static size_t in_get_buffer_size(const struct audio_stream *stream)
{
const struct submix_stream_in *in = reinterpret_cast<const struct submix_stream_in *>(stream);
ALOGV("in_get_buffer_size() returns %u",
in->dev->config.period_size * audio_stream_frame_size(stream));
return in->dev->config.period_size * audio_stream_frame_size(stream);
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
struct stream_out *out = (struct stream_out *)stream;
size_t buf_size;
buf_size = out->pcm_config.period_size *
audio_stream_frame_size((struct audio_stream *)stream);
ALOGV("%s : %d, period_size : %d, frame_size : %d",
__func__,
buf_size,
out->pcm_config.period_size,
audio_stream_frame_size((struct audio_stream *)stream));
return buf_size;
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
int ret;
struct stream_out *out = (struct stream_out *)stream;
pthread_mutex_lock(&out->dev->lock);
pthread_mutex_lock(&out->lock);
if (out->standby) {
ret = start_output_stream(out);
if (ret != 0) {
goto err;
}
out->standby = false;
}
pcm_write(out->pcm, (void *)buffer, bytes);
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&out->dev->lock);
return bytes;
err:
pthread_mutex_unlock(&out->lock);
if (ret != 0) {
usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) /
out_get_sample_rate(&stream->common));
}
return bytes;
}
static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
size_t bytes)
{
/* XXX: fake timing for audio input */
usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) /
in_get_sample_rate(&stream->common));
return bytes;
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
/* XXX: fake timing for audio output */
usleep(bytes * 1000000 / audio_stream_frame_size(&stream->common) /
out_get_sample_rate(&stream->common));
return bytes;
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
int ret = 0;
struct stream_out *out = (struct stream_out *)stream;
struct audio_device *adev = out ? out->dev : NULL;
if(adev == NULL) return -ENOSYS;
ALOGV("%s enter",__func__);
pthread_mutex_lock(&out->dev->lock);
pthread_mutex_lock(&out->lock);
// there is a possibility that the HD interface is open
// and normal pcm stream is still active. Feed the new
// interface to normal pcm stream
if(adev->active_pcm) {
if(adev->active_pcm != out->pcm)
out->pcm = adev->active_pcm;
}
if ((out->standby) || (!adev->active_pcm)) {
ret = start_output_stream(out);
if (ret != 0) {
goto err;
}
out->standby = false;
}
if(!out->pcm){
ALOGD("%s: null handle to write - device already closed",__func__);
goto err;
}
ret = pcm_write(out->pcm, (void *)buffer, bytes);
ALOGVV("%s: pcm_write returned = %d rate = %d",__func__,ret,out->pcm_config.rate);
err:
pthread_mutex_unlock(&out->lock);
pthread_mutex_unlock(&out->dev->lock);
if (ret != 0) {
uint64_t duration_ms = ((bytes * 1000)/
(audio_stream_frame_size(&stream->common)) /
(out_get_sample_rate(&stream->common)));
ALOGV("%s : silence written", __func__);
usleep(duration_ms * 1000);
}
ALOGV("%s exit",__func__);
return bytes;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
struct sco_stream_out *out = (struct sco_stream_out *) stream;
size_t size = audio_stream_frame_size(&out->stream.common) *
out->cfg.frame_num;
DBG("buf size %zd", size);
return size;
}
/* Returns bytes for ONE PERIOD */
size_t hdmi_out_get_buffer_size(const struct audio_stream *stream)
{
hdmi_out_t *out = (hdmi_out_t*)stream;
struct pcm_config *config = &out->config;
size_t ans;
ans = audio_stream_frame_size((struct audio_stream*)stream) * config->period_size;
TRACEM("stream=%p returning %u", stream, ans);
return ans;
}
static int adev_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 audio_device *adev = (struct audio_device *)dev;
struct stream_in *in;
int ret, buffer_size, frame_size;
int channel_count = popcount(config->channel_mask);
