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 void timespec_add(struct timespec *base, uint64_t time_us,
151 struct timespec *res)
153 res->tv_sec = base->tv_sec + time_us / 1000000;
154 res->tv_nsec = base->tv_nsec + (time_us % 1000000) * 1000;
156 if (res->tv_nsec >= 1000000000) {
158 res->tv_nsec -= 1000000000;
163 /* Bionic does not have clock_nanosleep() prototype in time.h even though
164 * it provides its implementation.
166 extern int clock_nanosleep(clockid_t clock_id, int flags,
167 const struct timespec *request,
168 struct timespec *remain);
171 static int sbc_get_presets(struct audio_preset *preset, size_t *len);
172 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
174 static int sbc_cleanup(void *codec_data);
175 static int sbc_get_config(void *codec_data, struct audio_input_config *config);
176 static size_t sbc_get_buffer_size(void *codec_data);
177 static size_t sbc_get_mediapacket_duration(void *codec_data);
178 static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
179 size_t len, struct media_packet *mp,
180 size_t mp_data_len, size_t *written);
185 int (*get_presets) (struct audio_preset *preset, size_t *len);
187 int (*init) (struct audio_preset *preset, uint16_t mtu,
189 int (*cleanup) (void *codec_data);
190 int (*get_config) (void *codec_data,
191 struct audio_input_config *config);
192 size_t (*get_buffer_size) (void *codec_data);
193 size_t (*get_mediapacket_duration) (void *codec_data);
194 ssize_t (*encode_mediapacket) (void *codec_data, const uint8_t *buffer,
195 size_t len, struct media_packet *mp,
196 size_t mp_data_len, size_t *written);
199 static const struct audio_codec audio_codecs[] = {
201 .type = A2DP_CODEC_SBC,
203 .get_presets = sbc_get_presets,
205 .init = sbc_codec_init,
206 .cleanup = sbc_cleanup,
207 .get_config = sbc_get_config,
208 .get_buffer_size = sbc_get_buffer_size,
209 .get_mediapacket_duration = sbc_get_mediapacket_duration,
210 .encode_mediapacket = sbc_encode_mediapacket,
214 #define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
216 #define MAX_AUDIO_ENDPOINTS NUM_CODECS
218 struct audio_endpoint {
220 const struct audio_codec *codec;
224 struct media_packet *mp;
229 struct timespec start;
232 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
235 AUDIO_A2DP_STATE_NONE,
236 AUDIO_A2DP_STATE_STANDBY,
237 AUDIO_A2DP_STATE_SUSPENDED,
238 AUDIO_A2DP_STATE_STARTED
241 struct a2dp_stream_out {
242 struct audio_stream_out stream;
244 struct audio_endpoint *ep;
245 enum a2dp_state_t audio_state;
246 struct audio_input_config cfg;
248 uint8_t *downmix_buf;
251 struct a2dp_audio_dev {
252 struct audio_hw_device dev;
253 struct a2dp_stream_out *out;
256 static const a2dp_sbc_t sbc_presets[] = {
258 .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
259 .channel_mode = SBC_CHANNEL_MODE_MONO |
260 SBC_CHANNEL_MODE_DUAL_CHANNEL |
261 SBC_CHANNEL_MODE_STEREO |
262 SBC_CHANNEL_MODE_JOINT_STEREO,
263 .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
264 .allocation_method = SBC_ALLOCATION_SNR |
265 SBC_ALLOCATION_LOUDNESS,
266 .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
267 SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
268 .min_bitpool = MIN_BITPOOL,
269 .max_bitpool = MAX_BITPOOL
272 .frequency = SBC_SAMPLING_FREQ_44100,
273 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
274 .subbands = SBC_SUBBANDS_8,
275 .allocation_method = SBC_ALLOCATION_LOUDNESS,
276 .block_length = SBC_BLOCK_LENGTH_16,
277 .min_bitpool = MIN_BITPOOL,
278 .max_bitpool = MAX_BITPOOL
281 .frequency = SBC_SAMPLING_FREQ_48000,
282 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
