#include <sys/socket.h>
#include <sys/un.h>
#include <unistd.h>
+#include <arpa/inet.h>
+#include <fcntl.h>
#include <hardware/audio.h>
#include <hardware/hardware.h>
#include "audio-msg.h"
+#include "ipc-common.h"
#include "hal-log.h"
#include "hal-msg.h"
-#include "../profiles/audio/a2dp-codecs.h"
+#include "hal-audio.h"
+#include "hal-utils.h"
+
+#define FIXED_A2DP_PLAYBACK_LATENCY_MS 25
+
+#define FIXED_BUFFER_SIZE (20 * 512)
+
+#define MAX_DELAY 100000 /* 100ms */
static const uint8_t a2dp_src_uuid[] = {
0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
static int listen_sk = -1;
static int audio_sk = -1;
-static bool close_thread = false;
static pthread_t ipc_th = 0;
-static pthread_mutex_t close_mutex = PTHREAD_MUTEX_INITIALIZER;
static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
-struct audio_input_config {
- uint32_t rate;
- uint32_t channels;
- audio_format_t format;
-};
-
-struct sbc_data {
- a2dp_sbc_t sbc;
-};
+static void timespec_add(struct timespec *base, uint64_t time_us,
+ struct timespec *res)
+{
+ res->tv_sec = base->tv_sec + time_us / 1000000;
+ res->tv_nsec = base->tv_nsec + (time_us % 1000000) * 1000;
-static int sbc_get_presets(struct audio_preset *preset, size_t *len);
-static int sbc_init(struct audio_preset *preset, void **codec_data);
-static int sbc_cleanup(void *codec_data);
-static int sbc_get_config(void *codec_data,
- struct audio_input_config *config);
+ if (res->tv_nsec >= 1000000000) {
+ res->tv_sec++;
+ res->tv_nsec -= 1000000000;
+ }
+}
-struct audio_codec {
- uint8_t type;
+static void timespec_diff(struct timespec *a, struct timespec *b,
+ struct timespec *res)
+{
+ res->tv_sec = a->tv_sec - b->tv_sec;
+ res->tv_nsec = a->tv_nsec - b->tv_nsec;
- int (*get_presets) (struct audio_preset *preset, size_t *len);
+ if (res->tv_nsec < 0) {
+ res->tv_sec--;
+ res->tv_nsec += 1000000000; /* 1sec */
+ }
+}
- int (*init) (struct audio_preset *preset, void **codec_data);
- int (*cleanup) (void *codec_data);
- int (*get_config) (void *codec_data,
- struct audio_input_config *config);
- ssize_t (*write_data) (void *codec_data, const void *buffer,
- size_t bytes);
-};
+static uint64_t timespec_diff_us(struct timespec *a, struct timespec *b)
+{
+ struct timespec res;
-static const struct audio_codec audio_codecs[] = {
- {
- .type = A2DP_CODEC_SBC,
+ timespec_diff(a, b, &res);
- .get_presets = sbc_get_presets,
+ return res.tv_sec * 1000000ll + res.tv_nsec / 1000ll;
+}
- .init = sbc_init,
- .cleanup = sbc_cleanup,
- .get_config = sbc_get_config,
- }
+#if defined(ANDROID)
+/*
+ * Bionic does not have clock_nanosleep() prototype in time.h even though
+ * it provides its implementation.