ALOGV("%s: enter", __func__);
*stream_in = NULL;
if (check_input_parameters(config->sample_rate, config->format, channel_count) != 0)
return -EINVAL;
in = (struct stream_in *)calloc(1, sizeof(struct 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.standby = in_standby;
in->stream.common.set_parameters = in_set_parameters;
in->stream.common.get_parameters = in_get_parameters;
in->stream.read = in_read;
in->stream.get_input_frames_lost = in_get_input_frames_lost;
in->device = devices;
in->source = AUDIO_SOURCE_DEFAULT;
in->dev = adev;
in->standby = true;
in->channel_mask = config->channel_mask;
/* Update config params with the requested sample rate and channels */
in->pcm_config = pcm_config_audio_capture;
in->pcm_config.channels = channel_count;
in->pcm_config.rate = config->sample_rate;
frame_size = audio_stream_frame_size((struct audio_stream *)in);
buffer_size = get_input_buffer_size(config->sample_rate,
config->format,
channel_count);
in->pcm_config.period_size = buffer_size / frame_size;
*stream_in = &in->stream;
ALOGV("%s: exit", __func__);
return 0;
}
static ssize_t out_write(struct audio_stream_out *stream, const void *buffer,
size_t bytes)
{
struct sco_stream_out *out = (struct sco_stream_out *) stream;
size_t frame_num = bytes / audio_stream_frame_size(&out->stream.common);
size_t output_frame_num = frame_num;
void *send_buf = out->downmix_buf;
size_t total;
DBG("write to fd %d bytes %zu", out->fd, bytes);
if (!out->downmix_buf) {
error("sco: downmix buffer not initialized");
return -1;
}
downmix_to_mono(out, buffer, frame_num);
if (out->resampler) {
int ret;
/* limit resampler's output within what resample buf can hold */
output_frame_num = out->resample_frame_num;
ret = out->resampler->resample_from_input(out->resampler,
send_buf,
&frame_num,
out->resample_buf,
&output_frame_num);
if (ret) {
error("Failed to resample frames: %zd input %zd (%s)",
frame_num, output_frame_num, strerror(ret));
return -1;
}
send_buf = out->resample_buf;
DBG("Resampled: frame_num %zd, output_frame_num %zd",
frame_num, output_frame_num);
}
total = output_frame_num * sizeof(int16_t) * 1;
DBG("total %zd", total);
if (!write_data(out, send_buf, total))
return -1;
return bytes;
}
static ssize_t in_read(struct audio_stream_in *stream, void *buffer,
size_t bytes)
{
struct stream_in *in = (struct stream_in *)stream;
struct audio_device *adev = in->dev;
int i, ret = -1;
ALOGV("%s enter",__func__);
pthread_mutex_lock(&in->lock);
if (in->standby) {
pthread_mutex_lock(&adev->lock);
ret = start_input_stream(in);
pthread_mutex_unlock(&adev->lock);
if (ret != 0) {
goto exit;
}
in->standby = false;
}
if (in->pcm) {
ret = pcm_read(in->pcm, buffer, bytes);
}
ALOGV("in_read returned %d bytes ret = %d",bytes,ret);
exit:
pthread_mutex_unlock(&in->lock);
if (ret != 0) {
in_standby(&in->stream.common);
uint64_t duration_ms = ((bytes * 1000)/
(audio_stream_frame_size(&in->stream.common)) /
(in_get_sample_rate(&in->stream.common)));
ALOGV("%s : silence read - read failed", __func__);
usleep(duration_ms * 1000);
}
ALOGV("%s exit",__func__);
return bytes;
}
void channel_remap(struct audio_stream_out *stream, const void *buffer,
size_t bytes)
{
hdmi_out_t *out = (hdmi_out_t*)stream;
struct hdmi_device_t *adev = (struct hdmi_device_t *)out->dev;
int x, y, frames;
int16_t *buf = (int16_t *)buffer;
int16_t *tmp_buf = (int16_t *)out->buffcpy;
frames = (bytes/audio_stream_frame_size(&out->stream_out.common));
while (frames--){
for(y = 0; y < (int)out->config.channels; y++){
for(x = 0; x < (int)out->config.channels; x++){
if (cea_channel_map[y] == adev->map[x]){
tmp_buf[y] = buf[x];
break;
}
}
}
tmp_buf += (int)out->config.channels;
buf += (int)out->config.channels;
}
}
ssize_t hdmi_out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
hdmi_out_t *out = (hdmi_out_t*)stream;
