#include <hardware/audio.h>
#include <hardware/hardware.h>
-#include <sbc/sbc.h>
-
#include "audio-msg.h"
#include "ipc-common.h"
#include "hal-log.h"
#include "hal-msg.h"
-#include "../profiles/audio/a2dp-codecs.h"
-#include "../src/shared/util.h"
+#include "hal-audio.h"
+#include "hal-utils.h"
+#include "hal.h"
#define FIXED_A2DP_PLAYBACK_LATENCY_MS 25
#define FIXED_BUFFER_SIZE (20 * 512)
-#define MAX_FRAMES_IN_PAYLOAD 15
+#define MAX_DELAY 100000 /* 100ms */
static const uint8_t a2dp_src_uuid[] = {
0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
static pthread_t ipc_th = 0;
static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
-#if __BYTE_ORDER == __LITTLE_ENDIAN
-
-struct rtp_header {
- unsigned cc:4;
- unsigned x:1;
- unsigned p:1;
- unsigned v:2;
-
- unsigned pt:7;
- unsigned m:1;
-
- uint16_t sequence_number;
- uint32_t timestamp;
- uint32_t ssrc;
- uint32_t csrc[0];
-} __attribute__ ((packed));
-
-struct rtp_payload {
- unsigned frame_count:4;
- unsigned rfa0:1;
- unsigned is_last_fragment:1;
- unsigned is_first_fragment:1;
- unsigned is_fragmented:1;
-} __attribute__ ((packed));
-
-#elif __BYTE_ORDER == __BIG_ENDIAN
-
-struct rtp_header {
- unsigned v:2;
- unsigned p:1;
- unsigned x:1;
- unsigned cc:4;
-
- unsigned m:1;
- unsigned pt:7;
-
- uint16_t sequence_number;
- uint32_t timestamp;
- uint32_t ssrc;
- uint32_t csrc[0];
-} __attribute__ ((packed));
-
-struct rtp_payload {
- unsigned is_fragmented:1;
- unsigned is_first_fragment:1;
- unsigned is_last_fragment:1;
- unsigned rfa0:1;
- unsigned frame_count:4;
-} __attribute__ ((packed));
-
-#else
-#error "Unknown byte order"
-#endif
-
-struct media_packet {
- struct rtp_header hdr;
- struct rtp_payload payload;
- uint8_t data[0];
-};
-
-struct audio_input_config {
- uint32_t rate;
- uint32_t channels;
- audio_format_t format;
-};
-
-struct sbc_data {
- a2dp_sbc_t sbc;
-
- sbc_t enc;
-
- size_t in_frame_len;
- size_t in_buf_size;
-
- size_t out_buf_size;
- uint8_t *out_buf;
-
- unsigned frame_duration;
- unsigned frames_per_packet;
-
- struct timespec start;
- unsigned frames_sent;
- uint32_t timestamp;
+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;
- uint16_t seq;
-};
+ if (res->tv_nsec >= 1000000000) {
+ res->tv_sec++;
+ res->tv_nsec -= 1000000000;
+ }
+}
-static inline void timespec_diff(struct timespec *a, struct timespec *b,
+static void timespec_diff(struct timespec *a, struct timespec *b,
struct timespec *res)
{
res->tv_sec = a->tv_sec - b->tv_sec;
}
}
-static int sbc_get_presets(struct audio_preset *preset, size_t *len);
-static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
- void **codec_data);
-static int sbc_cleanup(void *codec_data);
-static int sbc_get_config(void *codec_data, struct audio_input_config *config);
-static size_t sbc_get_buffer_size(void *codec_data);
-static size_t sbc_get_mediapacket_duration(void *codec_data);
-static void sbc_resume(void *codec_data);
-static ssize_t sbc_write_data(void *codec_data, const void *buffer,
- size_t bytes, int fd);
-
-struct audio_codec {
- uint8_t type;
-
- int (*get_presets) (struct audio_preset *preset, size_t *len);
-
- int (*init) (struct audio_preset *preset, uint16_t mtu,
- void **codec_data);
- int (*cleanup) (void *codec_data);
- int (*get_config) (void *codec_data,
- struct audio_input_config *config);
- size_t (*get_buffer_size) (void *codec_data);
- size_t (*get_mediapacket_duration) (void *codec_data);
- void (*resume) (void *codec_data);
- ssize_t (*write_data) (void *codec_data, const void *buffer,
- size_t bytes, int fd);
-};
+static uint64_t timespec_diff_us(struct timespec *a, struct timespec *b)
+{
+ struct timespec res;
+
+ timespec_diff(a, b, &res);
-static const struct audio_codec audio_codecs[] = {
- {
- .type = A2DP_CODEC_SBC,
+ return res.tv_sec * 1000000ll + res.tv_nsec / 1000ll;
+}
- .get_presets = sbc_get_presets,