283 .subbands = SBC_SUBBANDS_8,
284 .allocation_method = SBC_ALLOCATION_LOUDNESS,
285 .block_length = SBC_BLOCK_LENGTH_16,
286 .min_bitpool = MIN_BITPOOL,
287 .max_bitpool = MAX_BITPOOL
291 static int sbc_get_presets(struct audio_preset *preset, size_t *len)
296 uint8_t *ptr = (uint8_t *) preset;
297 size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
299 count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
301 for (i = 0; i < count; i++) {
302 preset = (struct audio_preset *) ptr;
304 if (new_len + preset_size > *len)
307 preset->len = sizeof(a2dp_sbc_t);
308 memcpy(preset->data, &sbc_presets[i], preset->len);
310 new_len += preset_size;
319 static int sbc_freq2int(uint8_t freq)
322 case SBC_SAMPLING_FREQ_16000:
324 case SBC_SAMPLING_FREQ_32000:
326 case SBC_SAMPLING_FREQ_44100:
328 case SBC_SAMPLING_FREQ_48000:
335 static const char *sbc_mode2str(uint8_t mode)
338 case SBC_CHANNEL_MODE_MONO:
340 case SBC_CHANNEL_MODE_DUAL_CHANNEL:
341 return "DualChannel";
342 case SBC_CHANNEL_MODE_STEREO:
344 case SBC_CHANNEL_MODE_JOINT_STEREO:
345 return "JointStereo";
351 static int sbc_blocks2int(uint8_t blocks)
354 case SBC_BLOCK_LENGTH_4:
356 case SBC_BLOCK_LENGTH_8:
358 case SBC_BLOCK_LENGTH_12:
360 case SBC_BLOCK_LENGTH_16:
367 static int sbc_subbands2int(uint8_t subbands)
379 static const char *sbc_allocation2str(uint8_t allocation)
381 switch (allocation) {
382 case SBC_ALLOCATION_SNR:
384 case SBC_ALLOCATION_LOUDNESS:
391 static void sbc_init_encoder(struct sbc_data *sbc_data)
393 a2dp_sbc_t *in = &sbc_data->sbc;
394 sbc_t *out = &sbc_data->enc;
396 sbc_init_a2dp(out, 0L, in, sizeof(*in));
398 out->endian = SBC_LE;
399 out->bitpool = in->max_bitpool;
401 DBG("frequency=%d channel_mode=%s block_length=%d subbands=%d "
402 "allocation=%s bitpool=%d-%d",
403 sbc_freq2int(in->frequency),
404 sbc_mode2str(in->channel_mode),
405 sbc_blocks2int(in->block_length),
406 sbc_subbands2int(in->subbands),
407 sbc_allocation2str(in->allocation_method),
408 in->min_bitpool, in->max_bitpool);
411 static int sbc_codec_init(struct audio_preset *preset, uint16_t payload_len,
414 struct sbc_data *sbc_data;
416 size_t out_frame_len;
419 if (preset->len != sizeof(a2dp_sbc_t)) {
420 error("SBC: preset size mismatch");
421 return AUDIO_STATUS_FAILED;
424 sbc_data = calloc(sizeof(struct sbc_data), 1);
426 return AUDIO_STATUS_FAILED;
428 memcpy(&sbc_data->sbc, preset->data, preset->len);
430 sbc_init_encoder(sbc_data);
432 in_frame_len = sbc_get_codesize(&sbc_data->enc);
433 out_frame_len = sbc_get_frame_length(&sbc_data->enc);
434 num_frames = payload_len / out_frame_len;
436 sbc_data->in_frame_len = in_frame_len;
437 sbc_data->in_buf_size = num_frames * in_frame_len;
439 sbc_data->out_frame_len = out_frame_len;
441 sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
442 sbc_data->frames_per_packet = num_frames;
444 DBG("in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
445 in_frame_len, out_frame_len, num_frames);
447 *codec_data = sbc_data;
449 return AUDIO_STATUS_SUCCESS;
452 static int sbc_cleanup(void *codec_data)
454 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
456 sbc_finish(&sbc_data->enc);
459 return AUDIO_STATUS_SUCCESS;
462 static int sbc_get_config(void *codec_data, struct audio_input_config *config)
464 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
466 switch (sbc_data->sbc.frequency) {
467 case SBC_SAMPLING_FREQ_16000:
468 config->rate = 16000;
470 case SBC_SAMPLING_FREQ_32000:
471 config->rate = 32000;
473 case SBC_SAMPLING_FREQ_44100:
474 config->rate = 44100;
476 case SBC_SAMPLING_FREQ_48000:
