From: Andrei Emeltchenko <[email protected]>
Checks that ipc is connected
---
android/ipc.c | 5 +++++
android/ipc.h | 1 +
2 files changed, 6 insertions(+)
diff --git a/android/ipc.c b/android/ipc.c
index 8cd34ea..137e6f9 100644
--- a/android/ipc.c
+++ b/android/ipc.c
@@ -287,6 +287,11 @@ static gboolean cmd_connect_cb(GIOChannel *io, GIOCondition cond,
return FALSE;
}
+bool ipc_is_connected(struct ipc *ipc)
+{
+ return ipc->cmd_watch;
+}
+
struct ipc *ipc_init(const char *path, size_t size, int max_service_id,
bool notifications,
ipc_disconnect_cb cb, void *cb_data)
diff --git a/android/ipc.h b/android/ipc.h
index cc4e92d..171a77c 100644
--- a/android/ipc.h
+++ b/android/ipc.h
@@ -36,6 +36,7 @@ struct ipc;
typedef void (*ipc_disconnect_cb) (void *data);
+bool ipc_is_connected(struct ipc *ipc);
struct ipc *ipc_init(const char *path, size_t size, int max_service_id,
bool notifications,
ipc_disconnect_cb cb, void *cb_data);
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
The new check takes into account number of channels.
---
android/client/if-sco.c | 7 ++++++-
1 file changed, 6 insertions(+), 1 deletion(-)
diff --git a/android/client/if-sco.c b/android/client/if-sco.c
index 555c6f9..35ed3b0 100644
--- a/android/client/if-sco.c
+++ b/android/client/if-sco.c
@@ -354,16 +354,21 @@ fail:
static void loop_p(int argc, const char **argv)
{
+ int chan_out, chan_in;
+
RETURN_IF_NULL(if_audio_sco);
RETURN_IF_NULL(stream_out);
RETURN_IF_NULL(stream_in);
+ chan_out = popcount(stream_out->common.get_channels(&stream_out->common));
+ chan_in = popcount(stream_in->common.get_channels(&stream_in->common));
+
if (!buffer_size || !buffer_size_in) {
haltest_error("Invalid buffer sizes. Streams opened\n");
return;
}
- if (buffer_size != buffer_size_in) {
+ if (buffer_size / chan_out != buffer_size_in / chan_in) {
haltest_error("read/write buffers differ, not supported\n");
return;
}
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/client/if-sco.c | 18 ++++++++++--------
1 file changed, 10 insertions(+), 8 deletions(-)
diff --git a/android/client/if-sco.c b/android/client/if-sco.c
index b7377f4..6983f4e 100644
--- a/android/client/if-sco.c
+++ b/android/client/if-sco.c
@@ -455,6 +455,9 @@ static void read_p(int argc, const char **argv)
static void stop_p(int argc, const char **argv)
{
+ RETURN_IF_NULL(if_audio_sco);
+ RETURN_IF_NULL(play_thread);
+
pthread_mutex_lock(&state_mutex);
if (current_state == STATE_STOPPED || current_state == STATE_STOPPING) {
pthread_mutex_unlock(&state_mutex);
@@ -464,6 +467,9 @@ static void stop_p(int argc, const char **argv)
current_state = STATE_STOPPING;
pthread_mutex_unlock(&state_mutex);
+ pthread_join(play_thread, NULL);
+ play_thread = 0;
+
pthread_mutex_lock(&outstream_mutex);
stream_out->common.standby(&stream_out->common);
pthread_mutex_unlock(&outstream_mutex);
@@ -522,10 +528,8 @@ static void close_output_stream_p(int argc, const char **argv)
RETURN_IF_NULL(if_audio_sco);
RETURN_IF_NULL(stream_out);
- stop_p(argc, argv);
-
- haltest_info("Waiting for playback thread...\n");
- pthread_join(play_thread, NULL);
+ if (play_thread)
+ stop_p(argc, argv);
if_audio_sco->close_output_stream(if_audio_sco, stream_out);
@@ -583,10 +587,8 @@ static void close_input_stream_p(int argc, const char **argv)
RETURN_IF_NULL(if_audio_sco);
RETURN_IF_NULL(stream_in);
- stop_p(argc, argv);
-
- haltest_info("Waiting for playback thread...\n");
- pthread_join(play_thread, NULL);
+ if (play_thread)
+ stop_p(argc, argv);
if_audio_sco->close_input_stream(if_audio_sco, stream_in);
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
Use config parameter when opening output stream.
---
android/hal-sco.c | 20 +++++++++++++++-----
1 file changed, 15 insertions(+), 5 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index f3ccb0d..292ced2 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -582,7 +582,7 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
size_t resample_size;
uint16_t mtu;
- DBG("");
+ DBG("config %p device flags 0x%02x", config, devices);
if (ipc_connect_sco(&fd, &mtu) != SCO_STATUS_SUCCESS) {
error("sco: cannot get fd");
@@ -612,10 +612,20 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
out->stream.write = out_write;
out->stream.get_render_position = out_get_render_position;
- /* Configuration for Android */
- out->cfg.format = AUDIO_STREAM_DEFAULT_FORMAT;
- out->cfg.channels = AUDIO_CHANNEL_OUT_STEREO;
- out->cfg.rate = AUDIO_STREAM_DEFAULT_RATE;
+ if (config) {
+ DBG("config: rate %u chan mask %x format %d offload %p",
+ config->sample_rate, config->channel_mask,
+ config->format, &config->offload_info);
+
+ out->cfg.format = config->format;
+ out->cfg.channels = config->channel_mask;
+ out->cfg.rate = config->sample_rate;
+ } else {
+ out->cfg.format = AUDIO_STREAM_DEFAULT_FORMAT;
+ out->cfg.channels = AUDIO_CHANNEL_OUT_STEREO;
+ out->cfg.rate = AUDIO_STREAM_DEFAULT_RATE;
+ }
+
out->cfg.frame_num = OUT_STREAM_FRAMES;
/* we get wrong mtu size for some reason */
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/hal-sco.c | 10 +++-------
1 file changed, 3 insertions(+), 7 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index 31c93b4..42267e0 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -654,8 +654,6 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
return -ENOMEM;
}
- DBG("size %zd", out_get_buffer_size(&out->stream.common));
-
/* Channel numbers for resampler */
chan_num = 1;
@@ -667,9 +665,6 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
goto failed;
}
- DBG("Created resampler: input rate [%d] output rate [%d] channels [%d]",
- out->cfg.rate, AUDIO_STREAM_SCO_RATE, chan_num);
-
out->resample_frame_num = get_resample_frame_num(AUDIO_STREAM_SCO_RATE,
out->cfg.rate,
out->cfg.frame_num, 1);
@@ -688,8 +683,9 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
goto failed;
}
- DBG("resampler: frame num %u buf size %zd bytes",
- out->resample_frame_num, resample_size);
+ DBG("Resampler: input %d output %d chan %d frames %u size %zd",
+ out->cfg.rate, AUDIO_STREAM_SCO_RATE, chan_num,
+ out->resample_frame_num, resample_size);
*stream_out = &out->stream;
adev->out = out;
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
Add read and resampling from 8000 to 44100.