+#if ANDROID_VERSION < PLATFORM_VER(6, 0, 0)
+/*
+ * 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
- .init = sbc_codec_init,
- .cleanup = sbc_cleanup,
- .get_config = sbc_get_config,
- .get_buffer_size = sbc_get_buffer_size,
- .get_mediapacket_duration = sbc_get_mediapacket_duration,
- .resume = sbc_resume,
- .write_data = sbc_write_data,
- }
+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 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);
-
- 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_freq2int(uint8_t freq)
-{
- switch (freq) {
- case SBC_SAMPLING_FREQ_16000:
- return 16000;
- case SBC_SAMPLING_FREQ_32000:
- return 32000;
- case SBC_SAMPLING_FREQ_44100:
- return 44100;
- case SBC_SAMPLING_FREQ_48000:
- return 48000;
- default:
- return 0;
- }
-}
-
-static const char *sbc_mode2str(uint8_t mode)
-{
- switch (mode) {
- case SBC_CHANNEL_MODE_MONO:
- return "Mono";
- case SBC_CHANNEL_MODE_DUAL_CHANNEL:
- return "DualChannel";
- case SBC_CHANNEL_MODE_STEREO:
- return "Stereo";
- case SBC_CHANNEL_MODE_JOINT_STEREO:
- return "JointStereo";
- default:
- return "(unknown)";
- }
-}
-
-static int sbc_blocks2int(uint8_t blocks)
-{
- switch (blocks) {
- case SBC_BLOCK_LENGTH_4:
- return 4;
- case SBC_BLOCK_LENGTH_8:
- return 8;
- case SBC_BLOCK_LENGTH_12:
- return 12;
- case SBC_BLOCK_LENGTH_16:
- return 16;
- default:
- return 0;
- }
-}
-
-static int sbc_subbands2int(uint8_t subbands)
-{
- switch (subbands) {
- case SBC_SUBBANDS_4:
- return 4;
- case SBC_SUBBANDS_8:
- return 8;
- default:
- return 0;
- }
-}
-
-static const char *sbc_allocation2str(uint8_t allocation)
-{
- switch (allocation) {
- case SBC_ALLOCATION_SNR:
- return "SNR";
- case SBC_ALLOCATION_LOUDNESS:
- return "Loudness";
- default:
- return "(unknown)";
- }
-}
-
-static void sbc_init_encoder(struct sbc_data *sbc_data)
-{
- a2dp_sbc_t *in = &sbc_data->sbc;
- sbc_t *out = &sbc_data->enc;
-
- sbc_init_a2dp(out, 0L, in, sizeof(*in));
-
- out->endian = SBC_LE;
- out->bitpool = in->max_bitpool;
-
- DBG("frequency=%d channel_mode=%s block_length=%d subbands=%d "
- "allocation=%s bitpool=%d-%d",
- sbc_freq2int(in->frequency),
- sbc_mode2str(in->channel_mode),
- sbc_blocks2int(in->block_length),
- sbc_subbands2int(in->subbands),
- sbc_allocation2str(in->allocation_method),
- in->min_bitpool, in->max_bitpool);
-}
-
-static int sbc_codec_init(struct audio_preset *preset, uint16_t mtu,
- void **codec_data)
-{
- struct sbc_data *sbc_data;
- size_t hdr_len = sizeof(struct media_packet);
- size_t in_frame_len;
- size_t out_frame_len;
- size_t num_frames;
-
- if (preset->len != sizeof(a2dp_sbc_t)) {
- error("SBC: preset size mismatch");
- return AUDIO_STATUS_FAILED;
- }
-
- sbc_data = calloc(sizeof(struct sbc_data), 1);
- if (!sbc_data)
- return AUDIO_STATUS_FAILED;
-
- memcpy(&sbc_data->sbc, preset->data, preset->len);
-
- sbc_init_encoder(sbc_data);
-
- in_frame_len = sbc_get_codesize(&sbc_data->enc);
- out_frame_len = sbc_get_frame_length(&sbc_data->enc);
- num_frames = (mtu - hdr_len) / out_frame_len;
-
- sbc_data->in_frame_len = in_frame_len;
- sbc_data->in_buf_size = num_frames * in_frame_len;
-
- sbc_data->out_buf_size = hdr_len + num_frames * out_frame_len;
- sbc_data->out_buf = calloc(1, sbc_data->out_buf_size);
-
- sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
- sbc_data->frames_per_packet = num_frames;
-
- DBG("mtu=%u in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
- mtu, in_frame_len, out_frame_len, num_frames);
-
- *codec_data = sbc_data;
-
- return AUDIO_STATUS_SUCCESS;
-}
-
-static int sbc_cleanup(void *codec_data)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
-
- sbc_finish(&sbc_data->enc);
- free(sbc_data->out_buf);
- 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 size_t sbc_get_buffer_size(void *codec_data)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
