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;
135 unsigned frame_duration;
136 unsigned frames_per_packet;
138 struct timespec start;
139 unsigned frames_sent;
145 static inline void timespec_diff(struct timespec *a, struct timespec *b,
146 struct timespec *res)
148 res->tv_sec = a->tv_sec - b->tv_sec;
149 res->tv_nsec = a->tv_nsec - b->tv_nsec;
151 if (res->tv_nsec < 0) {
153 res->tv_nsec += 1000000000; /* 1sec */
157 static int sbc_get_presets(struct audio_preset *preset, size_t *len);
158 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
160 static int sbc_cleanup(void *codec_data);
161 static int sbc_get_config(void *codec_data, struct audio_input_config *config);
162 static size_t sbc_get_buffer_size(void *codec_data);
163 static size_t sbc_get_mediapacket_duration(void *codec_data);
164 static void sbc_resume(void *codec_data);
165 static ssize_t sbc_write_data(void *codec_data, const void *buffer,
166 size_t bytes, int fd);
171 int (*get_presets) (struct audio_preset *preset, size_t *len);
173 int (*init) (struct audio_preset *preset, uint16_t mtu,
175 int (*cleanup) (void *codec_data);
176 int (*get_config) (void *codec_data,
177 struct audio_input_config *config);
178 size_t (*get_buffer_size) (void *codec_data);
179 size_t (*get_mediapacket_duration) (void *codec_data);
180 void (*resume) (void *codec_data);
181 ssize_t (*write_data) (void *codec_data, const void *buffer,
182 size_t bytes, int fd);
185 static const struct audio_codec audio_codecs[] = {
187 .type = A2DP_CODEC_SBC,
189 .get_presets = sbc_get_presets,
191 .init = sbc_codec_init,
192 .cleanup = sbc_cleanup,
193 .get_config = sbc_get_config,
194 .get_buffer_size = sbc_get_buffer_size,
195 .get_mediapacket_duration = sbc_get_mediapacket_duration,
196 .resume = sbc_resume,
197 .write_data = sbc_write_data,
201 #define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
203 #define MAX_AUDIO_ENDPOINTS NUM_CODECS
205 struct audio_endpoint {
207 const struct audio_codec *codec;
212 static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
215 AUDIO_A2DP_STATE_NONE,
216 AUDIO_A2DP_STATE_STANDBY,
217 AUDIO_A2DP_STATE_SUSPENDED,
218 AUDIO_A2DP_STATE_STARTED
221 struct a2dp_stream_out {
222 struct audio_stream_out stream;
224 struct audio_endpoint *ep;
225 enum a2dp_state_t audio_state;
226 struct audio_input_config cfg;
228 uint8_t *downmix_buf;
231 struct a2dp_audio_dev {
232 struct audio_hw_device dev;
233 struct a2dp_stream_out *out;
236 static const a2dp_sbc_t sbc_presets[] = {
238 .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
239 .channel_mode = SBC_CHANNEL_MODE_MONO |
240 SBC_CHANNEL_MODE_DUAL_CHANNEL |
241 SBC_CHANNEL_MODE_STEREO |
242 SBC_CHANNEL_MODE_JOINT_STEREO,
243 .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
244 .allocation_method = SBC_ALLOCATION_SNR |
245 SBC_ALLOCATION_LOUDNESS,
246 .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
247 SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
248 .min_bitpool = MIN_BITPOOL,
249 .max_bitpool = MAX_BITPOOL
252 .frequency = SBC_SAMPLING_FREQ_44100,
253 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
254 .subbands = SBC_SUBBANDS_8,
255 .allocation_method = SBC_ALLOCATION_LOUDNESS,
256 .block_length = SBC_BLOCK_LENGTH_16,
257 .min_bitpool = MIN_BITPOOL,
258 .max_bitpool = MAX_BITPOOL
261 .frequency = SBC_SAMPLING_FREQ_48000,
262 .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
263 .subbands = SBC_SUBBANDS_8,
264 .allocation_method = SBC_ALLOCATION_LOUDNESS,
265 .block_length = SBC_BLOCK_LENGTH_16,
266 .min_bitpool = MIN_BITPOOL,
267 .max_bitpool = MAX_BITPOOL
271 static int sbc_get_presets(struct audio_preset *preset, size_t *len)
276 uint8_t *ptr = (uint8_t *) preset;
277 size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