---
android/hal-sco.c | 151 +++++++++++++++++++++++++++++++++++++++++++++++++++++-
1 file changed, 149 insertions(+), 2 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index 943a541..2fa8fb7 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -40,6 +40,7 @@
#define OUT_BUFFER_SIZE 2560
#define OUT_STREAM_FRAMES 2560
+#define IN_STREAM_FRAMES /* 5292 */ 5120
#define SOCKET_POLL_TIMEOUT_MS 500
@@ -81,6 +82,10 @@ struct sco_stream_in {
struct audio_stream_in stream;
struct sco_audio_config cfg;
+
+ struct resampler_itfe *resampler;
+ int16_t *resample_buf;
+ uint32_t resample_frame_num;
};
struct sco_dev {
@@ -910,12 +915,109 @@ static int in_set_gain(struct audio_stream_in *stream, float gain)
return -ENOSYS;
}
+static bool read_data(struct sco_stream_in *in, char *buffer, size_t bytes)
+{
+ struct pollfd pfd;
+ size_t len, read_bytes = 0;
+
+ pfd.fd = sco_fd;
+ pfd.events = POLLIN | POLLHUP | POLLNVAL;
+
+ while (bytes > read_bytes) {
+ int ret;
+
+ /* poll for reading */
+ if (poll(&pfd, 1, SOCKET_POLL_TIMEOUT_MS) == 0) {
+ DBG("timeout fd %d", sco_fd);
+ return false;
+ }
+
+ if (pfd.revents & (POLLHUP | POLLNVAL)) {
+ error("error fd %d, events 0x%x", sco_fd, pfd.revents);
+ return false;
+ }
+
+ len = bytes - read_bytes > sco_mtu ? sco_mtu :
+ bytes - read_bytes;
+
+ ret = read(sco_fd, buffer + read_bytes, len);
+ if (ret > 0) {
+ read_bytes += ret;
+ DBG("read %d total %zd", ret, read_bytes);
+ continue;
+ }
+
+ if (errno == EAGAIN) {
+ ret = errno;
+ warn("read failed (%d)", ret);
+ continue;
+ }
+
+ if (errno != EINTR) {
+ ret = errno;
+ error("read failed (%d) fd %d bytes %zd", ret, sco_fd,
+ bytes);
+ return false;
+ }
+ }
+
+ DBG("read %zd bytes", read_bytes);
+
+ return true;
+}
+
static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
size_t bytes)
{
- DBG("");
+ struct sco_stream_in *in = (struct sco_stream_in *) stream;
+ size_t frame_size = audio_stream_frame_size(&stream->common);
+ size_t frame_num = bytes / frame_size;
+ size_t input_frame_num = frame_num;
+ void *read_buf = buffer;
+ size_t total, read_frames;
+ int ret;
- return -ENOSYS;
+ DBG("Read from fd %d bytes %zu", sco_fd, bytes);
+
+ if (!in->resampler && in->cfg.rate != AUDIO_STREAM_SCO_RATE) {
+ error("Cannot find resampler");
+ return -1;
+ }
+
+ if (in->resampler) {
+ input_frame_num = get_resample_frame_num(AUDIO_STREAM_SCO_RATE,
+ in->cfg.rate,
+ frame_num, 0);
+ if (input_frame_num > in->resample_frame_num) {
+ DBG("resize input frames from %zd to %d",
+ input_frame_num, in->resample_frame_num);
+ input_frame_num = in->resample_frame_num;
+ }
+
+ read_buf = in->resample_buf;
+ }
+
+ total = input_frame_num * sizeof(int16_t) * 1;
+
+ if(!read_data(in, read_buf, total))
+ return -1;
+
+ read_frames = input_frame_num;
+
+ ret = in->resampler->resample_from_input(in->resampler,
+ in->resample_buf,
+ &read_frames,
+ (int16_t *) buffer,
+ &frame_num);
+ if (ret) {
+ error("Failed to resample frames: %zd input %zd (%s)",
+ frame_num, input_frame_num, strerror(ret));
+ return -1;
+ }
+
+ DBG("resampler: remain %zd output %zd frames", read_frames, frame_num);
+
+ return bytes;
}
static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
@@ -933,6 +1035,8 @@ static int sco_open_input_stream(struct audio_hw_device *dev,
{
struct sco_dev *sco_dev = (struct sco_dev *) dev;
struct sco_stream_in *in;
+ int chan_num, ret;
+ size_t resample_size;
DBG("");
@@ -970,10 +1074,48 @@ static int sco_open_input_stream(struct audio_hw_device *dev,
in->cfg.rate = AUDIO_STREAM_DEFAULT_RATE;
}
+ in->cfg.frame_num = IN_STREAM_FRAMES;
+
+ /* Channel numbers for resampler */
+ chan_num = 1;
+
+ ret = create_resampler(AUDIO_STREAM_SCO_RATE, in->cfg.rate, chan_num,
+ RESAMPLER_QUALITY_DEFAULT, NULL,
+ &in->resampler);
+ if (ret) {
+ error("Failed to create resampler (%s)", strerror(ret));
+ goto failed;
+ }
+
+ in->resample_frame_num = get_resample_frame_num(AUDIO_STREAM_SCO_RATE,
+ in->cfg.rate,
+ in->cfg.frame_num, 0);
+
+ resample_size = sizeof(int16_t) * chan_num * in->resample_frame_num;
+
+ in->resample_buf = malloc(resample_size);
+ if (!in->resample_buf) {
+ error("failed to allocate resample buffer for %d frames",
+ in->resample_frame_num);
+ goto failed;
+ }
+
+ DBG("Resampler: input %d output %d chan %d frames %u size %zd",
+ AUDIO_STREAM_SCO_RATE, in->cfg.rate, chan_num,
+ in->resample_frame_num, resample_size);
+
*stream_in = &in->stream;
sco_dev->in = in;
return 0;
+failed:
+ if (in->resampler)
+ release_resampler(in->resampler);
+ free(in);
+ stream_in = NULL;
+ sco_dev->in = NULL;
+
+ return ret;
}
static void sco_close_input_stream(struct audio_hw_device *dev,
@@ -986,6 +1128,11 @@ static void sco_close_input_stream(struct audio_hw_device *dev,
close_sco_socket();
+ if (in->resampler) {
+ release_resampler(in->resampler);
+ free(in->resample_buf);
+ }
+
free(in);
sco_dev->in = NULL;
}
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
Release resampler on exit.
---
android/hal-sco.c | 4 +++-
1 file changed, 3 insertions(+), 1 deletion(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index 42267e0..943a541 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -729,8 +729,10 @@ static void sco_close_output_stream(struct audio_hw_device *dev,
close_sco_socket();
- if (out->resampler)
+ if (out->resampler) {
release_resampler(out->resampler);
+ free(out->resample_buf);
+ }
free(out->cache);
free(out->downmix_buf);
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/hal-sco.c | 56 +++++++++++++++++++++++++++++++++++++++++--------------
1 file changed, 42 insertions(+), 14 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index 5888275..701d15e 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -64,6 +64,8 @@ struct sco_stream_out {
int fd;
uint8_t *downmix_buf;
+ size_t samples;
+ struct timespec start;
struct resampler_itfe *resampler;
int16_t *resample_buf;
@@ -277,6 +279,21 @@ static void downmix_to_mono(struct sco_stream_out *out, const uint8_t *buffer,
}
}
+static uint64_t timespec_diff_us(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;
+
+ if (res.tv_nsec < 0) {
+ res.tv_sec--;
+ res.tv_nsec += 1000000000ll; /* 1sec */
+ }
+
+ return res.tv_sec * 1000000ll + res.tv_nsec / 1000ll;
+}
+
static bool write_data(struct sco_stream_out *out, const uint8_t *buffer,
size_t bytes)
{
@@ -284,13 +301,13 @@ static bool write_data(struct sco_stream_out *out, const uint8_t *buffer,
size_t len, written = 0;
int ret;
uint16_t mtu = /* out->cfg.mtu */ 48;
- uint8_t read_buf[mtu];
- bool do_write = false;
+ uint64_t audio_sent_us, audio_passed_us;
pfd.fd = out->fd;
pfd.events = POLLOUT | POLLIN | POLLHUP | POLLNVAL;
while (bytes > written) {
+ struct timespec now;
/* poll for sending */
if (poll(&pfd, 1, SOCKET_POLL_TIMEOUT_MS) == 0) {
@@ -303,27 +320,38 @@ static bool write_data(struct sco_stream_out *out, const uint8_t *buffer,
return false;
}
- /* FIXME synchronize by time instead of read() */
- if (pfd.revents & POLLIN) {
- ret = read(out->fd, read_buf, mtu);
- if (ret < 0) {
- error("Error reading fd %d (%s)", out->fd,
- strerror(errno));
- return false;
- }
- do_write = true;
+ clock_gettime(CLOCK_REALTIME, &now);
+ /* Mark start of the stream */
+ if (!out->samples)
+ memcpy(&out->start, &now, sizeof(out->start));
+
+ audio_sent_us = out->samples * 1000000ll / AUDIO_STREAM_SCO_RATE;
+ audio_passed_us = timespec_diff_us(&now, &out->start);
+ if ((int) (audio_sent_us - audio_passed_us) > 1500) {
+ struct timespec timeout = {0,
+ (audio_sent_us -
+ audio_passed_us) * 1000};
+ DBG("Sleeping for %d ms",
+ (int) (audio_sent_us - audio_passed_us));
+ nanosleep(&timeout, NULL);
+ } else if ((int)(audio_passed_us - audio_sent_us) > 50000) {
+ DBG("\n\nResync\n\n");
+ out->samples = 0;
+ memcpy(&out->start, &now, sizeof(out->start));
}
- if (!do_write)
- continue;
len = bytes - written > mtu ? mtu : bytes - written;
ret = write(out->fd, buffer + written, len);
if (ret > 0) {
written += ret;
- do_write = false;
+
+ out->samples += ret / 2;
+
+ DBG("written %d samples %zd total %zd bytes",
+ ret, out->samples, written);
continue;
}
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/client/if-sco.c | 51 +++++++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 51 insertions(+)
diff --git a/android/client/if-sco.c b/android/client/if-sco.c
index b7f5a80..c71d0b4 100644
--- a/android/client/if-sco.c
+++ b/android/client/if-sco.c
@@ -23,8 +23,10 @@
audio_hw_device_t *if_audio_sco = NULL;
static struct audio_stream_out *stream_out = NULL;
+static struct audio_stream_in *stream_in = NULL;
static size_t buffer_size = 0;
+static size_t buffer_size_in = 0;
static pthread_t play_thread = 0;
static pthread_mutex_t outstream_mutex = PTHREAD_MUTEX_INITIALIZER;
static pthread_mutex_t state_mutex = PTHREAD_MUTEX_INITIALIZER;
@@ -340,6 +342,53 @@ static void close_output_stream_p(int argc, const char **argv)
buffer_size = 0;
}
+static void open_input_stream_p(int argc, const char **argv)
+{
+ int err;
+
+ RETURN_IF_NULL(if_audio_sco);
+
+ pthread_mutex_lock(&state_mutex);
+ if (current_state == STATE_PLAYING) {
+ haltest_error("Already playing!\n");
+ pthread_mutex_unlock(&state_mutex);
+ return;
+ }
+ pthread_mutex_unlock(&state_mutex);
+
+ err = if_audio_sco->open_input_stream(if_audio_sco,
+ 0,
+ AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET,
+ NULL,
+ &stream_in);
+ if (err < 0) {
+ haltest_error("open output stream returned %d\n", err);
+ return;
+ }
+
+ buffer_size_in = stream_in->common.get_buffer_size(&stream_in->common);
+ if (buffer_size_in == 0)
+ haltest_error("Invalid buffer size received!\n");
+ else
+ haltest_info("Using buffer size: %zu\n", buffer_size_in);
+}
+
+static void close_input_stream_p(int argc, const char **argv)
+{
+ RETURN_IF_NULL(if_audio_sco);
+ RETURN_IF_NULL(stream_in);
+
+ stop_p(argc, argv);
+
+ haltest_info("Waiting for playback thread...\n");
+ pthread_join(play_thread, NULL);
+
+ if_audio_sco->close_input_stream(if_audio_sco, stream_in);
+
+ stream_in = NULL;
+ buffer_size_in = 0;
+}
+
static void cleanup_p(int argc, const char **argv)
{
int err;
@@ -499,6 +548,8 @@ static struct method methods[] = {
STD_METHOD(cleanup),
STD_METHOD(open_output_stream),
STD_METHOD(close_output_stream),
+ STD_METHOD(open_input_stream),
+ STD_METHOD(close_input_stream),
STD_METHODH(play, "<path to pcm file>"),
STD_METHOD(stop),
STD_METHOD(suspend),
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
Android may open input/output stream independently so we use global sco
file descriptor and mutexes.
