virtual HRESULT RenderAudioSamples (bool preroll)
{
guint8 *ptr;
gint seg;
gint len;
gint bpf;
guint written, written_sum;
HRESULT res;
GST_LOG_OBJECT (m_ringbuffer->sink, "Writing audio samples (preroll: %d)",
preroll);
if (!gst_audio_ring_buffer_prepare_read (GST_AUDIO_RING_BUFFER_CAST
(m_ringbuffer), &seg, &ptr, &len)) {
GST_WARNING_OBJECT (m_ringbuffer->sink, "No segment available");
return E_FAIL;
}
bpf =
GST_AUDIO_INFO_BPF (&GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer)->
spec.info);
len /= bpf;
GST_LOG_OBJECT (m_ringbuffer->sink,
"Write audio samples: %p size %d segment: %d", ptr, len, seg);
written_sum = 0;
do {
res =
m_ringbuffer->output->output->ScheduleAudioSamples (ptr, len,
0, 0, &written);
len -= written;
ptr += written * bpf;
written_sum += written;
} while (len > 0 && res == S_OK);
GST_LOG_OBJECT (m_ringbuffer->sink, "Wrote %u samples: 0x%08x", written_sum,
res);
gst_audio_ring_buffer_clear (GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer),
seg);
gst_audio_ring_buffer_advance (GST_AUDIO_RING_BUFFER_CAST (m_ringbuffer),
1);
return res;
}
/* HALOutput AudioUnit will request fairly arbitrarily-sized chunks
* of data, not of a fixed size. So, we keep track of where in
* the current ringbuffer segment we are, and only advance the segment
* once we've read the whole thing */
static OSStatus
gst_osx_audio_sink_io_proc (GstOsxAudioRingBuffer * buf,
AudioUnitRenderActionFlags * ioActionFlags,
const AudioTimeStamp * inTimeStamp,
UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList * bufferList)
{
guint8 *readptr;
gint readseg;
gint len;
gint stream_idx = buf->core_audio->stream_idx;
gint remaining = bufferList->mBuffers[stream_idx].mDataByteSize;
gint offset = 0;
while (remaining) {
if (!gst_audio_ring_buffer_prepare_read (GST_AUDIO_RING_BUFFER (buf),
&readseg, &readptr, &len))
return 0;
len -= buf->segoffset;
if (len > remaining)
len = remaining;
memcpy ((char *) bufferList->mBuffers[stream_idx].mData + offset,
readptr + buf->segoffset, len);
buf->segoffset += len;
offset += len;
remaining -= len;
if ((gint) buf->segoffset == GST_AUDIO_RING_BUFFER (buf)->spec.segsize) {
/* clear written samples */
gst_audio_ring_buffer_clear (GST_AUDIO_RING_BUFFER (buf), readseg);
/* we wrote one segment */
gst_audio_ring_buffer_advance (GST_AUDIO_RING_BUFFER (buf), 1);
buf->segoffset = 0;
}
}
return 0;
}
/* this is the callback of jack. This should RT-safe.
*/
static int
jack_process_cb (jack_nframes_t nframes, void *arg)
{
GstJackAudioSink *sink;
GstAudioRingBuffer *buf;
gint readseg, len;
guint8 *readptr;
gint i, j, flen, channels;
sample_t *data;
buf = GST_AUDIO_RING_BUFFER_CAST (arg);
sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
/* get target buffers */
for (i = 0; i < channels; i++) {
sink->buffers[i] =
(sample_t *) jack_port_get_buffer (sink->ports[i], nframes);
}
if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
flen = len / channels;
/* the number of samples must be exactly the segment size */
if (nframes * sizeof (sample_t) != flen)
goto wrong_size;
GST_DEBUG_OBJECT (sink, "copy %d frames: %p, %d bytes, %d channels",
nframes, readptr, flen, channels);
data = (sample_t *) readptr;
/* the samples in the ringbuffer have the channels interleaved, we need to
* deinterleave into the jack target buffers */
for (i = 0; i < nframes; i++) {
for (j = 0; j < channels; j++) {
sink->buffers[j][i] = *data++;
}
}
/* clear written samples in the ringbuffer */
gst_audio_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_audio_ring_buffer_advance (buf, 1);
} else {
GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes);
/* We are not allowed to read from the ringbuffer, write silence to all
* jack output buffers */
for (i = 0; i < channels; i++) {
memset (sink->buffers[i], 0, nframes * sizeof (sample_t));
}
}
return 0;
/* ERRORS */
wrong_size:
{
GST_ERROR_OBJECT (sink, "nbytes (%d) != flen (%d)",
(gint) (nframes * sizeof (sample_t)), flen);
return 1;
}
}
/* this internal thread does nothing else but write samples to the audio device.