+ */
+extern int clock_nanosleep(clockid_t clock_id, int flags,
+ const struct timespec *request,
+ struct timespec *remain);
+#endif
+
+static struct {
+ const audio_codec_get_t get_codec;
+ bool loaded;
+} audio_codecs[] = {
+ { .get_codec = codec_aptx, .loaded = false },
+ { .get_codec = codec_sbc, .loaded = false },
};
#define NUM_CODECS (sizeof(audio_codecs) / sizeof(audio_codecs[0]))
const struct audio_codec *codec;
void *codec_data;
int fd;
+
+ struct media_packet *mp;
+ size_t mp_data_len;
+
+ uint16_t seq;
+ uint32_t samples;
+ struct timespec start;
+
+ bool resync;
};
static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
struct audio_endpoint *ep;
enum a2dp_state_t audio_state;
struct audio_input_config cfg;
+
+ uint8_t *downmix_buf;
};
struct a2dp_audio_dev {
struct a2dp_stream_out *out;
};
-static const a2dp_sbc_t sbc_presets[] = {
- {
- .frequency = SBC_SAMPLING_FREQ_44100 | SBC_SAMPLING_FREQ_48000,
- .channel_mode = SBC_CHANNEL_MODE_MONO |
- SBC_CHANNEL_MODE_DUAL_CHANNEL |
- SBC_CHANNEL_MODE_STEREO |
- SBC_CHANNEL_MODE_JOINT_STEREO,
- .subbands = SBC_SUBBANDS_4 | SBC_SUBBANDS_8,
- .allocation_method = SBC_ALLOCATION_SNR |
- SBC_ALLOCATION_LOUDNESS,
- .block_length = SBC_BLOCK_LENGTH_4 | SBC_BLOCK_LENGTH_8 |
- SBC_BLOCK_LENGTH_12 | SBC_BLOCK_LENGTH_16,
- .min_bitpool = MIN_BITPOOL,
- .max_bitpool = MAX_BITPOOL
- },
- {
- .frequency = SBC_SAMPLING_FREQ_44100,
- .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
- .subbands = SBC_SUBBANDS_8,
- .allocation_method = SBC_ALLOCATION_LOUDNESS,
- .block_length = SBC_BLOCK_LENGTH_16,
- .min_bitpool = MIN_BITPOOL,
- .max_bitpool = MAX_BITPOOL
- },
- {
- .frequency = SBC_SAMPLING_FREQ_48000,
- .channel_mode = SBC_CHANNEL_MODE_JOINT_STEREO,
- .subbands = SBC_SUBBANDS_8,
- .allocation_method = SBC_ALLOCATION_LOUDNESS,
- .block_length = SBC_BLOCK_LENGTH_16,
- .min_bitpool = MIN_BITPOOL,
- .max_bitpool = MAX_BITPOOL
- },
-};
-
-static int sbc_get_presets(struct audio_preset *preset, size_t *len)
-{
- int i;
- int count;
- size_t new_len = 0;
- uint8_t *ptr = (uint8_t *) preset;
- size_t preset_size = sizeof(*preset) + sizeof(a2dp_sbc_t);
-
- DBG("");
-
- count = sizeof(sbc_presets) / sizeof(sbc_presets[0]);
-
- for (i = 0; i < count; i++) {
- preset = (struct audio_preset *) ptr;
-
- if (new_len + preset_size > *len)
- break;
-
- preset->len = sizeof(a2dp_sbc_t);
- memcpy(preset->data, &sbc_presets[i], preset->len);
-
- new_len += preset_size;
- ptr += preset_size;
- }
-
- *len = new_len;
-
- return i;
-}
-
-static int sbc_init(struct audio_preset *preset, void **codec_data)
-{
- struct sbc_data *sbc_data;
-
- DBG("");
-
- if (preset->len != sizeof(a2dp_sbc_t)) {
- DBG("preset size mismatch");
- return AUDIO_STATUS_FAILED;
- }
-
- sbc_data = calloc(sizeof(struct sbc_data), 1);
-
- memcpy(&sbc_data->sbc, preset->data, preset->len);
-
- *codec_data = sbc_data;
-
- return AUDIO_STATUS_SUCCESS;
-}
-
-static int sbc_cleanup(void *codec_data)
-{
- DBG("");
-
- free(codec_data);
-
- return AUDIO_STATUS_SUCCESS;
-}
-
-static int sbc_get_config(void *codec_data,
- struct audio_input_config *config)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
-
- switch (sbc_data->sbc.frequency) {
- case SBC_SAMPLING_FREQ_16000:
- config->rate = 16000;
- break;
- case SBC_SAMPLING_FREQ_32000:
- config->rate = 32000;
- break;
- case SBC_SAMPLING_FREQ_44100:
- config->rate = 44100;
- break;
- case SBC_SAMPLING_FREQ_48000:
- config->rate = 48000;
- break;
- default:
- return AUDIO_STATUS_FAILED;
- }
- config->channels = sbc_data->sbc.channel_mode == SBC_CHANNEL_MODE_MONO ?