-
- return sbc_data->in_buf_size;
-}
-
-static size_t sbc_get_mediapacket_duration(void *codec_data)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
-
- return sbc_data->frame_duration * sbc_data->frames_per_packet;
-}
-
-static void sbc_resume(void *codec_data)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
-
- DBG("");
-
- clock_gettime(CLOCK_MONOTONIC, &sbc_data->start);
-
- sbc_data->frames_sent = 0;
- sbc_data->timestamp = 0;
-}
-
-static int write_media_packet(int fd, struct sbc_data *sbc_data,
- struct media_packet *mp, size_t data_len)
-{
- struct timespec cur;
- struct timespec diff;
- unsigned expected_frames;
- int ret;
-
- while (true) {
- ret = write(fd, mp, sizeof(*mp) + data_len);
- if (ret >= 0)
- break;
-
- if (errno != EINTR)
- return -errno;
- }
-
- sbc_data->frames_sent += mp->payload.frame_count;
-
- clock_gettime(CLOCK_MONOTONIC, &cur);
- timespec_diff(&cur, &sbc_data->start, &diff);
- expected_frames = (diff.tv_sec * 1000000 + diff.tv_nsec / 1000) /
- sbc_data->frame_duration;
-
- /* AudioFlinger does not seem to provide any *working*
- * API to provide data in some interval and will just
- * send another buffer as soon as we process current
- * one. To prevent overflowing L2CAP socket, we need to
- * introduce some artificial delay here base on how many
- * audio frames were sent so far, i.e. if we're not
- * lagging behind audio stream, we can sleep for
- * duration of single media packet.
- */
- if (sbc_data->frames_sent >= expected_frames)
- usleep(sbc_data->frame_duration *
- mp->payload.frame_count);
-
- return ret;
-}
-
-static ssize_t sbc_write_data(void *codec_data, const void *buffer,
- size_t bytes, int fd)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
- size_t consumed = 0;
- size_t encoded = 0;
- struct media_packet *mp = (struct media_packet *) sbc_data->out_buf;
- size_t free_space = sbc_data->out_buf_size - sizeof(*mp);
- int ret;
- ssize_t bytes_read;
-
- mp->hdr.v = 2;
- mp->hdr.pt = 1;
- mp->hdr.ssrc = htonl(1);
- mp->hdr.timestamp = htonl(sbc_data->timestamp);
- mp->payload.frame_count = 0;
-
- while (bytes - consumed >= sbc_data->in_frame_len) {
- ssize_t written = 0;
-
- bytes_read = sbc_encode(&sbc_data->enc, buffer + consumed,
- sbc_data->in_frame_len,
- mp->data + encoded, free_space,
- &written);
-
- if (bytes_read < 0) {
- error("SBC: failed to encode block (%zd)", bytes_read);
- break;
- }
-
- mp->payload.frame_count++;
-
- consumed += bytes_read;
- encoded += written;
- free_space -= written;
-
- /* AudioFlinger provides PCM 16bit stereo only, thus sample size
- * is always 4 bytes
- */
- sbc_data->timestamp += (bytes_read / 4);
-
- /* write data if we either filled media packed or encoded all
- * input data
- */
- if (mp->payload.frame_count == sbc_data->frames_per_packet ||
- bytes == consumed ||
- mp->payload.frame_count ==
- MAX_FRAMES_IN_PAYLOAD) {
- mp->hdr.sequence_number = htons(sbc_data->seq++);
-
- ret = write_media_packet(fd, sbc_data, mp, encoded);
- if (ret < 0)
- return ret;
-
- encoded = 0;
- free_space = sbc_data->out_buf_size - sizeof(*mp);
- mp->hdr.timestamp = htonl(sbc_data->timestamp);
- mp->payload.frame_count = 0;
- }
- }
-
- if (consumed != bytes) {
- /* we should encode all input data
- * if we did not, something went wrong but we can't really
- * handle this so this is just sanity check
- */
- error("SBC: failed to encode complete input buffer");
- }
-
- /* we always assume that all data was processed and sent */
- return bytes;
-}
-
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)
{
goto failed;
}
- if (rsp_len)
- *rsp_len = cmd.len;
+ *rsp_len = cmd.len;
return AUDIO_STATUS_SUCCESS;
return result;
}
-static int ipc_open_stream_cmd(uint8_t endpoint_id, uint16_t *mtu, int *fd,
+static int ipc_open_stream_cmd(uint8_t *endpoint_id, uint16_t *mtu, int *fd,
