diff --git a/android/hal-audio.c b/android/hal-audio.c
index cf49514..78f89bd 100644
--- a/android/hal-audio.c
+++ b/android/hal-audio.c
size_t in_buf_size;
size_t out_frame_len;
- 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;
-
- uint16_t seq;
};
static inline void timespec_diff(struct timespec *a, struct timespec *b,
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);
+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 audio_codec {
uint8_t type;
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);
+ 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);
};
static const struct audio_codec audio_codecs[] = {
.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,
+ .encode_mediapacket = sbc_encode_mediapacket,
}
};
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;
};
static struct audio_endpoint audio_endpoints[MAX_AUDIO_ENDPOINTS];
sbc_data->in_buf_size = num_frames * in_frame_len;
sbc_data->out_frame_len = out_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;
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;
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)
+static int write_media_packet(struct a2dp_stream_out *out, size_t mp_data_len,
+ uint32_t input_samples)
{
+ struct audio_endpoint *ep = out->ep;
+ struct media_packet *mp = ep->mp;
struct timespec cur;
struct timespec diff;
- unsigned expected_frames;
+ uint32_t expected_samples;
int ret;
while (true) {
- ret = write(fd, mp, sizeof(*mp) + data_len);
+ ret = write(ep->fd, mp, sizeof(*mp) + mp_data_len);
if (ret >= 0)
break;
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;
+ timespec_diff(&cur, &ep->start, &diff);
+ expected_samples = (diff.tv_sec * 1000000ll + diff.tv_nsec / 1000ll) *
+ out->cfg.rate / 1000000ll;
/* AudioFlinger does not seem to provide any *working*
* API to provide data in some interval and will just
* 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);
+ if (ep->samples >= expected_samples)
+ usleep(input_samples * 1000000 / out->cfg.rate);
return ret;
}
return consumed;
}
-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;
- struct media_packet *mp = (struct media_packet *) sbc_data->out_buf;
- size_t free_space = sbc_data->out_buf_size - sizeof(*mp);
-
- mp->hdr.v = 2;
- mp->hdr.pt = 1;
- mp->hdr.ssrc = htonl(1);
-
- while (consumed < bytes) {
- size_t written = 0;
- ssize_t read;
- int ret;
-
- read = sbc_encode_mediapacket(codec_data, buffer + consumed,
- bytes - consumed, mp,
- free_space,
- &written);
-
- if (read <= 0) {
- error("sbc_encode_mediapacket failed (%zd)", read);
- goto done;
- }
-
- consumed += read;
-
- mp->hdr.sequence_number = htons(sbc_data->seq++);
- mp->hdr.timestamp = htonl(sbc_data->timestamp);
-
- /* AudioFlinger provides PCM 16bit stereo only, thus sample size
- * is always 4 bytes
- */
- sbc_data->timestamp += (read / 4);
-
- ret = write_media_packet(fd, sbc_data, mp, written);
- if (ret < 0)
- return ret;
- }
-
-done:
- /* 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)
{
codec->init(preset, mtu, &ep->codec_data);
codec->get_config(ep->codec_data, cfg);
+ 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);
+
+ ep->mp_data_len = mtu - sizeof(*ep->mp);
+
free(preset);
return true;
ep->fd = -1;
}
+ free(ep->mp);
+
ep->codec->cleanup(ep->codec_data);
ep->codec_data = NULL;
}
}
}
+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;
+ 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;
+
+ read = ep->codec->encode_mediapacket(ep->codec_data,
+ buffer + consumed,
+ bytes - consumed, mp,
+ free_space, &written);
+
+ /* This is non-fatal and we can just assume buffer was processed
+ * properly and wait for next one.
+ */
+ if (read <= 0)
+ return true;
+
+ consumed += read;
+
+ mp->hdr.sequence_number = htons(ep->seq++);
+ mp->hdr.timestamp = htonl(ep->samples);
+
+ /* AudioFlinger provides 16bit PCM, so sample size is 2 bytes
+ * multipled 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;
+
+ ret = write_media_packet(out, written, samples);
+ if (ret < 0)
+ return false;
+ }
+
+ 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 (ipc_resume_stream_cmd(out->ep->id) != AUDIO_STATUS_SUCCESS)
return -1;
- out->ep->codec->resume(out->ep->codec_data);
+ clock_gettime(CLOCK_MONOTONIC, &out->ep->start);
+ out->ep->samples = 0;
out->audio_state = AUDIO_A2DP_STATE_STARTED;
}
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)