---
android/hal-sco.c | 85 ++++++++++++++++++++++++++++++++-----------------------
1 file changed, 50 insertions(+), 35 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index e905d6b..31c93b4 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -46,6 +46,10 @@
static int listen_sk = -1;
static int ipc_sk = -1;
+static int sco_fd = -1;
+static uint16_t sco_mtu = 0;
+static pthread_mutex_t sco_mutex = PTHREAD_MUTEX_INITIALIZER;
+
static pthread_t ipc_th = 0;
static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
@@ -53,7 +57,6 @@ struct sco_audio_config {
uint32_t rate;
uint32_t channels;
uint32_t frame_num;
- uint16_t mtu;
audio_format_t format;
};
@@ -61,7 +64,6 @@ struct sco_stream_out {
struct audio_stream_out stream;
struct sco_audio_config cfg;
- int fd;
uint8_t *downmix_buf;
uint8_t *cache;
@@ -79,7 +81,6 @@ struct sco_stream_in {
struct audio_stream_in stream;
struct sco_audio_config cfg;
- int fd;
};
struct sco_dev {
@@ -257,18 +258,25 @@ failed:
return SCO_STATUS_FAILED;
}
-static int ipc_connect_sco(int *fd, uint16_t *mtu)
+static int ipc_connect_sco(void)
{
struct sco_rsp_connect rsp;
size_t rsp_len = sizeof(rsp);
- int ret;
+ int ret = SCO_STATUS_SUCCESS;
DBG("");
- ret = sco_ipc_cmd(SCO_SERVICE_ID, SCO_OP_CONNECT, 0, NULL, &rsp_len,
- &rsp, fd);
+ pthread_mutex_lock(&sco_mutex);
+
+ if (sco_fd < 0) {
+ ret = sco_ipc_cmd(SCO_SERVICE_ID, SCO_OP_CONNECT, 0, NULL,
+ &rsp_len, &rsp, &sco_fd);
+
+ /* Sometimes mtu returned is wrong */
+ sco_mtu = /* rsp.mtu */ 48;
+ }
- *mtu = rsp.mtu;
+ pthread_mutex_unlock(&sco_mutex);
return ret;
}
@@ -311,28 +319,27 @@ static bool write_data(struct sco_stream_out *out, const uint8_t *buffer,
struct pollfd pfd;
size_t len, written = 0;
int ret;
- uint16_t mtu = out->cfg.mtu;
uint8_t *p;
uint64_t audio_sent_us, audio_passed_us;
- pfd.fd = out->fd;
- pfd.events = POLLOUT | POLLIN | POLLHUP | POLLNVAL;
+ pfd.fd = sco_fd;
+ pfd.events = POLLOUT | POLLHUP | POLLNVAL;
while (bytes > written) {
struct timespec now;
/* poll for sending */
if (poll(&pfd, 1, SOCKET_POLL_TIMEOUT_MS) == 0) {
- DBG("timeout fd %d", out->fd);
+ DBG("timeout fd %d", sco_fd);
return false;
}
if (pfd.revents & (POLLHUP | POLLNVAL)) {
- error("error fd %d, events 0x%x", out->fd, pfd.revents);
+ error("error fd %d, events 0x%x", sco_fd, pfd.revents);
return false;
}
- len = bytes - written > mtu ? mtu : bytes - written;
+ len = bytes - written > sco_mtu ? sco_mtu : bytes - written;
clock_gettime(CLOCK_REALTIME, &now);
/* Mark start of the stream */
@@ -357,11 +364,11 @@ static bool write_data(struct sco_stream_out *out, const uint8_t *buffer,
if (out->cache_len) {
DBG("First packet cache_len %zd", out->cache_len);
memcpy(out->cache + out->cache_len, buffer,
- mtu - out->cache_len);
+ sco_mtu - out->cache_len);
p = out->cache;
}
- if (bytes - written >= mtu)
+ if (bytes - written >= sco_mtu)
p = (void *) buffer + written;
else {
memcpy(out->cache, buffer + written, bytes - written);
@@ -371,10 +378,10 @@ static bool write_data(struct sco_stream_out *out, const uint8_t *buffer,
continue;
}
- ret = write(out->fd, p, len);
+ ret = write(sco_fd, p, len);
if (ret > 0) {
if (out->cache_len) {
- written = mtu - out->cache_len;
+ written = sco_mtu - out->cache_len;
out->cache_len = 0;
} else
written += ret;
@@ -394,7 +401,7 @@ static bool write_data(struct sco_stream_out *out, const uint8_t *buffer,
if (errno != EINTR) {
ret = errno;
- error("write failed (%d) fd %d bytes %zd", ret, out->fd,
+ error("write failed (%d) fd %d bytes %zd", ret, sco_fd,
bytes);
return false;
}
@@ -414,7 +421,7 @@ static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
void *send_buf = out->downmix_buf;
size_t total;
- DBG("write to fd %d bytes %zu", out->fd, bytes);
+ DBG("write to fd %d bytes %zu", sco_fd, bytes);
if (!out->downmix_buf) {
error("sco: downmix buffer not initialized");
@@ -585,19 +592,17 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
{
struct sco_dev *adev = (struct sco_dev *) dev;
struct sco_stream_out *out;
- int fd = -1;
int chan_num, ret;
size_t resample_size;
- uint16_t mtu;
DBG("config %p device flags 0x%02x", config, devices);
- if (ipc_connect_sco(&fd, &mtu) != SCO_STATUS_SUCCESS) {
+ if (ipc_connect_sco() != SCO_STATUS_SUCCESS) {
error("sco: cannot get fd");
return -EIO;
}
- DBG("got sco fd %d mtu %u", fd, mtu);
+ DBG("got sco fd %d mtu %u", sco_fd, sco_mtu);
out = calloc(1, sizeof(struct sco_stream_out));
if (!out)
@@ -636,16 +641,13 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
out->cfg.frame_num = OUT_STREAM_FRAMES;
- /* we get wrong mtu size for some reason */
- out->cfg.mtu = /* mtu */ 48;
-
out->downmix_buf = malloc(out_get_buffer_size(&out->stream.common));
if (!out->downmix_buf) {
free(out);
return -ENOMEM;
}
- out->cache = malloc(out->cfg.mtu);
+ out->cache = malloc(sco_mtu);
if (!out->cache) {
free(out->downmix_buf);
free(out);
@@ -691,7 +693,6 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
*stream_out = &out->stream;
adev->out = out;
- out->fd = fd;
return 0;
failed:
@@ -707,18 +708,30 @@ failed:
return ret;
}
+static void close_sco_socket(void)
+{
+ DBG("");
+
+ pthread_mutex_lock(&sco_mutex);
+
+ if (sco_fd >= 0) {
+ shutdown(sco_fd, SHUT_RDWR);
+ close(sco_fd);
+ sco_fd = -1;
+ }
+
+ pthread_mutex_unlock(&sco_mutex);
+}
+
static void sco_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream_out)
{
struct sco_dev *sco_dev = (struct sco_dev *) dev;
struct sco_stream_out *out = (struct sco_stream_out *) stream_out;