477 config->rate = 48000;
480 return AUDIO_STATUS_FAILED;
482 config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
483 AUDIO_CHANNEL_OUT_MONO :
484 AUDIO_CHANNEL_OUT_STEREO;
485 config->format = AUDIO_FORMAT_PCM_16_BIT;
487 return AUDIO_STATUS_SUCCESS;
490 static size_t sbc_get_buffer_size(void *codec_data)
492 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
494 return sbc_data->in_buf_size;
497 static size_t sbc_get_mediapacket_duration(void *codec_data)
499 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
501 return sbc_data->frame_duration * sbc_data->frames_per_packet;
504 static ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
505 size_t len, struct media_packet *mp,
506 size_t mp_data_len, size_t *written)
508 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
511 uint8_t frame_count = 0;
513 while (len - consumed >= sbc_data->in_frame_len &&
514 mp_data_len - encoded >= sbc_data->out_frame_len &&
515 frame_count < MAX_FRAMES_IN_PAYLOAD) {
519 read = sbc_encode(&sbc_data->enc, buffer + consumed,
520 sbc_data->in_frame_len, mp->data + encoded,
521 mp_data_len - encoded, &written);
524 error("SBC: failed to encode block at frame %d (%zd)",
535 mp->payload.frame_count = frame_count;
540 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
541 void *param, size_t *rsp_len, void *rsp, int *fd)
547 char cmsgbuf[CMSG_SPACE(sizeof(int))];
549 size_t s_len = sizeof(s);
551 pthread_mutex_lock(&sk_mutex);
554 error("audio: Invalid cmd socket passed to audio_ipc_cmd");
558 if (!rsp || !rsp_len) {
559 memset(&s, 0, s_len);
564 memset(&msg, 0, sizeof(msg));
565 memset(&cmd, 0, sizeof(cmd));
567 cmd.service_id = service_id;
571 iv[0].iov_base = &cmd;
572 iv[0].iov_len = sizeof(cmd);
574 iv[1].iov_base = param;
580 ret = sendmsg(audio_sk, &msg, 0);
582 error("audio: Sending command failed:%s", strerror(errno));
586 /* socket was shutdown */
588 error("audio: Command socket closed");
592 memset(&msg, 0, sizeof(msg));
593 memset(&cmd, 0, sizeof(cmd));
595 iv[0].iov_base = &cmd;
596 iv[0].iov_len = sizeof(cmd);
598 iv[1].iov_base = rsp;
599 iv[1].iov_len = *rsp_len;
605 memset(cmsgbuf, 0, sizeof(cmsgbuf));
606 msg.msg_control = cmsgbuf;
607 msg.msg_controllen = sizeof(cmsgbuf);
610 ret = recvmsg(audio_sk, &msg, 0);
612 error("audio: Receiving command response failed:%s",
617 if (ret < (ssize_t) sizeof(cmd)) {
618 error("audio: Too small response received(%zd bytes)", ret);
622 if (cmd.service_id != service_id) {
623 error("audio: Invalid service id (%u vs %u)", cmd.service_id,
628 if (ret != (ssize_t) (sizeof(cmd) + cmd.len)) {
629 error("audio: Malformed response received(%zd bytes)", ret);
633 if (cmd.opcode != opcode && cmd.opcode != AUDIO_OP_STATUS) {
634 error("audio: Invalid opcode received (%u vs %u)",
639 if (cmd.opcode == AUDIO_OP_STATUS) {
640 struct ipc_status *s = rsp;
642 if (sizeof(*s) != cmd.len) {
643 error("audio: Invalid status length");
647 if (s->code == AUDIO_STATUS_SUCCESS) {
648 error("audio: Invalid success status response");
652 pthread_mutex_unlock(&sk_mutex);
657 pthread_mutex_unlock(&sk_mutex);
659 /* Receive auxiliary data in msg */
661 struct cmsghdr *cmsg;
665 for (cmsg = CMSG_FIRSTHDR(&msg); cmsg;
666 cmsg = CMSG_NXTHDR(&msg, cmsg)) {
667 if (cmsg->cmsg_level == SOL_SOCKET
668 && cmsg->cmsg_type == SCM_RIGHTS) {
669 memcpy(fd, CMSG_DATA(cmsg), sizeof(int));
681 return AUDIO_STATUS_SUCCESS;
684 /* Some serious issue happen on IPC - recover */
685 shutdown(audio_sk, SHUT_RDWR);
686 pthread_mutex_unlock(&sk_mutex);
688 return AUDIO_STATUS_FAILED;
691 static int ipc_open_cmd(const struct audio_codec *codec)
