#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 */
-#define SBC_QUALITY_MIN_BITPOOL 33
-#define SBC_QUALITY_STEP 5
-
static const uint8_t a2dp_src_uuid[] = {
0x00, 0x00, 0x11, 0x0a, 0x00, 0x00, 0x10, 0x00,
0x80, 0x00, 0x00, 0x80, 0x5f, 0x9b, 0x34, 0xfb };
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;
-
- uint16_t payload_len;
-
- size_t in_frame_len;
- size_t in_buf_size;
-
- size_t out_frame_len;
-
- unsigned frame_duration;
- unsigned frames_per_packet;
-};
-
static void timespec_add(struct timespec *base, uint64_t time_us,
struct timespec *res)
{
return res.tv_sec * 1000000ll + res.tv_nsec / 1000ll;
}
-#if defined(ANDROID)
+#if ANDROID_VERSION < PLATFORM_VER(6, 0, 0)
/*
* Bionic does not have clock_nanosleep() prototype in time.h even though
* it provides its implementation.
struct timespec *remain);
#endif
-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 ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
- size_t len, struct media_packet *mp,
- size_t mp_data_len, size_t *written);
-static bool sbc_update_qos(void *codec_data, uint8_t op);
-
-#define QOS_POLICY_DEFAULT 0x00
-#define QOS_POLICY_DECREASE 0x01
-
-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);
- ssize_t (*encode_mediapacket) (void *codec_data, const uint8_t *buffer,
- size_t len, struct media_packet *mp,
- size_t mp_data_len, size_t *written);
- bool (*update_qos) (void *codec_data, uint8_t op);
-};
-
-static const struct audio_codec audio_codecs[] = {
- {
- .type = A2DP_CODEC_SBC,
-
- .get_presets = sbc_get_presets,
-
- .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,
- .encode_mediapacket = sbc_encode_mediapacket,
- .update_qos = sbc_update_qos,
- }
+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]))
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 void sbc_codec_calculate(struct sbc_data *sbc_data)
-{
- size_t in_frame_len;
- size_t out_frame_len;
- size_t num_frames;
-
- in_frame_len = sbc_get_codesize(&sbc_data->enc);
- out_frame_len = sbc_get_frame_length(&sbc_data->enc);
- num_frames = sbc_data->payload_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_frame_len = out_frame_len;
-
- sbc_data->frame_duration = sbc_get_frame_duration(&sbc_data->enc);
- sbc_data->frames_per_packet = num_frames;
-
- DBG("in_frame_len=%zu out_frame_len=%zu frames_per_packet=%zu",
- in_frame_len, out_frame_len, num_frames);
-}
-
-static int sbc_codec_init(struct audio_preset *preset, uint16_t payload_len,
- void **codec_data)
-{
- struct sbc_data *sbc_data;
-
- 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);
-
- sbc_data->payload_len = payload_len;
-
- sbc_codec_calculate(sbc_data);
-
- *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(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 ssize_t sbc_encode_mediapacket(void *codec_data, const uint8_t *buffer,
- size_t len, struct media_packet *mp,
- size_t mp_data_len, size_t *written)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
- size_t consumed = 0;
- size_t encoded = 0;
- uint8_t frame_count = 0;
-
- while (len - consumed >= sbc_data->in_frame_len &&
- mp_data_len - encoded >= sbc_data->out_frame_len &&
- frame_count < MAX_FRAMES_IN_PAYLOAD) {
- ssize_t read;
- ssize_t written = 0;
-
- read = sbc_encode(&sbc_data->enc, buffer + consumed,
- sbc_data->in_frame_len, mp->data + encoded,
- mp_data_len - encoded, &written);
-
- if (read < 0) {
- error("SBC: failed to encode block at frame %d (%zd)",
- frame_count, read);
- break;
- }
-
- frame_count++;
- consumed += read;
- encoded += written;
- }
-
- *written = encoded;
- mp->payload.frame_count = frame_count;
-
- return consumed;
-}
-
-static bool sbc_update_qos(void *codec_data, uint8_t op)
-{
- struct sbc_data *sbc_data = (struct sbc_data *) codec_data;
- uint8_t curr_bitpool = sbc_data->enc.bitpool;
- uint8_t new_bitpool = curr_bitpool;
-
- switch (op) {
- case QOS_POLICY_DEFAULT:
- new_bitpool = sbc_data->sbc.max_bitpool;
- break;
-
- case QOS_POLICY_DECREASE:
- if (curr_bitpool > SBC_QUALITY_MIN_BITPOOL) {
- new_bitpool = curr_bitpool - SBC_QUALITY_STEP;
- if (new_bitpool < SBC_QUALITY_MIN_BITPOOL)
- new_bitpool = SBC_QUALITY_MIN_BITPOOL;
- }
- break;
- }
-
- if (new_bitpool == curr_bitpool)
- return false;
-
- sbc_data->enc.bitpool = new_bitpool;
-
- sbc_codec_calculate(sbc_data);
-
- info("SBC: bitpool changed: %d -> %d", curr_bitpool, new_bitpool);
-
- return true;
-}
-
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 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 (ep)
+ ep_id = ep->id;
- if (ipc_open_stream_cmd(ep->id, &mtu, &fd, &preset) !=
+ if (ipc_open_stream_cmd(&ep_id, &mtu, &fd, &preset) !=
AUDIO_STATUS_SUCCESS)
return false;
- DBG("mtu=%u", mtu);
+ 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;
+ }
- payload_len = mtu - sizeof(*ep->mp);
+ *epp = ep;
+
+ payload_len = mtu;
+ if (ep->codec->use_rtp)
+ payload_len -= sizeof(struct rtp_header);
ep->fd = fd;
ep->mp = calloc(mtu, 1);
if (!ep->mp)
goto failed;
- ep->mp->hdr.v = 2;
- ep->mp->hdr.pt = 1;
- ep->mp->hdr.ssrc = htonl(1);
+
+ 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;
{
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 < 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]));
+ for (i = 0; i < frames; i++) {
+ int16_t l = get_le16(&input[i * 2]);
+ int16_t r = get_le16(&input[i * 2 + 1]);
- put_unaligned(cpu_to_le16((l + r) / 2), &output[i]);
+ put_le16((l + r) / 2, &output[i]);
}
}
int ret;
while (true) {
- ret = write(ep->fd, mp, sizeof(*mp) + bytes);
+ ret = write(ep->fd, mp, bytes);
if (ret >= 0)
break;
{
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;
* prepare media packet in advance so we don't waste time after
* wakeup
*/
- mp->hdr.sequence_number = htons(ep->seq++);
- mp->hdr.timestamp = htonl(ep->samples);
+ 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,
}
}
- /* in resync mode we'll just drop mediapackets */
- if (!ep->resync) {
+ /* 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 (do_write) {
+ if (ep->codec->use_rtp)
+ written += sizeof(struct rtp_header);
+
if (!write_to_endpoint(ep, written))
return false;
+ }
}
/*
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();
+
+ if (codec->load && !codec->load())
+ continue;
+
+ audio_codecs[i].loaded = true;
+ }
/*
* Note that &a2dp_dev->dev.common is the same pointer as a2dp_dev.
OSDN Git Service