- AUDIO_CHANNEL_OUT_MONO :
- AUDIO_CHANNEL_OUT_STEREO;
- config->format = AUDIO_FORMAT_PCM_16_BIT;
-
- return AUDIO_STATUS_SUCCESS;
-}
-
-static void audio_ipc_cleanup(void)
-{
- if (audio_sk >= 0) {
- shutdown(audio_sk, SHUT_RDWR);
- audio_sk = -1;
- }
-}
-
static int audio_ipc_cmd(uint8_t service_id, uint8_t opcode, uint16_t len,
void *param, size_t *rsp_len, void *rsp, int *fd)
{
ssize_t ret;
struct msghdr msg;
struct iovec iv[2];
- struct hal_hdr cmd;
+ struct ipc_hdr cmd;
char cmsgbuf[CMSG_SPACE(sizeof(int))];
- struct hal_status s;
+ struct ipc_status s;
size_t s_len = sizeof(s);
+ pthread_mutex_lock(&sk_mutex);
+
if (audio_sk < 0) {
error("audio: Invalid cmd socket passed to audio_ipc_cmd");
goto failed;
msg.msg_iov = iv;
msg.msg_iovlen = 2;
- pthread_mutex_lock(&sk_mutex);
-
ret = sendmsg(audio_sk, &msg, 0);
if (ret < 0) {
error("audio: Sending command failed:%s", strerror(errno));
- pthread_mutex_unlock(&sk_mutex);
goto failed;
}
if (ret < 0) {
error("audio: Receiving command response failed:%s",
strerror(errno));
- pthread_mutex_unlock(&sk_mutex);
goto failed;
}
- pthread_mutex_unlock(&sk_mutex);
-
if (ret < (ssize_t) sizeof(cmd)) {
error("audio: Too small response received(%zd bytes)", ret);
goto failed;
}
if (cmd.opcode == AUDIO_OP_STATUS) {
- struct hal_status *s = rsp;
+ struct ipc_status *s = rsp;
if (sizeof(*s) != cmd.len) {
error("audio: Invalid status length");
goto failed;
}
+ pthread_mutex_unlock(&sk_mutex);
+
return s->code;
}
+ pthread_mutex_unlock(&sk_mutex);
+
/* Receive auxiliary data in msg */
if (fd) {
struct cmsghdr *cmsg;
break;
}
}
+
+ if (*fd < 0)
+ goto failed;
}
- if (rsp_len)
- *rsp_len = cmd.len;
+ *rsp_len = cmd.len;
return AUDIO_STATUS_SUCCESS;
failed:
/* Some serious issue happen on IPC - recover */
shutdown(audio_sk, SHUT_RDWR);
- audio_sk = -1;
+ pthread_mutex_unlock(&sk_mutex);
+
return AUDIO_STATUS_FAILED;
}
return result;
}
-static int ipc_open_stream_cmd(uint8_t endpoint_id,
- struct audio_preset **caps)
+static int ipc_open_stream_cmd(uint8_t *endpoint_id, uint16_t *mtu, int *fd,
+ struct audio_preset **caps)
{
char buf[BLUEZ_AUDIO_MTU];
struct audio_cmd_open_stream cmd;
if (!caps)
return AUDIO_STATUS_FAILED;
- cmd.id = endpoint_id;
+ cmd.id = *endpoint_id;
result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_OPEN_STREAM,
- sizeof(cmd), &cmd, &rsp_len, rsp, NULL);
-
+ sizeof(cmd), &cmd, &rsp_len, rsp, fd);
if (result == AUDIO_STATUS_SUCCESS) {
size_t buf_len = sizeof(struct audio_preset) +
rsp->preset[0].len;
+ *endpoint_id = rsp->id;
+ *mtu = rsp->mtu;
*caps = malloc(buf_len);
memcpy(*caps, &rsp->preset, buf_len);
} else {
cmd.id = endpoint_id;
result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_CLOSE_STREAM,
- sizeof(cmd), &cmd, NULL, NULL, NULL);
+ sizeof(cmd), &cmd, NULL, NULL, NULL);
return result;
}
cmd.id = endpoint_id;
result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_RESUME_STREAM,
- sizeof(cmd), &cmd, NULL, NULL, NULL);
+ sizeof(cmd), &cmd, NULL, NULL, NULL);
return result;
}
cmd.id = endpoint_id;
result = audio_ipc_cmd(AUDIO_SERVICE_ID, AUDIO_OP_SUSPEND_STREAM,
- sizeof(cmd), &cmd, NULL, NULL, NULL);
