Commit 7d64e16b authored by Andoni Morales Alastruey's avatar Andoni Morales Alastruey Committed by Sebastian Dröge
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directsoundsink: add support for ac-3 over spdif

parent 94e54887
......@@ -55,6 +55,7 @@
#include <gst/base/gstbasesink.h>
#include <gst/audio/streamvolume.h>
#include "gstdirectsoundsink.h"
#include <gst/audio/gstaudioiec61937.h>
#include <math.h>
......@@ -79,6 +80,8 @@ static void gst_directsound_sink_get_property (GObject * object, guint prop_id,
static GstCaps *gst_directsound_sink_getcaps (GstBaseSink * bsink,
GstCaps * filter);
static GstBuffer *gst_directsound_sink_payload (GstAudioBaseSink * sink,
GstBuffer * buf);
static gboolean gst_directsound_sink_prepare (GstAudioSink * asink,
GstAudioRingBufferSpec * spec);
static gboolean gst_directsound_sink_unprepare (GstAudioSink * asink);
......@@ -90,6 +93,7 @@ static guint gst_directsound_sink_delay (GstAudioSink * asink);
static void gst_directsound_sink_reset (GstAudioSink * asink);
static GstCaps *gst_directsound_probe_supported_formats (GstDirectSoundSink *
dsoundsink, const GstCaps * template_caps);
static gboolean gst_directsound_sink_query (GstBaseSink * pad, GstQuery * query);
static void gst_directsound_sink_set_volume (GstDirectSoundSink * sink,
gdouble volume, gboolean store);
......@@ -110,7 +114,7 @@ static GstStaticPadTemplate directsoundsink_sink_factory =
"format = (string) S8, "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, 2 ];"
"audio/x-iec958"));
"audio/x-ac3, framed = (boolean) true;"));
enum
{
......@@ -136,6 +140,7 @@ gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstBaseSinkClass *gstbasesink_class = GST_BASE_SINK_CLASS (klass);
GstAudioSinkClass *gstaudiosink_class = GST_AUDIO_SINK_CLASS (klass);
GstAudioBaseSinkClass *gstaudiobasesink_class = GST_AUDIO_BASE_SINK_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GST_DEBUG_CATEGORY_INIT (directsoundsink_debug, "directsoundsink", 0,
......@@ -150,6 +155,12 @@ gst_directsound_sink_class_init (GstDirectSoundSinkClass * klass)
gstbasesink_class->get_caps =
GST_DEBUG_FUNCPTR (gst_directsound_sink_getcaps);
gstbasesink_class->query =
GST_DEBUG_FUNCPTR (gst_directsound_sink_query);
gstaudiobasesink_class->payload =
GST_DEBUG_FUNCPTR (gst_directsound_sink_payload);
gstaudiosink_class->prepare =
GST_DEBUG_FUNCPTR (gst_directsound_sink_prepare);
gstaudiosink_class->unprepare =
......@@ -274,6 +285,82 @@ gst_directsound_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)
return caps;
}
static gboolean
gst_directsound_sink_acceptcaps (GstBaseSink * sink, GstQuery * query)
{
GstDirectSoundSink *dsink = GST_DIRECTSOUND_SINK (sink);
GstAudioRingBuffer *rbuf = GST_AUDIO_BASE_SINK (dsink)->ringbuffer;
GstPad *pad;
GstCaps *caps;
GstCaps *pad_caps;
GstStructure *st;
gboolean ret = FALSE;
GstAudioRingBufferSpec spec = { 0 };
if (G_UNLIKELY (dsink == NULL))
return FALSE;
pad = sink->sinkpad;
gst_query_parse_accept_caps (query, &caps);
GST_DEBUG_OBJECT (pad, "caps %" GST_PTR_FORMAT, caps);
pad_caps = gst_pad_query_caps (pad, NULL);
if (pad_caps) {
ret = gst_caps_can_intersect (pad_caps, caps);
gst_caps_unref (pad_caps);
if (!ret)
goto done;
}
/* If we've not got fixed caps, creating a stream might fail, so let's just
* return from here with default acceptcaps behaviour */
if (!gst_caps_is_fixed (caps))
goto done;
if (!gst_audio_ring_buffer_parse_caps (&rbuf->spec, caps))
goto done;
/* Make sure input is framed (one frame per buffer) and can be payloaded */
switch (rbuf->spec.type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
{
gboolean framed = FALSE, parsed = FALSE;
st = gst_caps_get_structure (caps, 0);
gst_structure_get_boolean (st, "framed", &framed);
gst_structure_get_boolean (st, "parsed", &parsed);
if ((!framed && !parsed) || gst_audio_iec61937_frame_size (&spec) <= 0)
goto done;
}
default:{
}
}
ret = TRUE;
done:
gst_object_unref (dsink);
gst_query_set_accept_caps_result (query, ret);