- DBG("dev %p stream %p fd %d", dev, out, sco_dev->out->fd);
+ DBG("dev %p stream %p fd %d", dev, out, sco_fd);
- if (out && out->fd >= 0) {
- close(out->fd);
- out->fd = -1;
- }
+ close_sco_socket();
if (out->resampler)
release_resampler(out->resampler);
@@ -971,7 +984,9 @@ static void sco_close_input_stream(struct audio_hw_device *dev,
struct sco_dev *sco_dev = (struct sco_dev *) dev;
struct sco_stream_in *in = (struct sco_stream_in *) stream_in;
- DBG("dev %p stream %p fd %d", dev, in, in->fd);
+ DBG("dev %p stream %p fd %d", dev, in, sco_fd);
+
+ close_sco_socket();
free(in);
sco_dev->in = NULL;
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/hal-sco.c | 175 +++++++++++++++++++++++++++++++++++++++++++++++++++++-
1 file changed, 173 insertions(+), 2 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index 292ced2..162ce25 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -75,9 +75,17 @@ struct sco_stream_out {
uint32_t resample_frame_num;
};
+struct sco_stream_in {
+ struct audio_stream_in stream;
+
+ struct sco_audio_config cfg;
+ int fd;
+};
+
struct sco_dev {
struct audio_hw_device dev;
struct sco_stream_out *out;
+ struct sco_stream_in *in;
};
/*
@@ -787,23 +795,186 @@ static size_t sco_get_input_buffer_size(const struct audio_hw_device *dev,
return -ENOSYS;
}
+static uint32_t in_get_sample_rate(const struct audio_stream *stream)
+{
+ struct sco_stream_in *in = (struct sco_stream_in *) stream;
+
+ DBG("rate %u", in->cfg.rate);
+
+ return in->cfg.rate;
+}
+
+static int in_set_sample_rate(struct audio_stream *stream, uint32_t rate)
+{
+ DBG("rate %u", rate);
+
+ return 0;
+}
+
+static size_t in_get_buffer_size(const struct audio_stream *stream)
+{
+ struct sco_stream_in *in = (struct sco_stream_in *) stream;
+ size_t size = audio_stream_frame_size(&in->stream.common) *
+ in->cfg.frame_num;
+
+ DBG("buf size %zd", size);
+
+ return size;
+}
+
+static uint32_t in_get_channels(const struct audio_stream *stream)
+{
+ struct sco_stream_in *in = (struct sco_stream_in *) stream;
+
+ DBG("channels num: %u", popcount(in->cfg.channels));
+
+ return in->cfg.channels;
+}
+
+static audio_format_t in_get_format(const struct audio_stream *stream)
+{
+ struct sco_stream_in *in = (struct sco_stream_in *) stream;
+
+ DBG("format: %u", in->cfg.format);
+
+ return in->cfg.format;
+}
+
+static int in_set_format(struct audio_stream *stream, audio_format_t format)
+{
+ DBG("");
+
+ return -ENOSYS;
+}
+
+static int in_standby(struct audio_stream *stream)
+{
+ DBG("");
+
+ return 0;
+}
+
+static int in_dump(const struct audio_stream *stream, int fd)
+{
+ DBG("");
+
+ return -ENOSYS;
+}
+
+static int in_set_parameters(struct audio_stream *stream, const char *kvpairs)
+{
+ DBG("%s", kvpairs);
+
+ return 0;
+}
+
+static char *in_get_parameters(const struct audio_stream *stream,
+ const char *keys)
+{
+ DBG("");
+
+ return strdup("");
+}
+
+static int in_add_audio_effect(const struct audio_stream *stream,
+ effect_handle_t effect)
+{
+ DBG("");
+
+ return -ENOSYS;
+}
+
+static int in_remove_audio_effect(const struct audio_stream *stream,
+ effect_handle_t effect)
+{
+ DBG("");
+
+ return -ENOSYS;
+}
+
+static int in_set_gain(struct audio_stream_in *stream, float gain)
+{
+ DBG("");
+
+ return -ENOSYS;
+}
+
+static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
+ size_t bytes)
+{
+ DBG("");
+
+ return -ENOSYS;
+}
+
+static uint32_t in_get_input_frames_lost(struct audio_stream_in *stream)
+{
+ DBG("");
+
+ return -ENOSYS;
+}
+
static int sco_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)
{
+ struct sco_dev *sco_dev = (struct sco_dev *) dev;
+ struct sco_stream_in *in;
+
DBG("");
+ in = calloc(1, sizeof(struct sco_stream_in));
+ if (!in)
+ return -ENOMEM;
+
+ in->stream.common.get_sample_rate = in_get_sample_rate;
+ in->stream.common.set_sample_rate = in_set_sample_rate;
+ in->stream.common.get_buffer_size = in_get_buffer_size;
+ in->stream.common.get_channels = in_get_channels;
+ in->stream.common.get_format = in_get_format;
+ in->stream.common.set_format = in_set_format;
+ in->stream.common.standby = in_standby;
+ in->stream.common.dump = in_dump;
+ in->stream.common.set_parameters = in_set_parameters;
+ in->stream.common.get_parameters = in_get_parameters;
+ in->stream.common.add_audio_effect = in_add_audio_effect;
+ in->stream.common.remove_audio_effect = in_remove_audio_effect;
+ in->stream.set_gain = in_set_gain;
+ in->stream.read = in_read;
+ in->stream.get_input_frames_lost = in_get_input_frames_lost;
+
+ if (config) {
+ DBG("config: rate %u chan mask %x format %d offload %p",
+ config->sample_rate, config->channel_mask,
+ config->format, &config->offload_info);
+
+ in->cfg.format = config->format;
+ in->cfg.channels = config->channel_mask;
+ in->cfg.rate = config->sample_rate;
+ } else {
+ in->cfg.format = AUDIO_STREAM_DEFAULT_FORMAT;
+ in->cfg.channels = AUDIO_CHANNEL_OUT_MONO;
+ in->cfg.rate = AUDIO_STREAM_DEFAULT_RATE;
+ }
+
+ *stream_in = &in->stream;
+ sco_dev->in = in;
+
return 0;
}
static void sco_close_input_stream(struct audio_hw_device *dev,
struct audio_stream_in *stream_in)
{
- DBG("");
+ struct sco_dev *sco_dev = (struct sco_dev *) dev;
+ struct sco_stream_in *in = (struct sco_stream_in *) stream_in;
+
+ DBG("dev %p stream %p fd %d", dev, in, in->fd);
- free(stream_in);
+ free(in);
+ sco_dev->in = NULL;
}
static int sco_dump(const audio_hw_device_t *device, int fd)
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
For 8k choose buffer size 576 which is multiple from 48 and 64.