+ sizeof(cmd), &cmd, NULL, NULL, NULL);
return result;
}
+struct register_state {
+ struct audio_endpoint *ep;
+ bool error;
+};
+
+static void register_endpoint(const struct audio_codec *codec,
+ struct register_state *state)
+{
+ struct audio_endpoint *ep = state->ep;
+
+ /* don't even try to register more endpoints if one failed */
+ if (state->error)
+ return;
+
+ ep->id = ipc_open_cmd(codec);
+
+ if (!ep->id) {
+ state->error = true;
+ error("Failed to register endpoint");
+ return;
+ }
+
+ ep->codec = codec;
+ ep->codec_data = NULL;
+ ep->fd = -1;
+
+ state->ep++;
+}
+
static int register_endpoints(void)
{
- struct audio_endpoint *ep = &audio_endpoints[0];
- size_t i;
+ struct register_state state;
+ unsigned int i;
- for (i = 0; i < NUM_CODECS; i++, ep++) {
- const struct audio_codec *codec = &audio_codecs[i];
+ state.ep = &audio_endpoints[0];
+ state.error = false;
- ep->id = ipc_open_cmd(codec);
+ for (i = 0; i < NUM_CODECS; i++) {
+ const struct audio_codec *codec = audio_codecs[i].get_codec();
- if (!ep->id)
- return AUDIO_STATUS_FAILED;
+ if (!audio_codecs[i].loaded)
+ continue;
- ep->codec = codec;
- ep->codec_data = NULL;
- ep->fd = -1;
+ register_endpoint(codec, &state);
}
- return AUDIO_STATUS_SUCCESS;
+ return state.error ? AUDIO_STATUS_FAILED : AUDIO_STATUS_SUCCESS;
}
static void unregister_endpoints(void)
}
}
+static bool open_endpoint(struct audio_endpoint **epp,
+ struct audio_input_config *cfg)
+{
+ struct audio_preset *preset;
+ struct audio_endpoint *ep = *epp;
+ const struct audio_codec *codec;
+ uint16_t mtu;
+ uint16_t payload_len;
+ int fd;
+ size_t i;
+ uint8_t ep_id = 0;
+
+ if (ep)
+ ep_id = ep->id;
+
+ if (ipc_open_stream_cmd(&ep_id, &mtu, &fd, &preset) !=
+ AUDIO_STATUS_SUCCESS)
+ return false;
+
+ DBG("ep_id=%d mtu=%u", ep_id, mtu);
+
+ for (i = 0; i < MAX_AUDIO_ENDPOINTS; i++)
+ if (audio_endpoints[i].id == ep_id) {
+ ep = &audio_endpoints[i];
+ break;
+ }
+
+ if (!ep) {
+ error("Cound not find opened endpoint");
+ goto failed;
+ }
+
+ *epp = ep;
+
+ payload_len = mtu;
+ if (ep->codec->use_rtp)
+ payload_len -= sizeof(struct rtp_header);
+
+ ep->fd = fd;
+
+ codec = ep->codec;
+ codec->init(preset, payload_len, &ep->codec_data);
+ codec->get_config(ep->codec_data, cfg);
+
+ ep->mp = calloc(mtu, 1);
+ if (!ep->mp)
+ goto failed;
+
+ if (ep->codec->use_rtp) {
+ struct media_packet_rtp *mp_rtp =
+ (struct media_packet_rtp *) ep->mp;
+ mp_rtp->hdr.v = 2;
+ mp_rtp->hdr.pt = 0x60;
+ mp_rtp->hdr.ssrc = htonl(1);
+ }
+
+ ep->mp_data_len = payload_len;
+
+ free(preset);
+
+ return true;
+
+failed:
+ close(fd);
+ free(preset);
+
+ return false;
+}
+
+static void close_endpoint(struct audio_endpoint *ep)
+{
+ ipc_close_stream_cmd(ep->id);
+ if (ep->fd >= 0) {
+ close(ep->fd);
+ ep->fd = -1;
+ }
+
+ free(ep->mp);
+
+ ep->codec->cleanup(ep->codec_data);
+ ep->codec_data = NULL;
+}
+
+static bool resume_endpoint(struct audio_endpoint *ep)
+{
+ if (ipc_resume_stream_cmd(ep->id) != AUDIO_STATUS_SUCCESS)
+ return false;
+
+ ep->samples = 0;
+ ep->resync = false;
+
+ ep->codec->update_qos(ep->codec_data, QOS_POLICY_DEFAULT);
+
+ return true;
+}
+
+static void downmix_to_mono(struct a2dp_stream_out *out, const uint8_t *buffer,