693 uint8_t buf[BLUEZ_AUDIO_MTU];
694 struct audio_cmd_open *cmd = (struct audio_cmd_open *) buf;
695 struct audio_rsp_open rsp;
696 size_t cmd_len = sizeof(buf) - sizeof(*cmd);
697 size_t rsp_len = sizeof(rsp);
702 memcpy(cmd->uuid, a2dp_src_uuid, sizeof(a2dp_src_uuid));
704 cmd->codec = codec->type;
705 cmd->presets = codec->get_presets(cmd->preset, &cmd_len);
707 cmd_len += sizeof(*cmd);
709 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN, cmd_len, cmd,
710 &rsp_len, &rsp, NULL);
712 if (result != AUDIO_STATUS_SUCCESS)
718 static int ipc_close_cmd(uint8_t endpoint_id)
720 struct audio_cmd_close cmd;
725 cmd.id = endpoint_id;
727 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE,
728 sizeof(cmd), &cmd, NULL, NULL, NULL);
733 static int ipc_open_stream_cmd(uint8_t endpoint_id, uint16_t *mtu, int *fd,
734 struct audio_preset **caps)
736 char buf[BLUEZ_AUDIO_MTU];
737 struct audio_cmd_open_stream cmd;
738 struct audio_rsp_open_stream *rsp =
739 (struct audio_rsp_open_stream *) &buf;
740 size_t rsp_len = sizeof(buf);
746 return AUDIO_STATUS_FAILED;
748 cmd.id = endpoint_id;
750 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN_STREAM,
751 sizeof(cmd), &cmd, &rsp_len, rsp, fd);
752 if (result == AUDIO_STATUS_SUCCESS) {
753 size_t buf_len = sizeof(struct audio_preset) +
756 *caps = malloc(buf_len);
757 memcpy(*caps, &rsp->preset, buf_len);
765 static int ipc_close_stream_cmd(uint8_t endpoint_id)
767 struct audio_cmd_close_stream cmd;
772 cmd.id = endpoint_id;
774 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE_STREAM,
775 sizeof(cmd), &cmd, NULL, NULL, NULL);
780 static int ipc_resume_stream_cmd(uint8_t endpoint_id)
782 struct audio_cmd_resume_stream cmd;
787 cmd.id = endpoint_id;
789 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_RESUME_STREAM,
790 sizeof(cmd), &cmd, NULL, NULL, NULL);
795 static int ipc_suspend_stream_cmd(uint8_t endpoint_id)
797 struct audio_cmd_suspend_stream cmd;
802 cmd.id = endpoint_id;
804 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_SUSPEND_STREAM,
805 sizeof(cmd), &cmd, NULL, NULL, NULL);
810 static int register_endpoints(void)
812 struct audio_endpoint *ep = &audio_endpoints[0];
815 for (i = 0; i < NUM_CODECS; i++, ep++) {
816 const struct audio_codec *codec = &audio_codecs[i];
818 ep->id = ipc_open_cmd(codec);
821 return AUDIO_STATUS_FAILED;
824 ep->codec_data = NULL;
828 return AUDIO_STATUS_SUCCESS;
831 static void unregister_endpoints(void)
835 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++) {
836 struct audio_endpoint *ep = &audio_endpoints[i];
839 ipc_close_cmd(ep->id);
840 memset(ep, 0, sizeof(*ep));
845 static int set_blocking(int fd)
849 flags = fcntl(fd, F_GETFL, 0);
852 error("fcntl(F_GETFL): %s (%d)", strerror(-err), -err);
856 if (fcntl(fd, F_SETFL, flags & ~O_NONBLOCK) < 0) {
858 error("fcntl(F_SETFL): %s (%d)", strerror(-err), -err);
865 static bool open_endpoint(struct audio_endpoint *ep,
866 struct audio_input_config *cfg)
868 struct audio_preset *preset;
869 const struct audio_codec *codec;
871 uint16_t payload_len;
874 if (ipc_open_stream_cmd(ep->id, &mtu, &fd, &preset) !=
875 AUDIO_STATUS_SUCCESS)
878 if (set_blocking(fd) < 0)
883 payload_len = mtu - sizeof(*ep->mp);
888 codec->init(preset, payload_len, &ep->codec_data);
889 codec->get_config(ep->codec_data, cfg);
891 ep->mp = calloc(mtu, 1);
896 ep->mp->hdr.ssrc = htonl(1);
898 ep->mp_data_len = payload_len;
911 static void close_endpoint(struct audio_endpoint *ep)
913 ipc_close_stream_cmd(ep->id);
921 ep->codec->cleanup(ep->codec_data);
922 ep->codec_data = NULL;