---
android/hal-sco.c | 10 +++++++++-
1 file changed, 9 insertions(+), 1 deletion(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index 5b05601..fe0d98c 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -40,7 +40,7 @@
#define OUT_BUFFER_SIZE 2560
#define OUT_STREAM_FRAMES 2560
-#define IN_STREAM_FRAMES /* 5292 */ 5120
+#define IN_STREAM_FRAMES 5292
#define SOCKET_POLL_TIMEOUT_MS 500
@@ -490,6 +490,10 @@ static size_t out_get_buffer_size(const struct audio_stream *stream)
size_t size = audio_stream_frame_size(&out->stream.common) *
out->cfg.frame_num;
+ /* buffer size without resampling */
+ if (out->cfg.rate == AUDIO_STREAM_SCO_RATE)
+ size = 576 * 2;
+
DBG("buf size %zd", size);
return size;
@@ -836,6 +840,10 @@ static size_t in_get_buffer_size(const struct audio_stream *stream)
size_t size = audio_stream_frame_size(&in->stream.common) *
in->cfg.frame_num;
+ /* buffer size without resampling */
+ if (in->cfg.rate == AUDIO_STREAM_SCO_RATE)
+ size = 576;
+
DBG("buf size %zd", size);
return size;
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/hal-sco.c | 33 +++++++++++++++++++--------------
1 file changed, 19 insertions(+), 14 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index e3d5242..5b05601 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -977,7 +977,7 @@ static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
size_t frame_num = bytes / frame_size;
size_t input_frame_num = frame_num;
void *read_buf = buffer;
- size_t total, read_frames;
+ size_t total = bytes;
int ret;
DBG("Read from fd %d bytes %zu", sco_fd, bytes);
@@ -998,27 +998,29 @@ static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
}
read_buf = in->resample_buf;
- }
- total = input_frame_num * sizeof(int16_t) * 1;
+ total = input_frame_num * sizeof(int16_t) * 1;
+ }
if(!read_data(in, read_buf, total))
return -1;
- read_frames = input_frame_num;
-
- ret = in->resampler->resample_from_input(in->resampler,
+ if (in->resampler) {
+ ret = in->resampler->resample_from_input(in->resampler,
in->resample_buf,
- &read_frames,
+ &input_frame_num,
(int16_t *) buffer,
&frame_num);
- if (ret) {
- error("Failed to resample frames: %zd input %zd (%s)",
- frame_num, input_frame_num, strerror(ret));
- return -1;
- }
+ if (ret) {
+ error("Failed to resample frames: %zd input %zd (%s)",
+ frame_num, input_frame_num,
+ strerror(ret));
+ return -1;
+ }
- DBG("resampler: remain %zd output %zd frames", read_frames, frame_num);
+ DBG("resampler: remain %zd output %zd frames", input_frame_num,
+ frame_num);
+ }
return bytes;
}
@@ -1084,6 +1086,9 @@ static int sco_open_input_stream(struct audio_hw_device *dev,
in->cfg.frame_num = IN_STREAM_FRAMES;
+ if (in->cfg.rate == AUDIO_STREAM_SCO_RATE)
+ goto skip_resampler;
+
/* Channel numbers for resampler */
chan_num = 1;
@@ -1111,7 +1116,7 @@ static int sco_open_input_stream(struct audio_hw_device *dev,
DBG("Resampler: input %d output %d chan %d frames %u size %zd",
AUDIO_STREAM_SCO_RATE, in->cfg.rate, chan_num,
in->resample_frame_num, resample_size);
-
+skip_resampler:
*stream_in = &in->stream;
sco_dev->in = in;
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/hal-sco.c | 13 +++++++------
1 file changed, 7 insertions(+), 6 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index 955f511..b4667e5 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -280,15 +280,16 @@ failed:
static int ipc_connect_sco(void)
{
- struct sco_rsp_connect rsp;
- size_t rsp_len = sizeof(rsp);
int ret = SCO_STATUS_SUCCESS;
- DBG("");
-
pthread_mutex_lock(&sco_mutex);
if (sco_fd < 0) {
+ struct sco_rsp_connect rsp;
+ size_t rsp_len = sizeof(rsp);
+
+ DBG("Connecting SCO");
+
ret = sco_ipc_cmd(SCO_SERVICE_ID, SCO_OP_CONNECT, 0, NULL,
&rsp_len, &rsp, &sco_fd);
@@ -443,7 +444,7 @@ static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
DBG("write to fd %d bytes %zu", sco_fd, bytes);
- if (sco_fd < 0)
+ if (ipc_connect_sco() != SCO_STATUS_SUCCESS)
return -1;
if (!out->downmix_buf) {
@@ -1007,7 +1008,7 @@ static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
DBG("Read from fd %d bytes %zu", sco_fd, bytes);
- if (sco_fd < 0)
+ if (ipc_connect_sco() != SCO_STATUS_SUCCESS)
return -1;
if (!in->resampler && in->cfg.rate != AUDIO_STREAM_SCO_RATE) {
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/hal-sco.c | 8 ++++----
1 file changed, 4 insertions(+), 4 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index 162ce25..e905d6b 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -713,11 +713,11 @@ static void sco_close_output_stream(struct audio_hw_device *dev,
struct sco_dev *sco_dev = (struct sco_dev *) dev;
struct sco_stream_out *out = (struct sco_stream_out *) stream_out;
- DBG("dev %p stream %p fd %d", dev, stream_out, sco_dev->out->fd);
+ DBG("dev %p stream %p fd %d", dev, out, sco_dev->out->fd);
- if (sco_dev->out && sco_dev->out->fd) {
- close(sco_dev->out->fd);
- sco_dev->out->fd = -1;
+ if (out && out->fd >= 0) {
+ close(out->fd);
+ out->fd = -1;
}
if (out->resampler)
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
Reads data from stream_in and write to specified file.
---
android/Android.mk | 3 +++
android/client/if-sco.c | 55 ++++++++++++++++++++++++++++++++++++++++++++-----
2 files changed, 53 insertions(+), 5 deletions(-)
diff --git a/android/Android.mk b/android/Android.mk
index a52aefe..79058e0 100644
--- a/android/Android.mk
+++ b/android/Android.mk
@@ -167,6 +167,9 @@ LOCAL_C_INCLUDES += \
$(call include-path-for, system-core) \
$(call include-path-for, libhardware) \
+LOCAL_C_INCLUDES += \
+ $(LOCAL_PATH)/bluez/android \
+
LOCAL_CFLAGS := $(BLUEZ_COMMON_CFLAGS)
LOCAL_SHARED_LIBRARIES := libhardware
diff --git a/android/client/if-sco.c b/android/client/if-sco.c
index 39c3e3a..b5fa926 100644
--- a/android/client/if-sco.c
+++ b/android/client/if-sco.c
@@ -15,6 +15,8 @@
*
*/
+#include "../src/shared/util.h"
+
#include "if-main.h"
#include "../hal-utils.h"
#include "pthread.h"
@@ -239,12 +241,29 @@ static void *playback_thread(void *data)
return NULL;
}
+static void write_stereo_pcm16(char *buffer, size_t len, FILE *out)
+{
+ const int16_t *input = (const void *) buffer;
+ int16_t sample[2];
+ size_t i;
+
+ for (i = 0; i < len / 2; i++) {
+ int16_t mono = get_unaligned(&input[i]);
+
+ put_unaligned(mono, &sample[0]);
+ put_unaligned(mono, &sample[1]);
+
+ fwrite(sample, sizeof(sample), 1, out);
+ }
+}
+
static void *read_thread(void *data)
{
int (*filbuff_cb) (short*, void*) = feed_from_in;
- short buffer[buffer_size / sizeof(short)];
+ short buffer[buffer_size_in / sizeof(short)];
size_t len = 0;
void *cb_data = NULL;
+ FILE *out = data;
pthread_mutex_lock(&state_mutex);
current_state = STATE_PLAYING;
@@ -266,9 +285,18 @@ static void *read_thread(void *data)
pthread_mutex_unlock(&state_mutex);
len = filbuff_cb(buffer, cb_data);
- haltest_info("len %zd\n", len);
+
+ haltest_info("Read %zd bytes\n", len);
+
+ if (out) {
+ write_stereo_pcm16((char *) buffer, len, out);
+ haltest_info("Written %zd bytes\n", len * 2);
+ }
} while (len);
+ if (out)
+ fclose(out);
+
pthread_mutex_lock(&state_mutex);
current_state = STATE_STOPPED;