+ size_t bytes)
+{
+ const int16_t *input = (const void *) buffer;
+ int16_t *output = (void *) out->downmix_buf;
+ size_t i, frames;
+
+ /* PCM 16bit stereo */
+ frames = bytes / (2 * sizeof(int16_t));
+
+ for (i = 0; i < frames; i++) {
+ int16_t l = get_le16(&input[i * 2]);
+ int16_t r = get_le16(&input[i * 2 + 1]);
+
+ put_le16((l + r) / 2, &output[i]);
+ }
+}
+
+static bool wait_for_endpoint(struct audio_endpoint *ep, bool *writable)
+{
+ int ret;
+
+ while (true) {
+ struct pollfd pollfd;
+
+ pollfd.fd = ep->fd;
+ pollfd.events = POLLOUT;
+ pollfd.revents = 0;
+
+ ret = poll(&pollfd, 1, 500);
+
+ if (ret >= 0) {
+ *writable = !!(pollfd.revents & POLLOUT);
+ break;
+ }
+
+ if (errno != EINTR) {
+ ret = errno;
+ error("poll failed (%d)", ret);
+ return false;
+ }
+ }
+
+ return true;
+}
+
+static bool write_to_endpoint(struct audio_endpoint *ep, size_t bytes)
+{
+ struct media_packet *mp = (struct media_packet *) ep->mp;
+ int ret;
+
+ while (true) {
+ ret = write(ep->fd, mp, bytes);
+
+ if (ret >= 0)
+ break;
+
+ /*
+ * this should not happen so let's issue warning, but do not
+ * fail, we can try to write next packet
+ */
+ if (errno == EAGAIN) {
+ ret = errno;
+ warn("write failed (%d)", ret);
+ break;
+ }
+
+ if (errno != EINTR) {
+ ret = errno;
+ error("write failed (%d)", ret);
+ return false;
+ }
+ }
+
+ return true;
+}
+
+static bool write_data(struct a2dp_stream_out *out, const void *buffer,
+ size_t bytes)
+{
+ struct audio_endpoint *ep = out->ep;
+ struct media_packet *mp = (struct media_packet *) ep->mp;
+ struct media_packet_rtp *mp_rtp = (struct media_packet_rtp *) ep->mp;
+ size_t free_space = ep->mp_data_len;
+ size_t consumed = 0;
+
+ while (consumed < bytes) {
+ size_t written = 0;
+ ssize_t read;
+ uint32_t samples;
+ int ret;
+ struct timespec current;
+ uint64_t audio_sent, audio_passed;
+ bool do_write = false;
+
+ /*
+ * prepare media packet in advance so we don't waste time after
+ * wakeup
+ */
+ if (ep->codec->use_rtp) {
+ mp_rtp->hdr.sequence_number = htons(ep->seq++);
+ mp_rtp->hdr.timestamp = htonl(ep->samples);
+ }
+ read = ep->codec->encode_mediapacket(ep->codec_data,
+ buffer + consumed,
+ bytes - consumed, mp,
+ free_space, &written);
+
+ /*
+ * not much we can do here, let's just ignore remaining
+ * data and continue
+ */
+ if (read <= 0)
+ return true;
+
+ /* calculate where are we and where we should be */
+ clock_gettime(CLOCK_MONOTONIC, ¤t);
+ if (!ep->samples)
+ memcpy(&ep->start, ¤t, sizeof(ep->start));
+ audio_sent = ep->samples * 1000000ll / out->cfg.rate;
+ audio_passed = timespec_diff_us(¤t, &ep->start);
+
+ /*
+ * if we're ahead of stream then wait for next write point,
+ * if we're lagging more than 100ms then stop writing and just
+ * skip data until we're back in sync
+ */
+ if (audio_sent > audio_passed) {
+ struct timespec anchor;
+
+ ep->resync = false;
+
+ timespec_add(&ep->start, audio_sent, &anchor);
+
+ while (true) {
+ ret = clock_nanosleep(CLOCK_MONOTONIC,
+ TIMER_ABSTIME, &anchor,
+ NULL);
+
+ if (!ret)
+ break;
+
+ if (ret != EINTR) {
+ error("clock_nanosleep failed (%d)",
+ ret);
+ return false;