925 static void downmix_to_mono(struct a2dp_stream_out *out, const uint8_t *buffer,
928 const int16_t *input = (const void *) buffer;
929 int16_t *output = (void *) out->downmix_buf;
932 for (i = 0; i < bytes / 2; i++) {
933 int16_t l = le16_to_cpu(get_unaligned(&input[i * 2]));
934 int16_t r = le16_to_cpu(get_unaligned(&input[i * 2 + 1]));
936 put_unaligned(cpu_to_le16((l + r) / 2), &output[i]);
940 static bool write_data(struct a2dp_stream_out *out, const void *buffer,
943 struct audio_endpoint *ep = out->ep;
944 struct media_packet *mp = (struct media_packet *) ep->mp;
945 size_t free_space = ep->mp_data_len;
948 while (consumed < bytes) {
954 struct timespec anchor;
956 time_us = ep->samples * 1000000ll / out->cfg.rate;
958 timespec_add(&ep->start, time_us, &anchor);
961 ret = clock_nanosleep(CLOCK_MONOTONIC, TIMER_ABSTIME,
968 error("clock_nanosleep failed (%d)", ret);
973 read = ep->codec->encode_mediapacket(ep->codec_data,
975 bytes - consumed, mp,
976 free_space, &written);
978 /* This is non-fatal and we can just assume buffer was processed
979 * properly and wait for next one.
986 mp->hdr.sequence_number = htons(ep->seq++);
987 mp->hdr.timestamp = htonl(ep->samples);
989 /* AudioFlinger provides 16bit PCM, so sample size is 2 bytes
990 * multipled by number of channels. Number of channels is simply
991 * number of bits set in channels mask.
993 samples = read / (2 * popcount(out->cfg.channels));
994 ep->samples += samples;
997 ret = write(ep->fd, mp, sizeof(*mp) + written);
1010 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
1013 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1014 const void *in_buf = buffer;
1015 size_t in_len = bytes;
1017 /* just return in case we're closing */
1018 if (out->audio_state == AUDIO_A2DP_STATE_NONE)
1021 /* We can auto-start only from standby */
1022 if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
1023 DBG("stream in standby, auto-start");
1025 if (ipc_resume_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1028 clock_gettime(CLOCK_MONOTONIC, &out->ep->start);
1029 out->ep->samples = 0;
1031 out->audio_state = AUDIO_A2DP_STATE_STARTED;
1034 if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
1035 error("audio: stream not started");
1039 if (out->ep->fd < 0) {
1040 error("audio: no transport socket");
1044 /* currently Android audioflinger is not able to provide mono stream on
1045 * A2DP output so down mixing needs to be done in hal-audio plugin.
1048 * AudioFlinger::PlaybackThread::readOutputParameters()
1049 * frameworks/av/services/audioflinger/Threads.cpp:1631
1051 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1052 if (!out->downmix_buf) {
1053 error("audio: downmix buffer not initialized");
1057 downmix_to_mono(out, buffer, bytes);
1059 in_buf = out->downmix_buf;
1063 if (!write_data(out, in_buf, in_len))
1069 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
1071 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1075 return out->cfg.rate;
1078 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1080 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1084 if (rate != out->cfg.rate) {
1085 warn("audio: cannot set sample rate to %d", rate);
1092 static size_t out_get_buffer_size(const struct audio_stream *stream)
1096 /* We should return proper buffer size calculated by codec (so each
1097 * input buffer is encoded into single media packed) but this does not
1098 * work well with AudioFlinger and causes problems. For this reason we
1099 * use magic value here and out_write code takes care of splitting
1100 * input buffer into multiple media packets.