279 count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
281 for (i = 0; i < count; i++) {
282 preset = (struct audio_preset *) ptr;
284 if (new_len + preset_size > *len)
287 preset->len = sizeof(a2dp_sbc_t);
288 memcpy(preset->data, &sbc_presets[i], preset->len);
290 new_len += preset_size;
299 static int sbc_freq2int(uint8_t freq)
302 case SBC_SAMPLING_FREQ_16000:
304 case SBC_SAMPLING_FREQ_32000:
306 case SBC_SAMPLING_FREQ_44100:
308 case SBC_SAMPLING_FREQ_48000:
315 static const char *sbc_mode2str(uint8_t mode)
318 case SBC_CHANNEL_MODE_MONO:
320 case SBC_CHANNEL_MODE_DUAL_CHANNEL:
321 return "DualChannel";
322 case SBC_CHANNEL_MODE_STEREO:
324 case SBC_CHANNEL_MODE_JOINT_STEREO:
325 return "JointStereo";
331 static int sbc_blocks2int(uint8_t blocks)
334 case SBC_BLOCK_LENGTH_4:
336 case SBC_BLOCK_LENGTH_8:
338 case SBC_BLOCK_LENGTH_12:
340 case SBC_BLOCK_LENGTH_16:
347 static int sbc_subbands2int(uint8_t subbands)
359 static const char *sbc_allocation2str(uint8_t allocation)
361 switch (allocation) {
362 case SBC_ALLOCATION_SNR:
364 case SBC_ALLOCATION_LOUDNESS:
371 static void sbc_init_encoder(struct sbc_data *sbc_data)
373 a2dp_sbc_t *in = &sbc_data->sbc;
374 sbc_t *out = &sbc_data->enc;
376 sbc_init_a2dp(out, 0L, in, sizeof(*in));
378 out->endian = SBC_LE;
379 out->bitpool = in->max_bitpool;
381 DBG("frequency=%d channel_mode=%s block_length=%d subbands=%d "
382 "allocation=%s bitpool=%d-%d",
383 sbc_freq2int(in->frequency),
384 sbc_mode2str(in->channel_mode),
385 sbc_blocks2int(in->block_length),
386 sbc_subbands2int(in->subbands),
387 sbc_allocation2str(in->allocation_method),
388 in->min_bitpool, in->max_bitpool);
391 static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
394 struct sbc_data *sbc_data;
395 size_t hdr_len = sizeof(struct media_packet);
397 size_t out_frame_len;
400 if (preset->len != sizeof(a2dp_sbc_t)) {
401 error("SBC: preset size mismatch");
402 return AUDIO_STATUS_FAILED;
405 sbc_data = calloc(sizeof(struct sbc_data), 1);
407 return AUDIO_STATUS_FAILED;
409 memcpy(&sbc_data->sbc, preset->data, preset->len);
411 sbc_init_encoder(sbc_data);
413 in_frame_len = sbc_get_codesize(&sbc_data->enc);
414 out_frame_len = sbc_get_frame_length(&sbc_data->enc);
415 num_frames = (mtu - hdr_len) / out_frame_len;
417 sbc_data->in_frame_len = in_frame_len;
418 sbc_data->in_buf_size = num_frames * in_frame_len;
420 sbc_data->out_buf_size = hdr_len + num_frames * out_frame_len;
421 sbc_data->out_buf = calloc(1, sbc_data->out_buf_size);
423 sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
424 sbc_data->frames_per_packet = num_frames;
426 DBG("mtu=%u in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
427 mtu, in_frame_len, out_frame_len, num_frames);
429 *codec_data = sbc_data;
431 return AUDIO_STATUS_SUCCESS;
434 static int sbc_cleanup(void *codec_data)
436 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
438 sbc_finish(&sbc_data->enc);
439 free(sbc_data->out_buf);
442 return AUDIO_STATUS_SUCCESS;
445 static int sbc_get_config(void *codec_data, struct audio_input_config *config)
447 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
449 switch (sbc_data->sbc.frequency) {
450 case SBC_SAMPLING_FREQ_16000:
451 config->rate = 16000;
453 case SBC_SAMPLING_FREQ_32000:
454 config->rate = 32000;
456 case SBC_SAMPLING_FREQ_44100:
457 config->rate = 44100;
459 case SBC_SAMPLING_FREQ_48000:
460 config->rate = 48000;
463 return AUDIO_STATUS_FAILED;
465 config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
466 AUDIO_CHANNEL_OUT_MONO :
467 AUDIO_CHANNEL_OUT_STEREO;
468 config->format = AUDIO_FORMAT_PCM_16_BIT;
470 return AUDIO_STATUS_SUCCESS;
473 static size_t sbc_get_buffer_size(void *codec_data)
475 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
477 return sbc_data->in_buf_size;