+ }
+ }
+ } else if (!ep->resync) {
+ uint64_t diff = audio_passed - audio_sent;
+
+ if (diff > MAX_DELAY) {
+ warn("lag is %jums, resyncing", diff / 1000);
+
+ ep->codec->update_qos(ep->codec_data,
+ QOS_POLICY_DECREASE);
+ ep->resync = true;
+ }
+ }
+
+ /* we send data only in case codec encoded some data, i.e. some
+ * codecs do internal buffering and output data only if full
+ * frame can be encoded
+ * in resync mode we'll just drop mediapackets
+ */
+ if (written > 0 && !ep->resync) {
+ /* wait some time for socket to be ready for write,
+ * but we'll just skip writing data if timeout occurs
+ */
+ if (!wait_for_endpoint(ep, &do_write))
+ return false;
+
+ if (do_write) {
+ if (ep->codec->use_rtp)
+ written += sizeof(struct rtp_header);
+
+ if (!write_to_endpoint(ep, written))
+ return false;
+ }
+ }
+
+ /*
+ * AudioFlinger provides 16bit PCM, so sample size is 2 bytes
+ * multiplied by number of channels. Number of channels is
+ * simply number of bits set in channels mask.
+ */
+ samples = read / (2 * popcount(out->cfg.channels));
+ ep->samples += samples;
+ consumed += read;
+ }
+
+ return true;
+}
+
static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
size_t bytes)
{
struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
+ const void *in_buf = buffer;
+ size_t in_len = bytes;
+
+ /* just return in case we're closing */
+ if (out->audio_state == AUDIO_A2DP_STATE_NONE)
+ return -1;
/* We can auto-start only from standby */
if (out->audio_state == AUDIO_A2DP_STATE_STANDBY) {
DBG("stream in standby, auto-start");
- if (ipc_resume_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
+ if (!resume_endpoint(out->ep))
return -1;
out->audio_state = AUDIO_A2DP_STATE_STARTED;
}
if (out->audio_state != AUDIO_A2DP_STATE_STARTED) {
- DBG("stream not started");
+ error("audio: stream not started");
return -1;
}
- /* TODO: encode data using codec */
+ if (out->ep->fd < 0) {
+ error("audio: no transport socket");
+ return -1;
+ }
+
+ /*
+ * currently Android audioflinger is not able to provide mono stream on
+ * A2DP output so down mixing needs to be done in hal-audio plugin.
+ *
+ * for reference see
+ * AudioFlinger::PlaybackThread::readOutputParameters()
+ * frameworks/av/services/audioflinger/Threads.cpp:1631
+ */
+ if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
+ if (!out->downmix_buf) {
+ error("audio: downmix buffer not initialized");
+ return -1;
+ }
+
+ downmix_to_mono(out, buffer, bytes);
+
+ in_buf = out->downmix_buf;
+ in_len = bytes / 2;
+ }
+
+ if (!write_data(out, in_buf, in_len))
+ return -1;
return bytes;
}
DBG("");
if (rate != out->cfg.rate) {
- DBG("cannot set sample rate to %d", rate);
+ warn("audio: cannot set sample rate to %d", rate);
return -1;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
DBG("");
- return -ENOSYS;
+
+ /*
+ * We should return proper buffer size calculated by codec (so each
+ * input buffer is encoded into single media packed) but this does not
+ * work well with AudioFlinger and causes problems. For this reason we
+ * use magic value here and out_write code takes care of splitting
+ * input buffer into multiple media packets.
