0];
1370 if (!open_endpoint(out->ep, &out->cfg))
1373 DBG("rate=%d channels=%d format=%d", out->cfg.rate,
1374 out->cfg.channels, out->cfg.format);
1376 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1377 out->downmix_buf = malloc(FIXED_BUFFER_SIZE / 2);
1378 if (!out->downmix_buf)
1382 *stream_out = &out->stream;
1383 a2dp_dev->out = out;
1385 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1390 error("audio: cannot open output stream");
1396 static void audio_close_output_stream(struct audio_hw_device *dev,
1397 struct audio_stream_out *stream)
1399 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1400 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1404 close_endpoint(a2dp_dev->out->ep);
1406 free(out->downmix_buf);
1409 a2dp_dev->out = NULL;
1412 static int audio_set_parameters(struct audio_hw_device *dev,
1413 const char *kvpairs)
1415 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1416 struct a2dp_stream_out *out = a2dp_dev->out;
1423 return out->stream.common.set_parameters((struct audio_stream *) out,
1427 static char *audio_get_parameters(const struct audio_hw_device *dev,
1434 static int audio_init_check(const struct audio_hw_device *dev)
1440 static int audio_set_voice_volume(struct audio_hw_device *dev, float volume)
1446 static int audio_set_master_volume(struct audio_hw_device *dev, float volume)
1452 static int audio_set_mode(struct audio_hw_device *dev, int mode)
1458 static int audio_set_mic_mute(struct audio_hw_device *dev, bool state)
1464 static int audio_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1470 static size_t audio_get_input_buffer_size(const struct audio_hw_device *dev,
1471 const struct audio_config *config)
1477 static int audio_open_input_stream(struct audio_hw_device *dev,
1478 audio_io_handle_t handle,
1479 audio_devices_t devices,
1480 struct audio_config *config,
1481 struct audio_stream_in **stream_in)
1483 struct audio_stream_in *in;
1487 in = calloc(1, sizeof(struct audio_stream_in));
1491 in->common.get_sample_rate = in_get_sample_rate;
1492 in->common.set_sample_rate = in_set_sample_rate;
1493 in->common.get_buffer_size = in_get_buffer_size;
1494 in->common.get_channels = in_get_channels;
1495 in->common.get_format = in_get_format;
1496 in->common.set_format = in_set_format;
1497 in->common.standby = in_standby;
1498 in->common.dump = in_dump;
1499 in->common.set_parameters = in_set_parameters;
1500 in->common.get_parameters = in_get_parameters;
1501 in->common.add_audio_effect = in_add_audio_effect;
1502 in->common.remove_audio_effect = in_remove_audio_effect;
1503 in->set_gain = in_set_gain;
1505 in->get_input_frames_lost = in_get_input_frames_lost;
1512 static void audio_close_input_stream(struct audio_hw_device *dev,
1513 struct audio_stream_in *stream_in)
1519 static int audio_dump(const audio_hw_device_t *device, int fd)
1525 static int audio_close(hw_device_t *device)
1527 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
1531 unregister_endpoints();
1533 shutdown(listen_sk, SHUT_RDWR);
1534 shutdown(audio_sk, SHUT_RDWR);
1536 pthread_join(ipc_th, NULL);
1545 static void *ipc_handler(void *data)
1554 DBG("Waiting for connection ...");
1556 sk = accept(listen_sk, NULL, NULL);
1563 if (err != ECONNABORTED && err != EINVAL)
1564 error("audio: Failed to accept socket: %d (%s)",
1565 err, strerror(err));
1570 pthread_mutex_lock(&sk_mutex);
1572 pthread_mutex_unlock(&sk_mutex);
1574 DBG("Audio IPC: Connected");