struct hdmi_device_t *adev = (struct hdmi_device_t *)out->dev;
ssize_t ret;
TRACEM("stream=%p buffer=%p bytes=%d", stream, buffer, bytes);
if (!out->up) {
if(hdmi_out_open_pcm(out)) {
ret = -ENOSYS;
goto exit;
}
}
if (out->config.channels > 2 && !adev->CEAMap){
channel_remap(stream, buffer, bytes);
ret = pcm_write(out->pcm, out->buffcpy, bytes);
} else {
ret = pcm_write(out->pcm, buffer, bytes);
}
exit:
if (ret != 0) {
ALOGE("Error writing to HDMI pcm: %s", pcm_get_error(out->pcm));
hdmi_out_standby((struct audio_stream*)stream);
unsigned int usecs = bytes * 1000000 /
audio_stream_frame_size((struct audio_stream*)stream) /
hdmi_out_get_sample_rate((struct audio_stream*)stream);
if (usecs >= 1000000L) {
usecs = 999999L;
}
usleep(usecs);
}
return bytes;
}
ssize_t InStream::read(void* buffer, size_t bytes) {
int ret = 0;
LOGFUNC("%s(%p, %p, %d)", __func__, this, buffer, bytes);
AutoMutex lock(mLock);
if (mStandby) {
if (startInputStream())
return -EBUSY;
}
ret = pcm_read(mPcm, buffer, bytes);
ALOGV("pcm_read(%p, %p, %d) returned %d", mPcm, buffer, bytes, ret);
if (ret >= 0 && mDev.mMicMute)
memset(buffer, 0, bytes);
if (ret < 0) {
ALOGE("pcm_read(%p, %p, %d) returned %d", mPcm, buffer, bytes, ret);
usleep(bytes * 1000000 /
audio_stream_frame_size(&audio_stream_in()->common) /
mConfig.rate);
}
return bytes;
}
static size_t out_get_buffer_size(const struct audio_stream *stream)
{
return pcm_config.period_size *
audio_stream_frame_size((struct audio_stream *)stream);
}
static ssize_t in_read(struct audio_stream_in *stream, void* buffer,
size_t bytes)
{
//ALOGV("in_read bytes=%u", bytes);
ssize_t frames_read = -1977;
struct submix_stream_in *in = reinterpret_cast<struct submix_stream_in *>(stream);
const size_t frame_size = audio_stream_frame_size(&stream->common);
const size_t frames_to_read = bytes / frame_size;
pthread_mutex_lock(&in->dev->lock);
const bool output_standby_transition = (in->output_standby != in->dev->output_standby);
in->output_standby = in->dev->output_standby;
if (in->dev->input_standby || output_standby_transition) {
in->dev->input_standby = false;
// keep track of when we exit input standby (== first read == start "real recording")
// or when we start recording silence, and reset projected time
int rc = clock_gettime(CLOCK_MONOTONIC, &in->record_start_time);
if (rc == 0) {
in->read_counter_frames = 0;
}
}
in->read_counter_frames += frames_to_read;
size_t remaining_frames = frames_to_read;
{
// about to read from audio source
sp<MonoPipeReader> source = in->dev->rsxSource.get();
if (source == 0) {
ALOGE("no audio pipe yet we're trying to read!");
pthread_mutex_unlock(&in->dev->lock);
usleep((bytes / frame_size) * 1000000 / in_get_sample_rate(&stream->common));
memset(buffer, 0, bytes);
return bytes;
}
pthread_mutex_unlock(&in->dev->lock);
// read the data from the pipe (it's non blocking)
int attempts = 0;
char* buff = (char*)buffer;
while ((remaining_frames > 0) && (attempts < MAX_READ_ATTEMPTS)) {
attempts++;
frames_read = source->read(buff, remaining_frames, AudioBufferProvider::kInvalidPTS);
if (frames_read > 0) {
remaining_frames -= frames_read;
buff += frames_read * frame_size;
//ALOGV(" in_read (att=%d) got %ld frames, remaining=%u",
// attempts, frames_read, remaining_frames);
} else {
//ALOGE(" in_read read returned %ld", frames_read);
usleep(READ_ATTEMPT_SLEEP_MS * 1000);
}
}
// done using the source
pthread_mutex_lock(&in->dev->lock);
source.clear();
pthread_mutex_unlock(&in->dev->lock);
}
if (remaining_frames > 0) {
ALOGV(" remaining_frames = %d", remaining_frames);
memset(((char*)buffer)+ bytes - (remaining_frames * frame_size), 0,
remaining_frames * frame_size);
}
// compute how much we need to sleep after reading the data by comparing the wall clock with
// the projected time at which we should return.