pthread_mutex_unlock(&state_mutex);
@@ -355,9 +383,26 @@ static void loop_p(int argc, const char **argv)
static void read_p(int argc, const char **argv)
{
+ const char *fname = NULL;
+ FILE *out = NULL;
+
RETURN_IF_NULL(if_audio_sco);
RETURN_IF_NULL(stream_in);
+ if (argc < 3) {
+ haltest_error("Invalid audio file path.\n");
+ haltest_info("Using read and through away\n");
+ } else {
+ fname = argv[2];
+ out = fopen(fname, "w");
+ if (!out) {
+ haltest_error("Cannot open file: %s\n", fname);
+ return;
+ }
+
+ haltest_info("Reading to file: %s\n", fname);
+ }
+
if (!buffer_size_in) {
haltest_error("Invalid buffer sizes. Streams opened\n");
return;
@@ -371,10 +416,10 @@ static void read_p(int argc, const char **argv)
}
pthread_mutex_unlock(&state_mutex);
- if (pthread_create(&play_thread, NULL, read_thread,
- stream_in) != 0)
+ if (pthread_create(&play_thread, NULL, read_thread, out) != 0) {
haltest_error("Cannot create playback thread!\n");
-
+ fclose(out);
+ }
}
static void stop_p(int argc, const char **argv)
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/handsfree.c | 13 +++++++++++--
1 file changed, 11 insertions(+), 2 deletions(-)
diff --git a/android/handsfree.c b/android/handsfree.c
index b530aa9..49e6a22 100644
--- a/android/handsfree.c
+++ b/android/handsfree.c
@@ -169,6 +169,8 @@ static GIOChannel *hsp_server = NULL;
static GIOChannel *sco_server = NULL;
+static void bt_sco_connect(const void *buf, uint16_t len);
+
static void device_set_state(uint8_t state)
{
struct hal_ev_handsfree_conn_state ev;
@@ -892,6 +894,9 @@ static void connect_sco_cb(GIOChannel *chan, GError *err, gpointer user_data)
sco_watch_cb, NULL);
device_set_audio_state(HAL_EV_HANDSFREE_AUDIO_STATE_CONNECTED);
+
+ if (ipc_is_connected(sco_ipc))
+ bt_sco_connect(NULL, 0);
}
static bool connect_sco(void)
@@ -2577,8 +2582,12 @@ static void bt_sco_connect(const void *buf, uint16_t len)
DBG("");
- if (!device.sco)
- goto failed;
+ if (!device.sco) {
+ DBG("SCO is not established, connecting...");
+ if (!connect_audio())
+ goto failed;
+ return;
+ }
err = NULL;
if (!bt_io_get(device.sco, &err, BT_IO_OPT_MTU, &rsp.mtu,
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/hal-sco.c | 81 ++++++++++++++++++++++++++++++++++++++-----------------
1 file changed, 57 insertions(+), 24 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index fe0d98c..955f511 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -54,6 +54,9 @@ static pthread_mutex_t sco_mutex = PTHREAD_MUTEX_INITIALIZER;
static pthread_t ipc_th = 0;
static pthread_mutex_t sk_mutex = PTHREAD_MUTEX_INITIALIZER;
+static struct sco_stream_in *sco_stream_in = NULL;
+static struct sco_stream_out *sco_stream_out = NULL;
+
struct sco_audio_config {
uint32_t rate;
uint32_t channels;
@@ -78,6 +81,18 @@ struct sco_stream_out {
uint32_t resample_frame_num;
};
+static void sco_close_socket(void)
+{
+ DBG("sco fd %d", sco_fd);
+
+ if (sco_fd < 0)
+ return;
+
+ shutdown(sco_fd, SHUT_RDWR);
+ close(sco_fd);
+ sco_fd = -1;
+}
+
struct sco_stream_in {
struct audio_stream_in stream;
@@ -428,6 +443,9 @@ static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
DBG("write to fd %d bytes %zu", sco_fd, bytes);
+ if (sco_fd < 0)
+ return -1;
+
if (!out->downmix_buf) {
error("sco: downmix buffer not initialized");
return -1;
@@ -606,17 +624,22 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
DBG("config %p device flags 0x%02x", config, devices);
+ if (sco_stream_out) {
+ DBG("stream_out already open");
+ return -EIO;
+ }
+
if (ipc_connect_sco() != SCO_STATUS_SUCCESS) {
error("sco: cannot get fd");
return -EIO;
}
- DBG("got sco fd %d mtu %u", sco_fd, sco_mtu);
-
out = calloc(1, sizeof(struct sco_stream_out));
if (!out)
return -ENOMEM;
+ DBG("stream %p sco fd %d mtu %u", out, sco_fd, sco_mtu);
+
out->stream.common.get_sample_rate = out_get_sample_rate;
out->stream.common.set_sample_rate = out_set_sample_rate;
out->stream.common.get_buffer_size = out_get_buffer_size;
@@ -701,6 +724,7 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
skip_resampler:
*stream_out = &out->stream;
adev->out = out;
+ sco_stream_out = out;
return 0;
failed:
@@ -712,25 +736,11 @@ failed:
free(out);
stream_out = NULL;
adev->out = NULL;
+ sco_stream_out = NULL;
return ret;
}
-static void close_sco_socket(void)
-{
- DBG("");
-
- pthread_mutex_lock(&sco_mutex);
-
- if (sco_fd >= 0) {
- shutdown(sco_fd, SHUT_RDWR);
- close(sco_fd);
- sco_fd = -1;
- }
-
- pthread_mutex_unlock(&sco_mutex);
-}
-
static void sco_close_output_stream(struct audio_hw_device *dev,
struct audio_stream_out *stream_out)
{
@@ -739,8 +749,6 @@ static void sco_close_output_stream(struct audio_hw_device *dev,
DBG("dev %p stream %p fd %d", dev, out, sco_fd);
- close_sco_socket();
-
if (out->resampler) {
release_resampler(out->resampler);
free(out->resample_buf);
@@ -750,6 +758,15 @@ static void sco_close_output_stream(struct audio_hw_device *dev,
free(out->downmix_buf);
free(out);
sco_dev->out = NULL;
+
+ pthread_mutex_lock(&sco_mutex);
+
+ sco_stream_out = NULL;
+
+ if (!sco_stream_in)
+ sco_close_socket();
+
+ pthread_mutex_unlock(&sco_mutex);
}
static int sco_set_parameters(struct audio_hw_device *dev,
@@ -990,6 +1007,9 @@ static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
DBG("Read from fd %d bytes %zu", sco_fd, bytes);
+ if (sco_fd < 0)
+ return -1;
+
if (!in->resampler && in->cfg.rate != AUDIO_STREAM_SCO_RATE) {
error("Cannot find resampler");
return -1;
@@ -1053,15 +1073,18 @@ static int sco_open_input_stream(struct audio_hw_device *dev,
DBG("config %p device flags 0x%02x", config, devices);
- if (ipc_connect_sco() != SCO_STATUS_SUCCESS) {
- error("sco: cannot get fd");
- return -EIO;
+ if (sco_stream_in) {
+ DBG("stream_in already open");
+ ret = -EIO;
+ goto failed2;
}
in = calloc(1, sizeof(struct sco_stream_in));
if (!in)
return -ENOMEM;
+ DBG("stream %p sco fd %d mtu %u", in, sco_fd, sco_mtu);
+
in->stream.common.get_sample_rate = in_get_sample_rate;
in->stream.common.set_sample_rate = in_set_sample_rate;
in->stream.common.get_buffer_size = in_get_buffer_size;
@@ -1127,14 +1150,17 @@ static int sco_open_input_stream(struct audio_hw_device *dev,
skip_resampler:
*stream_in = &in->stream;
sco_dev->in = in;
+ sco_stream_in = in;
return 0;
failed:
if (in->resampler)
release_resampler(in->resampler);
free(in);
+failed2:
stream_in = NULL;
sco_dev->in = NULL;
+ sco_stream_in = NULL;
return ret;
}
@@ -1147,8 +1173,6 @@ static void sco_close_input_stream(struct audio_hw_device *dev,
DBG("dev %p stream %p fd %d", dev, in, sco_fd);
- close_sco_socket();
-
if (in->resampler) {
release_resampler(in->resampler);
free(in->resample_buf);
@@ -1156,6 +1180,15 @@ static void sco_close_input_stream(struct audio_hw_device *dev,
free(in);
sco_dev->in = NULL;
+
+ pthread_mutex_lock(&sco_mutex);
+
+ sco_stream_in = NULL;
+
+ if (!sco_stream_out)
+ sco_close_socket();
+
+ pthread_mutex_unlock(&sco_mutex);
}
static int sco_dump(const audio_hw_device_t *device, int fd)
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
Use connect_pending to identify when we need to return msg to HAL.