480 static size_t sbc_get_mediapacket_duration(void *codec_data)
482 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
484 return sbc_data->frame_duration * sbc_data->frames_per_packet;
487 static void sbc_resume(void *codec_data)
489 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
493 clock_gettime(CLOCK_MONOTONIC, &sbc_data->start);
495 sbc_data->frames_sent = 0;
496 sbc_data->timestamp = 0;
499 static int write_media_packet(int fd, struct sbc_data *sbc_data,
500 struct media_packet *mp, size_t data_len)
503 struct timespec diff;
504 unsigned expected_frames;
508 ret = write(fd, mp, sizeof(*mp) + data_len);
516 sbc_data->frames_sent += mp->payload.frame_count;
518 clock_gettime(CLOCK_MONOTONIC, &cur);
519 timespec_diff(&cur, &sbc_data->start, &diff);
520 expected_frames = (diff.tv_sec * 1000000 + diff.tv_nsec / 1000) /
521 sbc_data->frame_duration;
523 /* AudioFlinger does not seem to provide any *working*
524 * API to provide data in some interval and will just
525 * send another buffer as soon as we process current
526 * one. To prevent overflowing L2CAP socket, we need to
527 * introduce some artificial delay here base on how many
528 * audio frames were sent so far, i.e. if we're not
529 * lagging behind audio stream, we can sleep for
530 * duration of single media packet.
532 if (sbc_data->frames_sent >= expected_frames)
533 usleep(sbc_data->frame_duration *
534 mp->payload.frame_count);
539 static ssize_t sbc_write_data(void *codec_data, const void *buffer,
540 size_t bytes, int fd)
542 struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
545 struct media_packet *mp = (struct media_packet *) sbc_data->out_buf;
546 size_t free_space = sbc_data->out_buf_size - sizeof(*mp);
552 mp->hdr.ssrc = htonl(1);
553 mp->hdr.timestamp = htonl(sbc_data->timestamp);
554 mp->payload.frame_count = 0;
556 while (bytes - consumed >= sbc_data->in_frame_len) {
559 bytes_read = sbc_encode(&sbc_data->enc, buffer + consumed,
560 sbc_data->in_frame_len,
561 mp->data + encoded, free_space,
564 if (bytes_read < 0) {
565 error("SBC: failed to encode block (%zd)", bytes_read);
569 mp->payload.frame_count++;
571 consumed += bytes_read;
573 free_space -= written;
575 /* AudioFlinger provides PCM 16bit stereo only, thus sample size
578 sbc_data->timestamp += (bytes_read / 4);
580 /* write data if we either filled media packed or encoded all
583 if (mp->payload.frame_count == sbc_data->frames_per_packet ||
585 mp->payload.frame_count ==
586 MAX_FRAMES_IN_PAYLOAD) {
587 mp->hdr.sequence_number = htons(sbc_data->seq++);
589 ret = write_media_packet(fd, sbc_data, mp, encoded);
594 free_space = sbc_data->out_buf_size - sizeof(*mp);
595 mp->hdr.timestamp = htonl(sbc_data->timestamp);
596 mp->payload.frame_count = 0;
600 if (consumed != bytes) {
601 /* we should encode all input data
602 * if we did not, something went wrong but we can't really
603 * handle this so this is just sanity check
605 error("SBC: failed to encode complete input buffer");
608 /* we always assume that all data was processed and sent */
612 static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
613 void *param, size_t *rsp_len, void *rsp, int *fd)
619 char cmsgbuf[CMSG_SPACE(sizeof(int))];
621 size_t s_len = sizeof(s);
623 pthread_mutex_lock(&sk_mutex);
626 error("audio: Invalid cmd socket passed to audio_ipc_cmd");
630 if (!rsp || !rsp_len) {
631 memset(&s, 0, s_len);
636 memset(&msg, 0, sizeof(msg));
637 memset(&cmd, 0, sizeof(cmd));
639 cmd.service_id = service_id;
643 iv[0].iov_base = &cmd;
644 iv[0].iov_len = sizeof(cmd);
646 iv[1].iov_base = param;
652 ret = sendmsg(audio_sk, &msg, 0);
654 error("audio: Sending command failed:%s", strerror(errno));
658 /* socket was shutdown */
660 error("audio: Command socket closed");
664 memset(&msg, 0, sizeof(msg));