+ */
+ return FIXED_BUFFER_SIZE;
}
static uint32_t out_get_channels(const struct audio_stream *stream)
{
- struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
-
DBG("");
- return out->cfg.channels;
+ /*
+ * AudioFlinger can only provide stereo stream, so we return it here and
+ * later we'll downmix this to mono in case codec requires it
+ */
+
+ return AUDIO_CHANNEL_OUT_STEREO;
}
static audio_format_t out_get_format(const struct audio_stream *stream)
DBG("%s", kvpairs);
str = strdup(kvpairs);
+ if (!str)
+ return -ENOMEM;
+
kvpair = strtok_r(str, ";", &saveptr);
for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
static uint32_t out_get_latency(const struct audio_stream_out *stream)
{
+ struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
+ struct audio_endpoint *ep = out->ep;
+ size_t pkt_duration;
+
DBG("");
- return -ENOSYS;
+
+ pkt_duration = ep->codec->get_mediapacket_duration(ep->codec_data);
+
+ return FIXED_A2DP_PLAYBACK_LATENCY_MS + pkt_duration / 1000;
}
static int out_set_volume(struct audio_stream_out *stream, float left,
{
struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
struct a2dp_stream_out *out;
- struct audio_preset *preset;
- const struct audio_codec *codec;
out = calloc(1, sizeof(struct a2dp_stream_out));
if (!out)
out->stream.write = out_write;
out->stream.get_render_position = out_get_render_position;
- /* TODO: for now we always use endpoint 0 */
- out->ep = &audio_endpoints[0];
+ /* We want to autoselect opened endpoint */
+ out->ep = NULL;
- if (ipc_open_stream_cmd(out->ep->id, &preset) != AUDIO_STATUS_SUCCESS)
+ if (!open_endpoint(&out->ep, &out->cfg))
goto fail;
- if (!preset)
- goto fail;
-
- codec = out->ep->codec;
-
- codec->init(preset, &out->ep->codec_data);
- codec->get_config(out->ep->codec_data, &out->cfg);
-
DBG("rate=%d channels=%d format=%d", out->cfg.rate,
- out->cfg.channels, out->cfg.format);
+ out->cfg.channels, out->cfg.format);
- free(preset);
+ if (out->cfg.channels == AUDIO_CHANNEL_OUT_MONO) {
+ out->downmix_buf = malloc(FIXED_BUFFER_SIZE / 2);
+ if (!out->downmix_buf)
+ goto fail;
+ }
*stream_out = &out->stream;
a2dp_dev->out = out;
return 0;
fail:
+ error("audio: cannot open output stream");
free(out);
*stream_out = NULL;
return -EIO;
struct audio_stream_out *stream)
{
struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
- struct audio_endpoint *ep = a2dp_dev->out->ep;
+ struct a2dp_stream_out *out = (struct a2dp_stream_out *) stream;
DBG("");
- ipc_close_stream_cmd(ep->id);
+ close_endpoint(a2dp_dev->out->ep);
- ep->codec->cleanup(ep->codec_data);
- ep->codec_data = NULL;
+ free(out->downmix_buf);
free(stream);
a2dp_dev->out = NULL;
static int audio_set_parameters(struct audio_hw_device *dev,
const char *kvpairs)
{
+ struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
+ struct a2dp_stream_out *out = a2dp_dev->out;
+
DBG("");
- return -ENOSYS;
+
+ if (!out)
+ return 0;
+
+ return out->stream.common.set_parameters((struct audio_stream *) out,
+ kvpairs);
}
static char *audio_get_parameters(const struct audio_hw_device *dev,
static int audio_close(hw_device_t *device)
{
struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
+ unsigned int i;
DBG("");
- pthread_mutex_lock(&close_mutex);
- audio_ipc_cleanup();
- close_thread = true;
- pthread_mutex_unlock(&close_mutex);
+ unregister_endpoints();
+
+ for (i = 0; i < NUM_CODECS; i++) {
+ const struct audio_codec *codec = audio_codecs[i].get_codec();
+
+ if (!audio_codecs[i].loaded)
+ continue;
+
+ if (codec->unload)
+ codec->unload();
+
+ audio_codecs[i].loaded = false;
+ }
+
+ shutdown(listen_sk, SHUT_RDWR);
+ shutdown(audio_sk, SHUT_RDWR);
pthread_join(ipc_th, NULL);
{
bool done = false;
struct pollfd pfd;
+ int sk;
DBG("");
while (!done) {
DBG("Waiting for connection ...");