---
android/handsfree.c | 41 ++++++++++++++++++++++++++++++-----------
1 file changed, 30 insertions(+), 11 deletions(-)
diff --git a/android/handsfree.c b/android/handsfree.c
index 49e6a22..28fcb2c 100644
--- a/android/handsfree.c
+++ b/android/handsfree.c
@@ -169,7 +169,8 @@ static GIOChannel *hsp_server = NULL;
static GIOChannel *sco_server = NULL;
-static void bt_sco_connect(const void *buf, uint16_t len);
+static bool connect_pending = false;
+static void bt_sco_connect_rsp(bool is_error);
static void device_set_state(uint8_t state)
{
@@ -896,7 +897,7 @@ static void connect_sco_cb(GIOChannel *chan, GError *err, gpointer user_data)
device_set_audio_state(HAL_EV_HANDSFREE_AUDIO_STATE_CONNECTED);
if (ipc_is_connected(sco_ipc))
- bt_sco_connect(NULL, 0);
+ bt_sco_connect_rsp(false);
}
static bool connect_sco(void)
@@ -2574,22 +2575,22 @@ static void disable_sco_server(void)
}
}
-static void bt_sco_connect(const void *buf, uint16_t len)
+static void bt_sco_connect_rsp(bool is_error)
{
int fd;
- GError *err;
+ GError *err = NULL;
struct sco_rsp_connect rsp;
- DBG("");
+ DBG("error %d pending %d", is_error, connect_pending);
- if (!device.sco) {
- DBG("SCO is not established, connecting...");
- if (!connect_audio())
- goto failed;
+ if (!connect_pending)
return;
- }
- err = NULL;
+ connect_pending = false;
+
+ if (is_error)
+ goto failed;
+
if (!bt_io_get(device.sco, &err, BT_IO_OPT_MTU, &rsp.mtu,
BT_IO_OPT_INVALID)) {
error("Unable to get MTU: %s\n", err->message);
@@ -2610,6 +2611,24 @@ failed:
ipc_send_rsp(sco_ipc, SCO_SERVICE_ID, SCO_OP_STATUS, SCO_STATUS_FAILED);
}
+static void bt_sco_connect(const void *buf, uint16_t len)
+{
+ DBG("");
+
+ connect_pending = true;
+
+ if (!device.sco) {
+ DBG("SCO is not established, connecting...");
+ if (!connect_audio()) {
+ DBG("Connect audio failed");
+ return bt_sco_connect_rsp(true);
+ } else
+ return;
+ }
+
+ return bt_sco_connect_rsp(false);
+}
+
static const struct ipc_handler sco_handlers[] = {
/* SCO_OP_CONNECT */
{ bt_sco_connect, false, 0 }
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/hal-sco.c | 6 +++++-
1 file changed, 5 insertions(+), 1 deletion(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index b4667e5..fc12b50 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -545,7 +545,11 @@ static int out_set_format(struct audio_stream *stream, audio_format_t format)
static int out_standby(struct audio_stream *stream)
{
- DBG("");
+ DBG("stream %p", stream);
+
+ pthread_mutex_lock(&sco_mutex);
+ sco_close_socket();
+ pthread_mutex_unlock(&sco_mutex);
return 0;
}
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/hal-sco.c | 7 ++++++-
1 file changed, 6 insertions(+), 1 deletion(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index 2fa8fb7..06e0261 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -1038,7 +1038,12 @@ static int sco_open_input_stream(struct audio_hw_device *dev,
int chan_num, ret;
size_t resample_size;
- DBG("");
+ DBG("config %p device flags 0x%02x", config, devices);
+
+ if (ipc_connect_sco() != SCO_STATUS_SUCCESS) {
+ error("sco: cannot get fd");
+ return -EIO;
+ }
in = calloc(1, sizeof(struct sco_stream_in));
if (!in)
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/client/if-sco.c | 28 +++++++++++++++++++++++++++-
1 file changed, 27 insertions(+), 1 deletion(-)
diff --git a/android/client/if-sco.c b/android/client/if-sco.c
index 35ed3b0..b7377f4 100644
--- a/android/client/if-sco.c
+++ b/android/client/if-sco.c
@@ -174,10 +174,24 @@ static void prepare_sample(void)
sample_pos = 0;
}
+static void mono_to_stereo_pcm16(const int16_t *in, int16_t *out, size_t samples)
+{
+ int16_t mono;
+ size_t i;
+
+ for (i = 0; i < samples; i++) {
+ mono = get_unaligned(&in[i]);
+
+ put_unaligned(mono, &out[2 * i]);
+ put_unaligned(mono, &out[2 * i + 1]);
+ }
+}
+
static void *playback_thread(void *data)
{
int (*filbuff_cb) (short*, void*);
short buffer[buffer_size / sizeof(short)];
+ short buffer_in[buffer_size_in / sizeof(short)];
size_t len = 0;
ssize_t w_len = 0;
FILE *in = data;
@@ -217,7 +231,19 @@ static void *playback_thread(void *data)
pthread_mutex_unlock(&state_mutex);
- len = filbuff_cb(buffer, cb_data);
+ if (data == stream_in) {
+ int chan_in = popcount(stream_in->common.get_channels(&stream_in->common));
+ int chan_out = popcount(stream_out->common.get_channels(&stream_out->common));
+
+ len = filbuff_cb(buffer_in, cb_data);
+
+ if (chan_in == 1 && chan_out == 2) {
+ mono_to_stereo_pcm16(buffer_in,
+ buffer,
+ buffer_size_in / 2);
+ }
+ } else
+ len = filbuff_cb(buffer, cb_data);
pthread_mutex_lock(&outstream_mutex);
if (!stream_out) {
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
Opening input/output audio streams makes use of config with sample rate.
---
android/client/if-sco.c | 40 ++++++++++++++++++++++++++++++++++------
1 file changed, 34 insertions(+), 6 deletions(-)
diff --git a/android/client/if-sco.c b/android/client/if-sco.c
index b5fa926..555c6f9 100644
--- a/android/client/if-sco.c
+++ b/android/client/if-sco.c
@@ -442,10 +442,21 @@ static void stop_p(int argc, const char **argv)
static void open_output_stream_p(int argc, const char **argv)
{
+ struct audio_config *config;
int err;
RETURN_IF_NULL(if_audio_sco);
+ if (argc < 3) {
+ haltest_info("No sampling rate specified. Use default conf\n");
+ config = NULL;
+ } else {
+ config = calloc(1, sizeof(struct audio_config));
+ config->sample_rate = atoi(argv[2]);
+ config->channel_mask = AUDIO_CHANNEL_OUT_STEREO;
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ }
+
pthread_mutex_lock(&state_mutex);
if (current_state == STATE_PLAYING) {
haltest_error("Already playing!\n");
@@ -458,11 +469,11 @@ static void open_output_stream_p(int argc, const char **argv)
0,
AUDIO_DEVICE_OUT_ALL_SCO,
AUDIO_OUTPUT_FLAG_NONE,
- NULL,
+ config,
&stream_out);
if (err < 0) {
haltest_error("open output stream returned %d\n", err);
- return;
+ goto failed;
}
buffer_size = stream_out->common.get_buffer_size(&stream_out->common);
@@ -470,6 +481,9 @@ static void open_output_stream_p(int argc, const char **argv)
haltest_error("Invalid buffer size received!\n");
else
haltest_info("Using buffer size: %zu\n", buffer_size);
+failed:
+ if (config)
+ free(config);
}
static void close_output_stream_p(int argc, const char **argv)
@@ -490,10 +504,21 @@ static void close_output_stream_p(int argc, const char **argv)
static void open_input_stream_p(int argc, const char **argv)
{
+ struct audio_config *config;
int err;
RETURN_IF_NULL(if_audio_sco);
+ if (argc < 3) {
+ haltest_info("No sampling rate specified. Use default conf\n");
+ config = NULL;
+ } else {
+ config = calloc(1, sizeof(struct audio_config));
+ config->sample_rate = atoi(argv[2]);
+ config->channel_mask = AUDIO_CHANNEL_OUT_MONO;
+ config->format = AUDIO_FORMAT_PCM_16_BIT;
+ }
+
pthread_mutex_lock(&state_mutex);
if (current_state == STATE_PLAYING) {
haltest_error("Already playing!\n");
@@ -505,11 +530,11 @@ static void open_input_stream_p(int argc, const char **argv)
err = if_audio_sco->open_input_stream(if_audio_sco,
0,
AUDIO_DEVICE_IN_BLUETOOTH_SCO_HEADSET,
- NULL,
+ config,
&stream_in);
if (err < 0) {
haltest_error("open output stream returned %d\n", err);
- return;
+ goto failed;
}
buffer_size_in = stream_in->common.get_buffer_size(&stream_in->common);
@@ -517,6 +542,9 @@ static void open_input_stream_p(int argc, const char **argv)
haltest_error("Invalid buffer size received!\n");
else
haltest_info("Using buffer size: %zu\n", buffer_size_in);
+failed:
+ if (config)
+ free(config);
}
static void close_input_stream_p(int argc, const char **argv)
@@ -693,8 +721,8 @@ static struct method methods[] = {
STD_METHOD(init),
STD_METHOD(cleanup),
STD_METHOD(open_output_stream),
- STD_METHOD(close_output_stream),
- STD_METHOD(open_input_stream),
+ STD_METHODH(close_output_stream, "sample_rate"),
+ STD_METHODH(open_input_stream, "sampling rate"),
STD_METHOD(close_input_stream),
STD_METHODH(play, "<path to pcm file>"),
STD_METHOD(read),
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/hal-sco.c | 5 ++++-
1 file changed, 4 insertions(+), 1 deletion(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index 06e0261..e3d5242 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -659,6 +659,9 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
return -ENOMEM;
}
+ if (out->cfg.rate == AUDIO_STREAM_SCO_RATE)
+ goto skip_resampler;
+
/* Channel numbers for resampler */
chan_num = 1;
@@ -691,7 +694,7 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
DBG("Resampler: input %d output %d chan %d frames %u size %zd",
out->cfg.rate, AUDIO_STREAM_SCO_RATE, chan_num,
out->resample_frame_num, resample_size);
-
+skip_resampler:
*stream_out = &out->stream;
adev->out = out;
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
loop command makes stream_in->read() and then stream_out->write(). At
the moment buffers shall be equal.