665 memset(&cmd, 0, sizeof(cmd));
667 iv[0].iov_base = &cmd;
668 iv[0].iov_len = sizeof(cmd);
670 iv[1].iov_base = rsp;
671 iv[1].iov_len = *rsp_len;
677 memset(cmsgbuf, 0, sizeof(cmsgbuf));
678 msg.msg_control = cmsgbuf;
679 msg.msg_controllen = sizeof(cmsgbuf);
682 ret = recvmsg(audio_sk, &msg, 0);
684 error("audio: Receiving command response failed:%s",
689 if (ret < (ssize_t) sizeof(cmd)) {
690 error("audio: Too small response received(%zd bytes)", ret);
694 if (cmd.service_id != service_id) {
695 error("audio: Invalid service id (%u vs %u)", cmd.service_id,
700 if (ret != (ssize_t) (sizeof(cmd) + cmd.len)) {
701 error("audio: Malformed response received(%zd bytes)", ret);
705 if (cmd.opcode != opcode && cmd.opcode != AUDIO_OP_STATUS) {
706 error("audio: Invalid opcode received (%u vs %u)",
711 if (cmd.opcode == AUDIO_OP_STATUS) {
712 struct ipc_status *s = rsp;
714 if (sizeof(*s) != cmd.len) {
715 error("audio: Invalid status length");
719 if (s->code == AUDIO_STATUS_SUCCESS) {
720 error("audio: Invalid success status response");
724 pthread_mutex_unlock(&sk_mutex);
729 pthread_mutex_unlock(&sk_mutex);
731 /* Receive auxiliary data in msg */
733 struct cmsghdr *cmsg;
737 for (cmsg = CMSG_FIRSTHDR(&msg); cmsg;
738 cmsg = CMSG_NXTHDR(&msg, cmsg)) {
739 if (cmsg->cmsg_level == SOL_SOCKET
740 && cmsg->cmsg_type == SCM_RIGHTS) {
741 memcpy(fd, CMSG_DATA(cmsg), sizeof(int));
753 return AUDIO_STATUS_SUCCESS;
756 /* Some serious issue happen on IPC - recover */
757 shutdown(audio_sk, SHUT_RDWR);
758 pthread_mutex_unlock(&sk_mutex);
760 return AUDIO_STATUS_FAILED;
763 static int ipc_open_cmd(const struct audio_codec *codec)
765 uint8_t buf[BLUEZ_AUDIO_MTU];
766 struct audio_cmd_open *cmd = (struct audio_cmd_open *) buf;
767 struct audio_rsp_open rsp;
768 size_t cmd_len = sizeof(buf) - sizeof(*cmd);
769 size_t rsp_len = sizeof(rsp);
774 memcpy(cmd->uuid, a2dp_src_uuid, sizeof(a2dp_src_uuid));
776 cmd->codec = codec->type;
777 cmd->presets = codec->get_presets(cmd->preset, &cmd_len);
779 cmd_len += sizeof(*cmd);
781 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN, cmd_len, cmd,
782 &rsp_len, &rsp, NULL);
784 if (result != AUDIO_STATUS_SUCCESS)
790 static int ipc_close_cmd(uint8_t endpoint_id)
792 struct audio_cmd_close cmd;
797 cmd.id = endpoint_id;
799 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE,
800 sizeof(cmd), &cmd, NULL, NULL, NULL);
805 static int ipc_open_stream_cmd(uint8_t endpoint_id, uint16_t *mtu, int *fd,
806 struct audio_preset **caps)
808 char buf[BLUEZ_AUDIO_MTU];
809 struct audio_cmd_open_stream cmd;
810 struct audio_rsp_open_stream *rsp =
811 (struct audio_rsp_open_stream *) &buf;
812 size_t rsp_len = sizeof(buf);
818 return AUDIO_STATUS_FAILED;
820 cmd.id = endpoint_id;
822 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN_STREAM,
823 sizeof(cmd), &cmd, &rsp_len, rsp, fd);
824 if (result == AUDIO_STATUS_SUCCESS) {
825 size_t buf_len = sizeof(struct audio_preset) +
828 *caps = malloc(buf_len);
829 memcpy(*caps, &rsp->preset, buf_len);
837 static int ipc_close_stream_cmd(uint8_t endpoint_id)
839 struct audio_cmd_close_stream cmd;
844 cmd.id = endpoint_id;
846 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE_STREAM,
847 sizeof(cmd), &cmd, NULL, NULL, NULL);
852 static int ipc_resume_stream_cmd(uint8_t endpoint_id)
854 struct audio_cmd_resume_stream cmd;
859 cmd.id = endpoint_id;
861 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_RESUME_STREAM,
862 sizeof(cmd), &cmd, NULL, NULL, NULL);
867 static int ipc_suspend_stream_cmd(uint8_t endpoint_id)
869 struct audio_cmd_suspend_stream cmd;
874 cmd.id = endpoint_id;
876 result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_SUSPEND_STREAM,
877 sizeof(cmd), &cmd, NULL, NULL, NULL);
882 static int register_endpoints(void)