1102 return FIXED_BUFFER_SIZE;
1105 static uint32_t out_get_channels(const struct audio_stream *stream)
1109 /* AudioFlinger can only provide stereo stream, so we return it here and
1110 * later we'll downmix this to mono in case codec requires it
1113 return AUDIO_CHANNEL_OUT_STEREO;
1116 static audio_format_t out_get_format(const struct audio_stream *stream)
1118 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1122 return out->cfg.format;
1125 static int out_set_format(struct audio_stream *stream, audio_format_t format)
1131 static int out_standby(struct audio_stream *stream)
1133 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1137 if (out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1138 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1140 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1146 static int out_dump(const struct audio_stream *stream, int fd)
1152 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
1154 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1158 bool enter_suspend = false;
1159 bool exit_suspend = false;
1163 str = strdup(kvpairs);
1167 kvpair = strtok_r(str, ";", &saveptr);
1169 for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
1172 keyval = strchr(kvpair, '=');
1179 if (!strcmp(kvpair, "closing")) {
1180 if (!strcmp(keyval, "true"))
1181 out->audio_state = AUDIO_A2DP_STATE_NONE;
1182 } else if (!strcmp(kvpair, "A2dpSuspended")) {
1183 if (!strcmp(keyval, "true"))
1184 enter_suspend = true;
1186 exit_suspend = true;
1192 if (enter_suspend && out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1193 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1195 out->audio_state = AUDIO_A2DP_STATE_SUSPENDED;
1198 if (exit_suspend && out->audio_state == AUDIO_A2DP_STATE_SUSPENDED)
1199 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1204 static char *out_get_parameters(const struct audio_stream *stream,
1211 static uint32_t out_get_latency(const struct audio_stream_out *stream)
1213 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1214 struct audio_endpoint *ep = out->ep;
1215 size_t pkt_duration;
1219 pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
1221 return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
1224 static int out_set_volume(struct audio_stream_out *stream, float left,
1228 /* volume controlled in audioflinger mixer (digital) */
1232 static int out_get_render_position(const struct audio_stream_out *stream,
1233 uint32_t *dsp_frames)
1239 static int out_add_audio_effect(const struct audio_stream *stream,
1240 effect_handle_t effect)
1246 static int out_remove_audio_effect(const struct audio_stream *stream,
1247 effect_handle_t effect)
1253 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1259 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1265 static size_t in_get_buffer_size(const struct audio_stream *stream)
1271 static uint32_t in_get_channels(const struct audio_stream *stream)
1277 static audio_format_t in_get_format(const struct audio_stream *stream)
1283 static int in_set_format(struct audio_stream *stream, audio_format_t format)
1289 static int in_standby(struct audio_stream *stream)
1295 static int in_dump(const struct audio_stream *stream, int fd)
1301 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1307 static char *in_get_parameters(const struct audio_stream *stream,
1314 static int in_set_gain(struct audio_stream_in *stream, float gain)
1320 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1327 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1333 static int in_add_audio_effect(const struct audio_stream *stream,
1334 effect_handle_t effect)
1340 static int in_remove_audio_effect(const struct audio_stream *stream,
1341 effect_handle_t effect)
1347 static int audio_open_output_stream(struct audio_hw_device *dev,
1348 audio_io_handle_t handle,
1349 audio_devices_t devices,
1350 audio_output_flags_t flags,
1351 struct audio_config *config,
1352 struct audio_stream_out **stream_out)