- audio_sk = accept(listen_sk, NULL, NULL);
- if (audio_sk < 0) {
+
+ sk = accept(listen_sk, NULL, NULL);
+ if (sk < 0) {
int err = errno;
- error("audio: Failed to accept socket: %d (%s)", err,
- strerror(err));
- continue;
+
+ if (err == EINTR)
+ continue;
+
+ if (err != ECONNABORTED && err != EINVAL)
+ error("audio: Failed to accept socket: %d (%s)",
+ err, strerror(err));
+
+ break;
}
+ pthread_mutex_lock(&sk_mutex);
+ audio_sk = sk;
+ pthread_mutex_unlock(&sk_mutex);
+
DBG("Audio IPC: Connected");
if (register_endpoints() != AUDIO_STATUS_SUCCESS) {
unregister_endpoints();
+ pthread_mutex_lock(&sk_mutex);
shutdown(audio_sk, SHUT_RDWR);
+ close(audio_sk);
+ audio_sk = -1;
+ pthread_mutex_unlock(&sk_mutex);
+
continue;
}
/* Check if socket is still alive. Empty while loop.*/
while (poll(&pfd, 1, -1) < 0 && errno == EINTR);
- if (pfd.revents & (POLLHUP | POLLERR | POLLNVAL)) {
- info("Audio HAL: Socket closed");
- audio_sk = -1;
- }
+ info("Audio HAL: Socket closed");
- /*Check if audio_dev is closed */
- pthread_mutex_lock(&close_mutex);
- done = close_thread;
- close_thread = false;
- pthread_mutex_unlock(&close_mutex);
+ pthread_mutex_lock(&sk_mutex);
+ close(audio_sk);
+ audio_sk = -1;
+ pthread_mutex_unlock(&sk_mutex);
}
- unregister_endpoints();
+ /* audio_sk is closed at this point, just cleanup endpoints states */
+ memset(audio_endpoints, 0, sizeof(audio_endpoints));
info("Closing Audio IPC thread");
return NULL;
sk = socket(PF_LOCAL, SOCK_SEQPACKET, 0);
if (sk < 0) {
- err = errno;
- error("audio: Failed to create socket: %d (%s)", err,
- strerror(err));
+ err = -errno;
+ error("audio: Failed to create socket: %d (%s)", -err,
+ strerror(-err));
return err;
}
sizeof(BLUEZ_AUDIO_SK_PATH));
if (bind(sk, (struct sockaddr *) &addr, sizeof(addr)) < 0) {
- err = errno;
- error("audio: Failed to bind socket: %d (%s)", err,
- strerror(err));
+ err = -errno;
+ error("audio: Failed to bind socket: %d (%s)", -err,
+ strerror(-err));
goto failed;
}
if (listen(sk, 1) < 0) {
- err = errno;
- error("audio: Failed to listen on the socket: %d (%s)", err,
- strerror(err));
+ err = -errno;
+ error("audio: Failed to listen on the socket: %d (%s)", -err,
+ strerror(-err));
goto failed;
}
err = -err;
ipc_th = 0;
error("audio: Failed to start Audio IPC thread: %d (%s)",
- err, strerror(err));
+ -err, strerror(-err));
goto failed;
}
hw_device_t **device)
{
struct a2dp_audio_dev *a2dp_dev;
+ size_t i;
int err;
DBG("");
}
err = audio_ipc_init();
- if (err)
- return -err;
+ if (err < 0)
+ return err;
a2dp_dev = calloc(1, sizeof(struct a2dp_audio_dev));
if (!a2dp_dev)
a2dp_dev->dev.close_input_stream = audio_close_input_stream;
a2dp_dev->dev.dump = audio_dump;
- /* Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
+ for (i = 0; i < NUM_CODECS; i++) {
+ const struct audio_codec *codec = audio_codecs[i].get_codec();
+
+ if (codec->load && !codec->load())
+ continue;
+
+ audio_codecs[i].loaded = true;
+ }
+
+ /*
+ * Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
* This results from the structure of following structs:a2dp_audio_dev,
- * audio_hw_device. We will rely on this later in the code.*/
+ * audio_hw_device. We will rely on this later in the code.
+ */
*device = &a2dp_dev->dev.common;
return 0;
struct audio_module HAL_MODULE_INFO_SYM = {
.common = {
- .tag = HARDWARE_MODULE_TAG,
- .version_major = 1,
- .version_minor = 0,
- .id = AUDIO_HARDWARE_MODULE_ID,
- .name = "A2DP Bluez HW HAL",
- .author = "Intel Corporation",
- .methods = &hal_module_methods,
+ .tag = HARDWARE_MODULE_TAG,
+ .version_major = 1,
+ .version_minor = 0,
+ .id = AUDIO_HARDWARE_MODULE_ID,
+ .name = "A2DP Bluez HW HAL",
+ .author = "Intel Corporation",
+ .methods = &hal_module_methods,
},
};
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