* It will write each segment in the ringbuffer and will update the play
* pointer.
* The start/stop methods control the thread.
*/
static void
audioringbuffer_thread_func (GstAudioRingBuffer * buf)
{
GstAudioSink *sink;
GstAudioSinkClass *csink;
GstAudioSinkRingBuffer *abuf = GST_AUDIO_SINK_RING_BUFFER_CAST (buf);
WriteFunc writefunc;
GstMessage *message;
GValue val = { 0 };
sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf));
csink = GST_AUDIO_SINK_GET_CLASS (sink);
GST_DEBUG_OBJECT (sink, "enter thread");
GST_OBJECT_LOCK (abuf);
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
GST_OBJECT_UNLOCK (abuf);
writefunc = csink->write;
if (writefunc == NULL)
goto no_function;
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_ENTER, GST_ELEMENT_CAST (sink));
g_value_init (&val, GST_TYPE_G_THREAD);
g_value_set_boxed (&val, g_thread_self ());
gst_message_set_stream_status_object (message, &val);
g_value_unset (&val);
GST_DEBUG_OBJECT (sink, "posting ENTER stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
while (TRUE) {
gint left, len;
guint8 *readptr;
gint readseg;
/* buffer must be started */
if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
gint written;
left = len;
do {
written = writefunc (sink, readptr, left);
GST_LOG_OBJECT (sink, "transfered %d bytes of %d from segment %d",
written, left, readseg);
if (written < 0 || written > left) {
/* might not be critical, it e.g. happens when aborting playback */
GST_WARNING_OBJECT (sink,
"error writing data in %s (reason: %s), skipping segment (left: %d, written: %d)",
GST_DEBUG_FUNCPTR_NAME (writefunc),
(errno > 1 ? g_strerror (errno) : "unknown"), left, written);
break;
}
left -= written;
readptr += written;
} while (left > 0);
/* clear written samples */
gst_audio_ring_buffer_clear (buf, readseg);
/* we wrote one segment */
gst_audio_ring_buffer_advance (buf, 1);
} else {
GST_OBJECT_LOCK (abuf);
if (!abuf->running)
goto stop_running;
if (G_UNLIKELY (g_atomic_int_get (&buf->state) ==
GST_AUDIO_RING_BUFFER_STATE_STARTED)) {
GST_OBJECT_UNLOCK (abuf);
continue;
}
GST_DEBUG_OBJECT (sink, "signal wait");
GST_AUDIO_SINK_RING_BUFFER_SIGNAL (buf);
GST_DEBUG_OBJECT (sink, "wait for action");
GST_AUDIO_SINK_RING_BUFFER_WAIT (buf);
GST_DEBUG_OBJECT (sink, "got signal");
if (!abuf->running)
goto stop_running;
GST_DEBUG_OBJECT (sink, "continue running");
GST_OBJECT_UNLOCK (abuf);
}
}
/* Will never be reached */
g_assert_not_reached ();
return;
/* ERROR */
no_function:
{
GST_DEBUG_OBJECT (sink, "no write function, exit thread");
return;
}
stop_running:
{
GST_OBJECT_UNLOCK (abuf);
GST_DEBUG_OBJECT (sink, "stop running, exit thread");
message = gst_message_new_stream_status (GST_OBJECT_CAST (buf),
GST_STREAM_STATUS_TYPE_LEAVE, GST_ELEMENT_CAST (sink));
g_value_init (&val, GST_TYPE_G_THREAD);
g_value_set_boxed (&val, g_thread_self ());
gst_message_set_stream_status_object (message, &val);
g_value_unset (&val);
GST_DEBUG_OBJECT (sink, "posting LEAVE stream status");
gst_element_post_message (GST_ELEMENT_CAST (sink), message);
return;
}
}