1355 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1356 struct a2dp_stream_out *out;
1358 out = calloc(1, sizeof(struct a2dp_stream_out));
1364 out->stream.common.get_sample_rate = out_get_sample_rate;
1365 out->stream.common.set_sample_rate = out_set_sample_rate;
1366 out->stream.common.get_buffer_size = out_get_buffer_size;
1367 out->stream.common.get_channels = out_get_channels;
1368 out->stream.common.get_format = out_get_format;
1369 out->stream.common.set_format = out_set_format;
1370 out->stream.common.standby = out_standby;
1371 out->stream.common.dump = out_dump;
1372 out->stream.common.set_parameters = out_set_parameters;
1373 out->stream.common.get_parameters = out_get_parameters;
1374 out->stream.common.add_audio_effect = out_add_audio_effect;
1375 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1376 out->stream.get_latency = out_get_latency;
1377 out->stream.set_volume = out_set_volume;
1378 out->stream.write = out_write;
1379 out->stream.get_render_position = out_get_render_position;
1381 /* TODO: for now we always use endpoint 0 */
1382 out->ep = &audio_endpoints[0];
1384 if (!open_endpoint(out->ep, &out->cfg))
1387 DBG("rate=%d channels=%d format=%d", out->cfg.rate,
1388 out->cfg.channels, out->cfg.format);
1390 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1391 out->downmix_buf = malloc(FIXED_BUFFER_SIZE / 2);
1392 if (!out->downmix_buf)
1396 *stream_out = &out->stream;
1397 a2dp_dev->out = out;
1399 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1404 error("audio: cannot open output stream");
1410 static void audio_close_output_stream(struct audio_hw_device *dev,
1411 struct audio_stream_out *stream)
1413 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1414 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1418 close_endpoint(a2dp_dev->out->ep);
1420 free(out->downmix_buf);
1423 a2dp_dev->out = NULL;
1426 static int audio_set_parameters(struct audio_hw_device *dev,
1427 const char *kvpairs)
1429 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1430 struct a2dp_stream_out *out = a2dp_dev->out;
1437 return out->stream.common.set_parameters((struct audio_stream *) out,
1441 static char *audio_get_parameters(const struct audio_hw_device *dev,
1448 static int audio_init_check(const struct audio_hw_device *dev)
1454 static int audio_set_voice_volume(struct audio_hw_device *dev, float volume)
1460 static int audio_set_master_volume(struct audio_hw_device *dev, float volume)
1466 static int audio_set_mode(struct audio_hw_device *dev, int mode)
1472 static int audio_set_mic_mute(struct audio_hw_device *dev, bool state)
1478 static int audio_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1484 static size_t audio_get_input_buffer_size(const struct audio_hw_device *dev,
1485 const struct audio_config *config)
1491 static int audio_open_input_stream(struct audio_hw_device *dev,
1492 audio_io_handle_t handle,
1493 audio_devices_t devices,
1494 struct audio_config *config,
1495 struct audio_stream_in **stream_in)
1497 struct audio_stream_in *in;
1501 in = calloc(1, sizeof(struct audio_stream_in));
1505 in->common.get_sample_rate = in_get_sample_rate;
1506 in->common.set_sample_rate = in_set_sample_rate;
1507 in->common.get_buffer_size = in_get_buffer_size;
1508 in->common.get_channels = in_get_channels;
1509 in->common.get_format = in_get_format;
1510 in->common.set_format = in_set_format;
1511 in->common.standby = in_standby;
1512 in->common.dump = in_dump;
1513 in->common.set_parameters = in_set_parameters;
1514 in->common.get_parameters = in_get_parameters;
1515 in->common.add_audio_effect = in_add_audio_effect;
1516 in->common.remove_audio_effect = in_remove_audio_effect;
1517 in->set_gain = in_set_gain;
1519 in->get_input_frames_lost = in_get_input_frames_lost;
1526 static void audio_close_input_stream(struct audio_hw_device *dev,
1527 struct audio_stream_in *stream_in)
1533 static int audio_dump(const audio_hw_device_t *device, int fd)
1539 static int audio_close(hw_device_t *device)