struct audio_preset **caps)
{
char buf[BLUEZ_AUDIO_MTU];
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, 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);
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 int set_blocking(int fd)
-{
- int flags;
-
- flags = fcntl(fd, F_GETFL, 0);
- if (flags < 0) {
- int err = -errno;
- error("fcntl(F_GETFL): %s (%d)", strerror(-err), -err);
- return err;
- }
-
- if (fcntl(fd, F_SETFL, flags & ~O_NONBLOCK) < 0) {
- int err = -errno;
- error("fcntl(F_SETFL): %s (%d)", strerror(-err), -err);
- return err;
- }
-
- return 0;
-}
-
-static bool open_endpoint(struct audio_endpoint *ep,
+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 (ipc_open_stream_cmd(ep->id, &mtu, &fd, &preset) !=
+ if (ep)
+ ep_id = ep->id;
+
+ if (ipc_open_stream_cmd(&ep_id, &mtu, &fd, &preset) !=
AUDIO_STATUS_SUCCESS)
return false;
- if (set_blocking(fd) < 0)
+ 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, mtu, &ep->codec_data);
+ 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;
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;
+ 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;
+}
- for (i = 0; i < bytes / 2; i++) {
- int16_t l = le16_to_cpu(get_unaligned(&input[i * 2]));
- int16_t r = le16_to_cpu(get_unaligned(&input[i * 2 + 1]));
+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;
+ }
- put_unaligned(cpu_to_le16((l + r) / 2), &output[i]);
+ 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)
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->ep->codec->resume(out->ep->codec_data);
-
out->audio_state = AUDIO_A2DP_STATE_STARTED;
}
return -1;
}
- /* currently Android audioflinger is not able to provide mono stream on
+ /*
+ * 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
downmix_to_mono(out, buffer, bytes);
- return out->ep->codec->write_data(out->ep->codec_data,
- out->downmix_buf,
- bytes / 2,
- out->ep->fd) * 2;
+ in_buf = out->downmix_buf;
+ in_len = bytes / 2;
}
- return out->ep->codec->write_data(out->ep->codec_data, buffer,
- bytes, out->ep->fd);
+ if (!write_data(out, in_buf, in_len))
+ return -1;
+
+ return bytes;
}
static uint32_t out_get_sample_rate(const struct audio_stream *stream)
{
DBG("");
- /* We should return proper buffer size calculated by codec (so each
+ /*
+ * 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
{
DBG("");
- /* AudioFlinger can only provide stereo stream, so we return it here and
+ /*
+ * AudioFlinger can only provide stereo stream, so we return it here and
* later we'll downmix this to mono in case codec requires it
*/
DBG("%s", kvpairs);
str = strdup(kvpairs);
+ if (!str)
+ return -ENOMEM;
+
kvpair = strtok_r(str, ";", &saveptr);
for (; kvpair && *kvpair; kvpair = strtok_r(NULL, ";", &saveptr)) {
return -ENOSYS;
}
-static int audio_open_output_stream(struct audio_hw_device *dev,
+static int audio_open_output_stream_real(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
audio_output_flags_t flags,
struct audio_config *config,
- struct audio_stream_out **stream_out)
-
+ struct audio_stream_out **stream_out,
+ const char *address)
{
struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *) dev;
struct a2dp_stream_out *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 (!open_endpoint(out->ep, &out->cfg))
+ if (!open_endpoint(&out->ep, &out->cfg))
goto fail;
DBG("rate=%d channels=%d format=%d", out->cfg.rate,
return -EIO;
}
+#if ANDROID_VERSION >= PLATFORM_VER(5, 0, 0)
+static int audio_open_output_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ struct audio_stream_out **stream_out,
+ const char *address)
+{
+ return audio_open_output_stream_real(dev, handle, devices, flags,
+ config, stream_out, address);
+}
+#else
+static int audio_open_output_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ audio_output_flags_t flags,
+ struct audio_config *config,