884 struct audio_endpoint *ep = &audio_endpoints[0];
887 for (i = 0; i < NUM_CODECS; i++, ep++) {
888 const struct audio_codec *codec = &audio_codecs[i];
890 ep->id = ipc_open_cmd(codec);
893 return AUDIO_STATUS_FAILED;
896 ep->codec_data = NULL;
900 return AUDIO_STATUS_SUCCESS;
903 static void unregister_endpoints(void)
907 for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++) {
908 struct audio_endpoint *ep = &audio_endpoints[i];
911 ipc_close_cmd(ep->id);
912 memset(ep, 0, sizeof(*ep));
917 static int set_blocking(int fd)
921 flags = fcntl(fd, F_GETFL, 0);
924 error("fcntl(F_GETFL): %s (%d)", strerror(-err), -err);
928 if (fcntl(fd, F_SETFL, flags & ~O_NONBLOCK) < 0) {
930 error("fcntl(F_SETFL): %s (%d)", strerror(-err), -err);
937 static bool open_endpoint(struct audio_endpoint *ep,
938 struct audio_input_config *cfg)
940 struct audio_preset *preset;
941 const struct audio_codec *codec;
945 if (ipc_open_stream_cmd(ep->id, &mtu, &fd, &preset) !=
946 AUDIO_STATUS_SUCCESS)
949 if (set_blocking(fd) < 0)
955 codec->init(preset, mtu, &ep->codec_data);
956 codec->get_config(ep->codec_data, cfg);
969 static void close_endpoint(struct audio_endpoint *ep)
971 ipc_close_stream_cmd(ep->id);
977 ep->codec->cleanup(ep->codec_data);
978 ep->codec_data = NULL;
981 static void downmix_to_mono(struct a2dp_stream_out *out, const uint8_t *buffer,
984 const int16_t *input = (const void *) buffer;
985 int16_t *output = (void *) out->downmix_buf;
988 for (i = 0; i < bytes / 2; i++) {
989 int16_t l = le16_to_cpu(get_unaligned(&input[i * 2]));
990 int16_t r = le16_to_cpu(get_unaligned(&input[i * 2 + 1]));
992 put_unaligned(cpu_to_le16((l + r) / 2), &output[i]);
996 static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
999 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1001 /* just return in case we're closing */
1002 if (out->audio_state == AUDIO_A2DP_STATE_NONE)
1005 /* We can auto-start only from standby */
1006 if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
1007 DBG("stream in standby, auto-start");
1009 if (ipc_resume_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1012 out->ep->codec->resume(out->ep->codec_data);
1014 out->audio_state = AUDIO_A2DP_STATE_STARTED;
1017 if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
1018 error("audio: stream not started");
1022 if (out->ep->fd < 0) {
1023 error("audio: no transport socket");
1027 /* currently Android audioflinger is not able to provide mono stream on
1028 * A2DP output so down mixing needs to be done in hal-audio plugin.
1031 * AudioFlinger::PlaybackThread::readOutputParameters()
1032 * frameworks/av/services/audioflinger/Threads.cpp:1631
1034 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1035 if (!out->downmix_buf) {
1036 error("audio: downmix buffer not initialized");
1040 downmix_to_mono(out, buffer, bytes);
1042 return out->ep->codec->write_data(out->ep->codec_data,
1048 return out->ep->codec->write_data(out->ep->codec_data, buffer,
1049 bytes, out->ep->fd);
1052 static uint32_t out_get_sample_rate(const struct audio_stream *stream)
1054 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1058 return out->cfg.rate;
1061 static int out_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1063 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1067 if (rate != out->cfg.rate) {
1068 warn("audio: cannot set sample rate to %d", rate);
1075 static size_t out_get_buffer_size(const struct audio_stream *stream)
1079 /* We should return proper buffer size calculated by codec (so each
1080 * input buffer is encoded into single media packed) but this does not
1081 * work well with AudioFlinger and causes problems. For this reason we
1082 * use magic value here and out_write code takes care of splitting
1083 * input buffer into multiple media packets.