1541 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
1545 unregister_endpoints();
1547 shutdown(listen_sk, SHUT_RDWR);
1548 shutdown(audio_sk, SHUT_RDWR);
1550 pthread_join(ipc_th, NULL);
1559 static void *ipc_handler(void *data)
1568 DBG("Waiting for connection ...");
1570 sk = accept(listen_sk, NULL, NULL);
1577 if (err != ECONNABORTED && err != EINVAL)
1578 error("audio: Failed to accept socket: %d (%s)",
1579 err, strerror(err));
1584 pthread_mutex_lock(&sk_mutex);
1586 pthread_mutex_unlock(&sk_mutex);
1588 DBG("Audio IPC: Connected");
1590 if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
1591 error("audio: Failed to register endpoints");
1593 unregister_endpoints();
1595 pthread_mutex_lock(&sk_mutex);
1596 shutdown(audio_sk, SHUT_RDWR);
1599 pthread_mutex_unlock(&sk_mutex);
1604 memset(&pfd, 0, sizeof(pfd));
1606 pfd.events = POLLHUP | POLLERR | POLLNVAL;
1608 /* Check if socket is still alive. Empty while loop.*/
1609 while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
1611 if (pfd.revents & (POLLHUP | POLLERR | POLLNVAL)) {
1612 info("Audio HAL: Socket closed");
1614 pthread_mutex_lock(&sk_mutex);
1617 pthread_mutex_unlock(&sk_mutex);
1621 /* audio_sk is closed at this point, just cleanup endpoints states */
1622 memset(audio_endpoints, 0, sizeof(audio_endpoints));
1624 info("Closing Audio IPC thread");
1628 static int audio_ipc_init(void)
1630 struct sockaddr_un addr;
1636 sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
1639 error("audio: Failed to create socket: %d (%s)", -err,
1644 memset(&addr, 0, sizeof(addr));
1645 addr.sun_family = AF_UNIX;
1647 memcpy(addr.sun_path, BLUEZ_AUDIO_SK_PATH,
1648 sizeof(BLUEZ_AUDIO_SK_PATH));
1650 if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
1652 error("audio: Failed to bind socket: %d (%s)", -err,
1657 if (listen(sk, 1) < 0) {
1659 error("audio: Failed to listen on the socket: %d (%s)", -err,
1666 err = pthread_create(&ipc_th, NULL, ipc_handler, NULL);
1670 error("audio: Failed to start Audio IPC thread: %d (%s)",
1671 -err, strerror(-err));
1682 static int audio_open(const hw_module_t *module, const char *name,
1683 hw_device_t **device)
1685 struct a2dp_audio_dev *a2dp_dev;
1690 if (strcmp(name, AUDIO_HARDWARE_INTERFACE)) {
1691 error("audio: interface %s not matching [%s]", name,
1692 AUDIO_HARDWARE_INTERFACE);
1696 err = audio_ipc_init();
1700 a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
1704 a2dp_dev->dev.common.tag = HARDWARE_DEVICE_TAG;
1705 a2dp_dev->dev.common.version = AUDIO_DEVICE_API_VERSION_CURRENT;
1706 a2dp_dev->dev.common.module = (struct hw_module_t *) module;
1707 a2dp_dev->dev.common.close = audio_close;
1709 a2dp_dev->dev.init_check = audio_init_check;
1710 a2dp_dev->dev.set_voice_volume = audio_set_voice_volume;
1711 a2dp_dev->dev.set_master_volume = audio_set_master_volume;
1712 a2dp_dev->dev.set_mode = audio_set_mode;
1713 a2dp_dev->dev.set_mic_mute = audio_set_mic_mute;
1714 a2dp_dev->dev.get_mic_mute = audio_get_mic_mute;
1715 a2dp_dev->dev.set_parameters = audio_set_parameters;
1716 a2dp_dev->dev.get_parameters = audio_get_parameters;
1717 a2dp_dev->dev.get_input_buffer_size = audio_get_input_buffer_size;
1718 a2dp_dev->dev.open_output_stream = audio_open_output_stream;
1719 a2dp_dev->dev.close_output_stream = audio_close_output_stream;
1720 a2dp_dev->dev.open_input_stream = audio_open_input_stream;
1721 a2dp_dev->dev.close_input_stream = audio_close_input_stream;
1722 a2dp_dev->dev.dump = audio_dump;
1724 /* Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
1725 * This results from the structure of following structs:a2dp_audio_dev,
1726 * audio_hw_device. We will rely on this later in the code.*/
1727 *device = &a2dp_dev->dev.common;
1732 static struct hw_module_methods_t hal_module_methods = {
1736 struct audio_module HAL_MODULE_INFO_SYM = {
1738 .tag = HARDWARE_MODULE_TAG,
1741 .id = AUDIO_HARDWARE_MODULE_ID,
1742 .name = "A2DP Bluez HW HAL",
1743 .author = "Intel Corporation",
1744 .methods = &hal_module_methods,