+ struct audio_stream_out **stream_out)
+{
+ return audio_open_output_stream_real(dev, handle, devices, flags,
+ config, stream_out, NULL);
+}
+#endif
+
static void audio_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream)
{
return -ENOSYS;
}
-static int audio_open_input_stream(struct audio_hw_device *dev,
+static int audio_open_input_stream_real(struct audio_hw_device *dev,
audio_io_handle_t handle,
audio_devices_t devices,
struct audio_config *config,
- struct audio_stream_in **stream_in)
+ struct audio_stream_in **stream_in,
+ audio_input_flags_t flags,
+ const char *address,
+ audio_source_t source)
{
struct audio_stream_in *in;
return 0;
}
+#if ANDROID_VERSION >= PLATFORM_VER(5, 0, 0)
+static int audio_open_input_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ struct audio_config *config,
+ struct audio_stream_in **stream_in,
+ audio_input_flags_t flags,
+ const char *address,
+ audio_source_t source)
+{
+ return audio_open_input_stream_real(dev, handle, devices, config,
+ stream_in, flags, address,
+ source);
+}
+#else
+static int audio_open_input_stream(struct audio_hw_device *dev,
+ audio_io_handle_t handle,
+ audio_devices_t devices,
+ struct audio_config *config,
+ struct audio_stream_in **stream_in)
+{
+ return audio_open_input_stream_real(dev, handle, devices, config,
+ stream_in, 0, NULL, 0);
+}
+#endif
+
static void audio_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *stream_in)
{
return -ENOSYS;
}
+#if ANDROID_VERSION >= PLATFORM_VER(5, 0, 0)
+static int set_master_mute(struct audio_hw_device *dev, bool mute)
+{
+ DBG("");
+ return -ENOSYS;
+}
+
+static int get_master_mute(struct audio_hw_device *dev, bool *mute)
+{
+ DBG("");
+ return -ENOSYS;
+}
+
+static int create_audio_patch(struct audio_hw_device *dev,
+ unsigned int num_sources,
+ const struct audio_port_config *sources,
+ unsigned int num_sinks,
+ const struct audio_port_config *sinks,
+ audio_patch_handle_t *handle)
+{
+ DBG("");
+ return -ENOSYS;
+}
+
+static int release_audio_patch(struct audio_hw_device *dev,
+ audio_patch_handle_t handle)
+{
+ DBG("");
+ return -ENOSYS;
+}
+
+static int get_audio_port(struct audio_hw_device *dev, struct audio_port *port)
+{
+ DBG("");
+ return -ENOSYS;
+}
+
+static int set_audio_port_config(struct audio_hw_device *dev,
+ const struct audio_port_config *config)
+{
+ DBG("");
+ return -ENOSYS;
+}
+#endif
+
static int audio_close(hw_device_t *device)
{
struct a2dp_audio_dev *a2dp_dev = (struct a2dp_audio_dev *)device;
+ unsigned int i;
DBG("");
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);
/* 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");
+ info("Audio HAL: Socket closed");
- pthread_mutex_lock(&sk_mutex);
- close(audio_sk);
- audio_sk = -1;
- pthread_mutex_unlock(&sk_mutex);
- }
+ pthread_mutex_lock(&sk_mutex);
+ close(audio_sk);
+ audio_sk = -1;
+ pthread_mutex_unlock(&sk_mutex);
}
/* audio_sk is closed at this point, just cleanup endpoints states */
hw_device_t **device)
{
struct a2dp_audio_dev *a2dp_dev;
+ size_t i;
int err;
DBG("");
a2dp_dev->dev.open_input_stream = audio_open_input_stream;
a2dp_dev->dev.close_input_stream = audio_close_input_stream;
a2dp_dev->dev.dump = audio_dump;
+#if ANDROID_VERSION >= PLATFORM_VER(5, 0, 0)
+ a2dp_dev->dev.set_master_mute = set_master_mute;
+ a2dp_dev->dev.get_master_mute = get_master_mute;
+ a2dp_dev->dev.create_audio_patch = create_audio_patch;
+ a2dp_dev->dev.release_audio_patch = release_audio_patch;
+ a2dp_dev->dev.get_audio_port = get_audio_port;
+ a2dp_dev->dev.set_audio_port_config = set_audio_port_config;
+#endif
+
+ for (i = 0; i < NUM_CODECS; i++) {
+ const struct audio_codec *codec = audio_codecs[i].get_codec();
- /* Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
+ 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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