1085 return FIXED_BUFFER_SIZE;
1088 static uint32_t out_get_channels(const struct audio_stream *stream)
1092 /* AudioFlinger can only provide stereo stream, so we return it here and
1093 * later we'll downmix this to mono in case codec requires it
1096 return AUDIO_CHANNEL_OUT_STEREO;
1099 static audio_format_t out_get_format(const struct audio_stream *stream)
1101 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1105 return out->cfg.format;
1108 static int out_set_format(struct audio_stream *stream, audio_format_t format)
1114 static int out_standby(struct audio_stream *stream)
1116 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1120 if (out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1121 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1123 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1129 static int out_dump(const struct audio_stream *stream, int fd)
1135 static int out_set_parameters(struct audio_stream *stream, const char *kvpairs)
1137 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1141 bool enter_suspend = false;
1142 bool exit_suspend = false;
1146 str = strdup(kvpairs);
1147 kvpair = strtok_r(str, ";", &saveptr);
1149 for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
1152 keyval = strchr(kvpair, '=');
1159 if (!strcmp(kvpair, "closing")) {
1160 if (!strcmp(keyval, "true"))
1161 out->audio_state = AUDIO_A2DP_STATE_NONE;
1162 } else if (!strcmp(kvpair, "A2dpSuspended")) {
1163 if (!strcmp(keyval, "true"))
1164 enter_suspend = true;
1166 exit_suspend = true;
1172 if (enter_suspend && out->audio_state == AUDIO_A2DP_STATE_STARTED) {
1173 if (ipc_suspend_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
1175 out->audio_state = AUDIO_A2DP_STATE_SUSPENDED;
1178 if (exit_suspend && out->audio_state == AUDIO_A2DP_STATE_SUSPENDED)
1179 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1184 static char *out_get_parameters(const struct audio_stream *stream,
1191 static uint32_t out_get_latency(const struct audio_stream_out *stream)
1193 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1194 struct audio_endpoint *ep = out->ep;
1195 size_t pkt_duration;
1199 pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
1201 return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
1204 static int out_set_volume(struct audio_stream_out *stream, float left,
1208 /* volume controlled in audioflinger mixer (digital) */
1212 static int out_get_render_position(const struct audio_stream_out *stream,
1213 uint32_t *dsp_frames)
1219 static int out_add_audio_effect(const struct audio_stream *stream,
1220 effect_handle_t effect)
1226 static int out_remove_audio_effect(const struct audio_stream *stream,
1227 effect_handle_t effect)
1233 static uint32_t in_get_sample_rate(const struct audio_stream *stream)
1239 static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
1245 static size_t in_get_buffer_size(const struct audio_stream *stream)
1251 static uint32_t in_get_channels(const struct audio_stream *stream)
1257 static audio_format_t in_get_format(const struct audio_stream *stream)
1263 static int in_set_format(struct audio_stream *stream, audio_format_t format)
1269 static int in_standby(struct audio_stream *stream)
1275 static int in_dump(const struct audio_stream *stream, int fd)
1281 static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
1287 static char *in_get_parameters(const struct audio_stream *stream,
1294 static int in_set_gain(struct audio_stream_in *stream, float gain)
1300 static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
1307 static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
1313 static int in_add_audio_effect(const struct audio_stream *stream,
1314 effect_handle_t effect)
1320 static int in_remove_audio_effect(const struct audio_stream *stream,
1321 effect_handle_t effect)
1327 static int audio_open_output_stream(struct audio_hw_device *dev,
1328 audio_io_handle_t handle,
1329 audio_devices_t devices,
1330 audio_output_flags_t flags,
1331 struct audio_config *config,