---
android/client/if-sco.c | 45 ++++++++++++++++++++++++++++++++++++++++++---
1 file changed, 42 insertions(+), 3 deletions(-)
diff --git a/android/client/if-sco.c b/android/client/if-sco.c
index b4c3506..39c3e3a 100644
--- a/android/client/if-sco.c
+++ b/android/client/if-sco.c
@@ -148,6 +148,11 @@ static int feed_from_generator(short *buffer, void *data)
return buffer_size;
}
+static int feed_from_in(short *buffer, void *data)
+{
+ return stream_in->read(stream_in, buffer, buffer_size_in);
+}
+
static void prepare_sample(void)
{
int x;
@@ -179,8 +184,12 @@ static void *playback_thread(void *data)
/* Use file or fall back to generator */
if (in) {
- filbuff_cb = feed_from_file;
- cb_data = in;
+ if (data == stream_in)
+ filbuff_cb = feed_from_in;
+ else {
+ filbuff_cb = feed_from_file;
+ cb_data = in;
+ }
} else {
prepare_sample();
filbuff_cb = feed_from_generator;
@@ -218,7 +227,7 @@ static void *playback_thread(void *data)
pthread_mutex_unlock(&outstream_mutex);
} while (len && w_len > 0);
- if (in)
+ if (in && data != stream_in)
fclose(in);
pthread_mutex_lock(&state_mutex);
@@ -315,6 +324,35 @@ fail:
fclose(in);
}
+static void loop_p(int argc, const char **argv)
+{
+ RETURN_IF_NULL(if_audio_sco);
+ RETURN_IF_NULL(stream_out);
+ RETURN_IF_NULL(stream_in);
+
+ if (!buffer_size || !buffer_size_in) {
+ haltest_error("Invalid buffer sizes. Streams opened\n");
+ return;
+ }
+
+ if (buffer_size != buffer_size_in) {
+ haltest_error("read/write buffers differ, not supported\n");
+ return;
+ }
+
+ pthread_mutex_lock(&state_mutex);
+ if (current_state != STATE_STOPPED) {
+ haltest_error("Already playing or stream suspended!\n");
+ pthread_mutex_unlock(&state_mutex);
+ return;
+ }
+ pthread_mutex_unlock(&state_mutex);
+
+ if (pthread_create(&play_thread, NULL, playback_thread,
+ stream_in) != 0)
+ haltest_error("Cannot create playback thread!\n");
+}
+
static void read_p(int argc, const char **argv)
{
RETURN_IF_NULL(if_audio_sco);
@@ -615,6 +653,7 @@ static struct method methods[] = {
STD_METHOD(close_input_stream),
STD_METHODH(play, "<path to pcm file>"),
STD_METHOD(read),
+ STD_METHOD(loop),
STD_METHOD(stop),
STD_METHOD(suspend),
STD_METHOD(resume),
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
Read command makes stream_in->read() call to Audio HAL.
---
android/client/if-sco.c | 64 +++++++++++++++++++++++++++++++++++++++++++++++++
1 file changed, 64 insertions(+)
diff --git a/android/client/if-sco.c b/android/client/if-sco.c
index c71d0b4..b4c3506 100644
--- a/android/client/if-sco.c
+++ b/android/client/if-sco.c
@@ -230,6 +230,45 @@ static void *playback_thread(void *data)
return NULL;
}
+static void *read_thread(void *data)
+{
+ int (*filbuff_cb) (short*, void*) = feed_from_in;
+ short buffer[buffer_size / sizeof(short)];
+ size_t len = 0;
+ void *cb_data = NULL;
+
+ pthread_mutex_lock(&state_mutex);
+ current_state = STATE_PLAYING;
+ pthread_mutex_unlock(&state_mutex);
+
+ do {
+ pthread_mutex_lock(&state_mutex);
+
+ if (current_state == STATE_STOPPING) {
+ haltest_info("Detected stopping\n");
+ pthread_mutex_unlock(&state_mutex);
+ break;
+ } else if (current_state == STATE_SUSPENDED) {
+ pthread_mutex_unlock(&state_mutex);
+ usleep(500);
+ continue;
+ }
+
+ pthread_mutex_unlock(&state_mutex);
+
+ len = filbuff_cb(buffer, cb_data);
+ haltest_info("len %zd\n", len);
+ } while (len);
+
+ pthread_mutex_lock(&state_mutex);
+ current_state = STATE_STOPPED;
+ pthread_mutex_unlock(&state_mutex);
+
+ haltest_info("Done reading.\n");
+
+ return NULL;
+}
+
static void play_p(int argc, const char **argv)
{
const char *fname = NULL;
@@ -276,6 +315,30 @@ fail:
fclose(in);
}
+static void read_p(int argc, const char **argv)
+{
+ RETURN_IF_NULL(if_audio_sco);
+ RETURN_IF_NULL(stream_in);
+
+ if (!buffer_size_in) {
+ haltest_error("Invalid buffer sizes. Streams opened\n");
+ return;
+ }
+
+ pthread_mutex_lock(&state_mutex);
+ if (current_state != STATE_STOPPED) {
+ haltest_error("Already playing or stream suspended!\n");
+ pthread_mutex_unlock(&state_mutex);
+ return;
+ }
+ pthread_mutex_unlock(&state_mutex);
+
+ if (pthread_create(&play_thread, NULL, read_thread,
+ stream_in) != 0)
+ haltest_error("Cannot create playback thread!\n");
+
+}
+
static void stop_p(int argc, const char **argv)
{
pthread_mutex_lock(&state_mutex);
@@ -551,6 +614,7 @@ static struct method methods[] = {
STD_METHOD(open_input_stream),
STD_METHOD(close_input_stream),
STD_METHODH(play, "<path to pcm file>"),
+ STD_METHOD(read),
STD_METHOD(stop),
STD_METHOD(suspend),
STD_METHOD(resume),
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
---
android/hal-sco.c | 6 ++++--
1 file changed, 4 insertions(+), 2 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index 701d15e..ecf7a09 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -300,7 +300,7 @@ static bool write_data(struct sco_stream_out *out, const uint8_t *buffer,
struct pollfd pfd;
size_t len, written = 0;
int ret;
- uint16_t mtu = /* out->cfg.mtu */ 48;
+ uint16_t mtu = out->cfg.mtu;
uint64_t audio_sent_us, audio_passed_us;
pfd.fd = out->fd;
@@ -594,7 +594,9 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
out->cfg.channels = AUDIO_CHANNEL_OUT_STEREO;
out->cfg.rate = AUDIO_STREAM_DEFAULT_RATE;
out->cfg.frame_num = OUT_STREAM_FRAMES;
- out->cfg.mtu = mtu;
+
+ /* we get wrong mtu size for some reason */
+ out->cfg.mtu = /* mtu */ 48;
out->downmix_buf = malloc(out_get_buffer_size(&out->stream.common));
if (!out->downmix_buf) {
--
1.8.3.2
From: Andrei Emeltchenko <[email protected]>
SCO cached is used when Android writes with packet sizes which cannot
fit to 48 bytes SCO frames. Remaining frames are cached and written next
time Android perform out->write().
---
android/hal-sco.c | 38 +++++++++++++++++++++++++++++++++++---
1 file changed, 35 insertions(+), 3 deletions(-)
diff --git a/android/hal-sco.c b/android/hal-sco.c
index ecf7a09..f3ccb0d 100644
--- a/android/hal-sco.c
+++ b/android/hal-sco.c
@@ -64,6 +64,9 @@ struct sco_stream_out {
int fd;
uint8_t *downmix_buf;
+ uint8_t *cache;
+ size_t cache_len;
+
size_t samples;
struct timespec start;
@@ -301,6 +304,7 @@ static bool write_data(struct sco_stream_out *out, const uint8_t *buffer,
size_t len, written = 0;
int ret;
uint16_t mtu = out->cfg.mtu;
+ uint8_t *p;
uint64_t audio_sent_us, audio_passed_us;
pfd.fd = out->fd;
@@ -320,6 +324,7 @@ static bool write_data(struct sco_stream_out *out, const uint8_t *buffer,
return false;
}
+ len = bytes - written > mtu ? mtu : bytes - written;
clock_gettime(CLOCK_REALTIME, &now);
/* Mark start of the stream */
@@ -341,12 +346,30 @@ static bool write_data(struct sco_stream_out *out, const uint8_t *buffer,
memcpy(&out->start, &now, sizeof(out->start));
}
+ if (out->cache_len) {
+ DBG("First packet cache_len %zd", out->cache_len);
+ memcpy(out->cache + out->cache_len, buffer,
+ mtu - out->cache_len);
+ p = out->cache;
+ }
- len = bytes - written > mtu ? mtu : bytes - written;
+ if (bytes - written >= mtu)
+ p = (void *) buffer + written;
+ else {
+ memcpy(out->cache, buffer + written, bytes - written);
+ out->cache_len = bytes - written;
+ DBG("Last packet, cache %zd bytes", bytes - written);
+ written += bytes - written;
+ continue;
+ }
- ret = write(out->fd, buffer + written, len);
+ ret = write(out->fd, p, len);
if (ret > 0) {
- written += ret;
+ if (out->cache_len) {
+ written = mtu - out->cache_len;
+ out->cache_len = 0;
+ } else
+ written += ret;
out->samples += ret / 2;
@@ -604,6 +627,13 @@ static int sco_open_output_stream(struct audio_hw_device *dev,
return -ENOMEM;
}
+ out->cache = malloc(out->cfg.mtu);
+ if (!out->cache) {
+ free(out->downmix_buf);
+ free(out);
+ return -ENOMEM;
+ }
+
DBG("size %zd", out_get_buffer_size(&out->stream.common));
/* Channel numbers for resampler */
@@ -650,6 +680,7 @@ failed:
if (out->resampler)
release_resampler(out->resampler);
+ free(out->cache);
free(out->downmix_buf);
free(out);
stream_out = NULL;
@@ -674,6 +705,7 @@ static void sco_close_output_stream(struct audio_hw_device *dev,
if (out->resampler)
release_resampler(out->resampler);
+ free(out->cache);
free(out->downmix_buf);
free(out);
sco_dev->out = NULL;
--
1.8.3.2