1576 if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
1577 error("audio: Failed to register endpoints");
1579 unregister_endpoints();
1581 pthread_mutex_lock(&sk_mutex);
1582 shutdown(audio_sk, SHUT_RDWR);
1585 pthread_mutex_unlock(&sk_mutex);
1590 memset(&pfd, 0, sizeof(pfd));
1592 pfd.events = POLLHUP | POLLERR | POLLNVAL;
1594 /* Check if socket is still alive. Empty while loop.*/
1595 while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
1597 if (pfd.revents & (POLLHUP | POLLERR | POLLNVAL)) {
1598 info("Audio HAL: Socket closed");
1600 pthread_mutex_lock(&sk_mutex);
1603 pthread_mutex_unlock(&sk_mutex);
1607 /* audio_sk is closed at this point, just cleanup endpoints states */
1608 memset(audio_endpoints, 0, sizeof(audio_endpoints));
1610 info("Closing Audio IPC thread");
1614 static int audio_ipc_init(void)
1616 struct sockaddr_un addr;
1622 sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
1625 error("audio: Failed to create socket: %d (%s)", -err,
1630 memset(&addr, 0, sizeof(addr));
1631 addr.sun_family = AF_UNIX;
1633 memcpy(addr.sun_path, BLUEZ_AUDIO_SK_PATH,
1634 sizeof(BLUEZ_AUDIO_SK_PATH));
1636 if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
1638 error("audio: Failed to bind socket: %d (%s)", -err,
1643 if (listen(sk, 1) < 0) {
1645 error("audio: Failed to listen on the socket: %d (%s)", -err,
1652 err = pthread_create(&ipc_th, NULL, ipc_handler, NULL);
1656 error("audio: Failed to start Audio IPC thread: %d (%s)",
1657 -err, strerror(-err));
1668 static int audio_open(const hw_module_t *module, const char *name,
1669 hw_device_t **device)
1671 struct a2dp_audio_dev *a2dp_dev;
1676 if (strcmp(name, AUDIO_HARDWARE_INTERFACE)) {
1677 error("audio: interface %s not matching [%s]", name,
1678 AUDIO_HARDWARE_INTERFACE);
1682 err = audio_ipc_init();
1686 a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
1690 a2dp_dev->dev.common.tag = HARDWARE_DEVICE_TAG;
1691 a2dp_dev->dev.common.version = AUDIO_DEVICE_API_VERSION_CURRENT;
1692 a2dp_dev->dev.common.module = (struct hw_module_t *) module;
1693 a2dp_dev->dev.common.close = audio_close;
1695 a2dp_dev->dev.init_check = audio_init_check;
1696 a2dp_dev->dev.set_voice_volume = audio_set_voice_volume;
1697 a2dp_dev->dev.set_master_volume = audio_set_master_volume;
1698 a2dp_dev->dev.set_mode = audio_set_mode;
1699 a2dp_dev->dev.set_mic_mute = audio_set_mic_mute;
1700 a2dp_dev->dev.get_mic_mute = audio_get_mic_mute;
1701 a2dp_dev->dev.set_parameters = audio_set_parameters;
1702 a2dp_dev->dev.get_parameters = audio_get_parameters;
1703 a2dp_dev->dev.get_input_buffer_size = audio_get_input_buffer_size;
1704 a2dp_dev->dev.open_output_stream = audio_open_output_stream;
1705 a2dp_dev->dev.close_output_stream = audio_close_output_stream;
1706 a2dp_dev->dev.open_input_stream = audio_open_input_stream;
1707 a2dp_dev->dev.close_input_stream = audio_close_input_stream;
1708 a2dp_dev->dev.dump = audio_dump;
1710 /* Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
1711 * This results from the structure of following structs:a2dp_audio_dev,
1712 * audio_hw_device. We will rely on this later in the code.*/
1713 *device = &a2dp_dev->dev.common;
1718 static struct hw_module_methods_t hal_module_methods = {
1722 struct audio_module HAL_MODULE_INFO_SYM = {
1724 .tag = HARDWARE_MODULE_TAG,
1727 .id = AUDIO_HARDWARE_MODULE_ID,
1728 .name = "A2DP Bluez HW HAL",
1729 .author = "Intel Corporation",
1730 .methods = &hal_module_methods,