1332 struct audio_stream_out **stream_out)
1335 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1336 struct a2dp_stream_out *out;
1338 out = calloc(1, sizeof(struct a2dp_stream_out));
1344 out->stream.common.get_sample_rate = out_get_sample_rate;
1345 out->stream.common.set_sample_rate = out_set_sample_rate;
1346 out->stream.common.get_buffer_size = out_get_buffer_size;
1347 out->stream.common.get_channels = out_get_channels;
1348 out->stream.common.get_format = out_get_format;
1349 out->stream.common.set_format = out_set_format;
1350 out->stream.common.standby = out_standby;
1351 out->stream.common.dump = out_dump;
1352 out->stream.common.set_parameters = out_set_parameters;
1353 out->stream.common.get_parameters = out_get_parameters;
1354 out->stream.common.add_audio_effect = out_add_audio_effect;
1355 out->stream.common.remove_audio_effect = out_remove_audio_effect;
1356 out->stream.get_latency = out_get_latency;
1357 out->stream.set_volume = out_set_volume;
1358 out->stream.write = out_write;
1359 out->stream.get_render_position = out_get_render_position;
1361 /* TODO: for now we always use endpoint 0 */
1362 out->ep = &audio_endpoints[0];
1364 if (!open_endpoint(out->ep, &out->cfg))
1367 DBG("rate=%d channels=%d format=%d", out->cfg.rate,
1368 out->cfg.channels, out->cfg.format);
1370 if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
1371 out->downmix_buf = malloc(FIXED_BUFFER_SIZE / 2);
1372 if (!out->downmix_buf)
1376 *stream_out = &out->stream;
1377 a2dp_dev->out = out;
1379 out->audio_state = AUDIO_A2DP_STATE_STANDBY;
1384 error("audio: cannot open output stream");
1390 static void audio_close_output_stream(struct audio_hw_device *dev,
1391 struct audio_stream_out *stream)
1393 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1394 struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
1398 close_endpoint(a2dp_dev->out->ep);
1400 free(out->downmix_buf);
1403 a2dp_dev->out = NULL;
1406 static int audio_set_parameters(struct audio_hw_device *dev,
1407 const char *kvpairs)
1409 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
1410 struct a2dp_stream_out *out = a2dp_dev->out;
1417 return out->stream.common.set_parameters((struct audio_stream *) out,
1421 static char *audio_get_parameters(const struct audio_hw_device *dev,
1428 static int audio_init_check(const struct audio_hw_device *dev)
1434 static int audio_set_voice_volume(struct audio_hw_device *dev, float volume)
1440 static int audio_set_master_volume(struct audio_hw_device *dev, float volume)
1446 static int audio_set_mode(struct audio_hw_device *dev, int mode)
1452 static int audio_set_mic_mute(struct audio_hw_device *dev, bool state)
1458 static int audio_get_mic_mute(const struct audio_hw_device *dev, bool *state)
1464 static size_t audio_get_input_buffer_size(const struct audio_hw_device *dev,
1465 const struct audio_config *config)
1471 static int audio_open_input_stream(struct audio_hw_device *dev,
1472 audio_io_handle_t handle,
1473 audio_devices_t devices,
1474 struct audio_config *config,
1475 struct audio_stream_in **stream_in)
1477 struct audio_stream_in *in;
1481 in = calloc(1, sizeof(struct audio_stream_in));
1485 in->common.get_sample_rate = in_get_sample_rate;
1486 in->common.set_sample_rate = in_set_sample_rate;
1487 in->common.get_buffer_size = in_get_buffer_size;
1488 in->common.get_channels = in_get_channels;
1489 in->common.get_format = in_get_format;
1490 in->common.set_format = in_set_format;
1491 in->common.standby = in_standby;
1492 in->common.dump = in_dump;
1493 in->common.set_parameters = in_set_parameters;
1494 in->common.get_parameters = in_get_parameters;
1495 in->common.add_audio_effect = in_add_audio_effect;
1496 in->common.remove_audio_effect = in_remove_audio_effect;
1497 in->set_gain = in_set_gain;
1499 in->get_input_frames_lost = in_get_input_frames_lost;
1506 static void audio_close_input_stream(struct audio_hw_device *dev,
1507 struct audio_stream_in *stream_in)
1513 static int audio_dump(const audio_hw_device_t *device, int fd)
1519 static int audio_close(hw_device_t *device)