struct timespec time_after_read;// wall clock after reading from the pipe
struct timespec record_duration;// observed record duration
int rc = clock_gettime(CLOCK_MONOTONIC, &time_after_read);
const uint32_t sample_rate = in_get_sample_rate(&stream->common);
if (rc == 0) {
// for how long have we been recording?
record_duration.tv_sec = time_after_read.tv_sec - in->record_start_time.tv_sec;
record_duration.tv_nsec = time_after_read.tv_nsec - in->record_start_time.tv_nsec;
if (record_duration.tv_nsec < 0) {
record_duration.tv_sec--;
record_duration.tv_nsec += 1000000000;
}
// read_counter_frames contains the number of frames that have been read since the beginning
// of recording (including this call): it's converted to usec and compared to how long we've
// been recording for, which gives us how long we must wait to sync the projected recording
// time, and the observed recording time
long projected_vs_observed_offset_us =
((int64_t)(in->read_counter_frames
- (record_duration.tv_sec*sample_rate)))
* 1000000 / sample_rate
- (record_duration.tv_nsec / 1000);
ALOGV(" record duration %5lds %3ldms, will wait: %7ldus",
record_duration.tv_sec, record_duration.tv_nsec/1000000,
projected_vs_observed_offset_us);
if (projected_vs_observed_offset_us > 0) {
usleep(projected_vs_observed_offset_us);
}
}
ALOGV("in_read returns %d", bytes);
return bytes;
}
static size_t in_get_buffer_size(const struct audio_stream *stream)
{
struct stream_in *in = (struct stream_in *)stream;
return in->pcm_config.period_size * audio_stream_frame_size(stream);
}
static ssize_t out_write(struct audio_stream_out *stream, const void* buffer,
size_t bytes)
{
//ALOGV("out_write(bytes=%d)", bytes);
ssize_t written_frames = 0;
struct submix_stream_out *out = reinterpret_cast<struct submix_stream_out *>(stream);
const size_t frame_size = audio_stream_frame_size(&stream->common);
const size_t frames = bytes / frame_size;
pthread_mutex_lock(&out->dev->lock);
out->dev->output_standby = false;
sp<MonoPipe> sink = out->dev->rsxSink.get();
if (sink != 0) {
if (sink->isShutdown()) {
sink.clear();
pthread_mutex_unlock(&out->dev->lock);
// the pipe has already been shutdown, this buffer will be lost but we must
// simulate timing so we don't drain the output faster than realtime
usleep(frames * 1000000 / out_get_sample_rate(&stream->common));
return bytes;
}
} else {
pthread_mutex_unlock(&out->dev->lock);
ALOGE("out_write without a pipe!");
ALOG_ASSERT("out_write without a pipe!");
return 0;
}
pthread_mutex_unlock(&out->dev->lock);
written_frames = sink->write(buffer, frames);
if (written_frames < 0) {
if (written_frames == (ssize_t)NEGOTIATE) {
ALOGE("out_write() write to pipe returned NEGOTIATE");
pthread_mutex_lock(&out->dev->lock);
sink.clear();
pthread_mutex_unlock(&out->dev->lock);
written_frames = 0;
return 0;
} else {
// write() returned UNDERRUN or WOULD_BLOCK, retry
ALOGE("out_write() write to pipe returned unexpected %16lx", written_frames);
written_frames = sink->write(buffer, frames);
}
}
pthread_mutex_lock(&out->dev->lock);
sink.clear();
pthread_mutex_unlock(&out->dev->lock);
if (written_frames < 0) {
ALOGE("out_write() failed writing to pipe with %16lx", written_frames);
return 0;
} else {
ALOGV("out_write() wrote %lu bytes)", written_frames * frame_size);
return written_frames * frame_size;
}
}
static int adev_open_output_stream(struct audio_hw_device *dev,
uint32_t devices, int *format,
uint32_t *channels, uint32_t *sample_rate,
struct audio_stream_out **stream_out)
{
struct adev_a2dp *adev = (struct adev_a2dp *)dev;
struct astream_out *out;
int ret;
pthread_mutex_lock(&adev->lock);
/* one output stream at a time */
if (adev->output) {
LOGV("output exists");
ret = -EBUSY;
goto err_output_exists;
}
out = calloc(1, sizeof(struct astream_out));
if (!out) {
ret = -ENOMEM;
goto err_alloc;
}
pthread_mutex_init(&out->lock, NULL);
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;
out->stream.common.get_channels = out_get_channels;
out->stream.common.get_format = out_get_format;
out->stream.common.set_format = out_set_format;
out->stream.common.standby = out_standby;
out->stream.common.dump = out_dump;
out->stream.common.set_parameters = out_set_parameters;
out->stream.common.get_parameters = out_get_parameters;
out->stream.common.set_device = out_set_device;
out->stream.common.get_device = out_get_device;
out->stream.common.add_audio_effect = out_add_audio_effect;
out->stream.common.remove_audio_effect = out_remove_audio_effect;
out->stream.get_latency = out_get_latency;
out->stream.set_volume = out_set_volume;
out->stream.write = out_write;
out->stream.get_render_position = out_get_render_position;
out->sample_rate = 44100;
out->buffer_size = 512 * 20;
out->channels = AUDIO_CHANNEL_OUT_STEREO;
out->format = AUDIO_FORMAT_PCM_16_BIT;
out->fd = -1;
out->device = devices;
out->bt_enabled = adev->bt_enabled;
out->suspended = adev->suspended;
/* for now, buffer_duration_us is precalculated and never changed.
* if the sample rate or the format ever changes on the fly, we'd have
* to recalculate this */
out->buffer_duration_us = ((out->buffer_size * 1000 ) /
audio_stream_frame_size(&out->stream.common) /
out->sample_rate) * 1000;
if (!_out_validate_parms(out, format ? *format : 0,
channels ? *channels : 0,
sample_rate ? *sample_rate : 0)) {
LOGV("invalid parameters");
ret = -EINVAL;
goto err_validate_parms;
}
int err = pthread_create(&out->buf_thread, (const pthread_attr_t *) NULL, _out_buf_thread_func, out);
if (err != 0) {
goto err_validate_parms;
}
/* PCM format is always 16bit, stereo */
out->buf_size = (out->buffer_size * BUF_NUM_PERIODS) / sizeof(int32_t);
out->buf = (uint32_t *)malloc(out->buf_size * sizeof(int32_t));
if (!out->buf) {
goto err_validate_parms;
}
/* XXX: check return code? */
if (adev->bt_enabled)
_out_init_locked(out, "00:00:00:00:00:00");
adev->output = out;
if (format)
*format = out->format;
if (channels)
*channels = out->channels;
if (sample_rate)
*sample_rate = out->sample_rate;
pthread_mutex_unlock(&adev->lock);
*stream_out = &out->stream;
return 0;
err_validate_parms:
free(out);
err_alloc:
err_output_exists:
pthread_mutex_unlock(&adev->lock);
*stream_out = NULL;
return ret;
}