1521 struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
1525 unregister_endpoints();
1527 shutdown(listen_sk, SHUT_RDWR);
1528 shutdown(audio_sk, SHUT_RDWR);
1530 pthread_join(ipc_th, NULL);
1539 static void *ipc_handler(void *data)
1548 DBG("Waiting for connection ...");
1550 sk = accept(listen_sk, NULL, NULL);
1557 if (err != ECONNABORTED && err != EINVAL)
1558 error("audio: Failed to accept socket: %d (%s)",
1559 err, strerror(err));
1564 pthread_mutex_lock(&sk_mutex);
1566 pthread_mutex_unlock(&sk_mutex);
1568 DBG("Audio IPC: Connected");
1570 if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
1571 error("audio: Failed to register endpoints");
1573 unregister_endpoints();
1575 pthread_mutex_lock(&sk_mutex);
1576 shutdown(audio_sk, SHUT_RDWR);
1579 pthread_mutex_unlock(&sk_mutex);
1584 memset(&pfd, 0, sizeof(pfd));
1586 pfd.events = POLLHUP | POLLERR | POLLNVAL;
1588 /* Check if socket is still alive. Empty while loop.*/
1589 while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
1591 if (pfd.revents & (POLLHUP | POLLERR | POLLNVAL)) {
1592 info("Audio HAL: Socket closed");
1594 pthread_mutex_lock(&sk_mutex);
1597 pthread_mutex_unlock(&sk_mutex);
1601 /* audio_sk is closed at this point, just cleanup endpoints states */
1602 memset(audio_endpoints, 0, sizeof(audio_endpoints));
1604 info("Closing Audio IPC thread");
1608 static int audio_ipc_init(void)
1610 struct sockaddr_un addr;
1616 sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
1619 error("audio: Failed to create socket: %d (%s)", -err,
1624 memset(&addr, 0, sizeof(addr));
1625 addr.sun_family = AF_UNIX;
1627 memcpy(addr.sun_path, BLUEZ_AUDIO_SK_PATH,
1628 sizeof(BLUEZ_AUDIO_SK_PATH));
1630 if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
1632 error("audio: Failed to bind socket: %d (%s)", -err,
1637 if (listen(sk, 1) < 0) {
1639 error("audio: Failed to listen on the socket: %d (%s)", -err,
1646 err = pthread_create(&ipc_th, NULL, ipc_handler, NULL);
1650 error("audio: Failed to start Audio IPC thread: %d (%s)",
1651 -err, strerror(-err));
1662 static int audio_open(const hw_module_t *module, const char *name,
1663 hw_device_t **device)
1665 struct a2dp_audio_dev *a2dp_dev;
1670 if (strcmp(name, AUDIO_HARDWARE_INTERFACE)) {
1671 error("audio: interface %s not matching [%s]", name,
1672 AUDIO_HARDWARE_INTERFACE);
1676 err = audio_ipc_init();
1680 a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
1684 a2dp_dev->dev.common.tag = HARDWARE_DEVICE_TAG;
1685 a2dp_dev->dev.common.version = AUDIO_DEVICE_API_VERSION_CURRENT;
1686 a2dp_dev->dev.common.module = (struct hw_module_t *) module;
1687 a2dp_dev->dev.common.close = audio_close;
1689 a2dp_dev->dev.init_check = audio_init_check;
1690 a2dp_dev->dev.set_voice_volume = audio_set_voice_volume;
1691 a2dp_dev->dev.set_master_volume = audio_set_master_volume;
1692 a2dp_dev->dev.set_mode = audio_set_mode;
1693 a2dp_dev->dev.set_mic_mute = audio_set_mic_mute;
1694 a2dp_dev->dev.get_mic_mute = audio_get_mic_mute;
1695 a2dp_dev->dev.set_parameters = audio_set_parameters;
1696 a2dp_dev->dev.get_parameters = audio_get_parameters;
1697 a2dp_dev->dev.get_input_buffer_size = audio_get_input_buffer_size;
1698 a2dp_dev->dev.open_output_stream = audio_open_output_stream;
1699 a2dp_dev->dev.close_output_stream = audio_close_output_stream;
1700 a2dp_dev->dev.open_input_stream = audio_open_input_stream;
1701 a2dp_dev->dev.close_input_stream = audio_close_input_stream;
1702 a2dp_dev->dev.dump = audio_dump;
1704 /* Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
1705 * This results from the structure of following structs:a2dp_audio_dev,
1706 * audio_hw_device. We will rely on this later in the code.*/
1707 *device = &a2dp_dev->dev.common;
1712 static struct hw_module_methods_t hal_module_methods = {
1716 struct audio_module HAL_MODULE_INFO_SYM = {
1718 .tag = HARDWARE_MODULE_TAG,
1721 .id = AUDIO_HARDWARE_MODULE_ID,
1722 .name = "A2DP Bluez HW HAL",
1723 .author = "Intel Corporation",
1724 .methods = &hal_module_methods,