return TRUE;
}
static gboolean
gst_directsound_sink_query (GstBaseSink * sink, GstQuery * query)
{
gboolean res = TRUE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_ACCEPT_CAPS:
res = gst_directsound_sink_acceptcaps (sink, query);
break;
default:
res = FALSE;
}
return res;
}
static gboolean
gst_directsound_sink_open (GstAudioSink * asink)
{
......@@ -301,6 +388,14 @@ gst_directsound_sink_open (GstAudioSink * asink)
return TRUE;
}
static boolean
gst_directsound_sink_is_spdif_format (GstDirectSoundSink * dsoundsink)
{
GstAudioRingBufferFormatType type;
type = GST_AUDIO_BASE_SINK (dsoundsink)->ringbuffer->spec.type;
return type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3;
}
static gboolean
gst_directsound_sink_prepare (GstAudioSink * asink,
GstAudioRingBufferSpec * spec)
......@@ -318,7 +413,7 @@ gst_directsound_sink_prepare (GstAudioSink * asink,
/* fill the WAVEFORMATEX structure with spec params */
memset (&wfx, 0, sizeof (wfx));
if (spec->type != GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958) {
if (gst_directsound_sink_is_spdif_format (dsoundsink)) {
wfx.cbSize = sizeof (wfx);
wfx.wFormatTag = WAVE_FORMAT_PCM;
wfx.nChannels = spec->info.channels;
......@@ -361,7 +456,7 @@ gst_directsound_sink_prepare (GstAudioSink * asink,
dsoundsink->buffer_size = spec->segsize * spec->segtotal;
GST_INFO_OBJECT (dsoundsink,
"GstRingBufferSpec->channels: %d, GstRingBufferSpec->rate: %d, GstRingBufferSpec->bytes_per_sample: %d\n"
"GstAudioRingBufferSpec->channels: %d, GstAudioRingBufferSpec->rate: %d, GstAudioRingBufferSpec->bytes_per_sample: %d\n"
"WAVEFORMATEX.nSamplesPerSec: %ld, WAVEFORMATEX.wBitsPerSample: %d, WAVEFORMATEX.nBlockAlign: %d, WAVEFORMATEX.nAvgBytesPerSec: %ld\n"
"Size of dsound circular buffer=>%d\n", spec->info.channels,
spec->info.rate, spec->info.bpf, wfx.nSamplesPerSec, wfx.wBitsPerSample,
......@@ -371,7 +466,7 @@ gst_directsound_sink_prepare (GstAudioSink * asink,
memset (&descSecondary, 0, sizeof (DSBUFFERDESC));
descSecondary.dwSize = sizeof (DSBUFFERDESC);
descSecondary.dwFlags = DSBCAPS_GETCURRENTPOSITION2 | DSBCAPS_GLOBALFOCUS;
if (spec->type != GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958)
if (!gst_directsound_sink_is_spdif_format (dsoundsink))
descSecondary.dwFlags |= DSBCAPS_CTRLVOLUME;
descSecondary.dwBufferBytes = dsoundsink->buffer_size;
......@@ -436,10 +531,6 @@ gst_directsound_sink_write (GstAudioSink * asink, gpointer data, guint length)
dsoundsink = GST_DIRECTSOUND_SINK (asink);
/* Fix endianness */
if (dsoundsink->type == GST_AUDIO_RING_BUFFER_FORMAT_TYPE_IEC958)
_swab (data, data, length);
GST_DSOUND_LOCK (dsoundsink);
/* get current buffer status */
......@@ -639,8 +730,7 @@ gst_directsound_probe_supported_formats (GstDirectSoundSink * dsoundsink,
GST_INFO_OBJECT (dsoundsink, "AC3 passthrough not supported "
"(IDirectSound_CreateSoundBuffer returned: %s)\n",
DXGetErrorString9 (hRes));
caps =
gst_caps_subtract (caps, gst_caps_new_empty_simple ("audio/x-iec958"));
caps = gst_caps_subtract (caps, gst_caps_new_empty_simple ("audio/x-ac3"));
} else {
GST_INFO_OBJECT (dsoundsink, "AC3 passthrough supported");
hRes = IDirectSoundBuffer_Release (dsoundsink->pDSBSecondary);
......@@ -651,12 +741,61 @@ gst_directsound_probe_supported_formats (GstDirectSoundSink * dsoundsink,
}
}
#else
caps = gst_caps_subtract (caps, gst_caps_new_simple ("audio/x-iec958", NULL));
caps = gst_caps_subtract (caps, gst_caps_new_simple ("audio/x-ac3", NULL));
#endif
return caps;
}
static GstBuffer *
gst_directsound_sink_payload (GstAudioBaseSink * sink, GstBuffer * buf)
{
switch (sink->ringbuffer->spec.type) {
case GST_AUDIO_RING_BUFFER_FORMAT_TYPE_AC3:
{
gint framesize = gst_audio_iec61937_frame_size (&sink->ringbuffer->spec);
GstBuffer *out;
GstMapInfo infobuf, infoout;
gboolean success;
if (framesize <= 0)
return NULL;
out = gst_buffer_new_and_alloc (framesize);
if (!gst_buffer_map (buf, &infobuf, GST_MAP_READWRITE))
{
gst_buffer_unref (out);
return NULL;
}
if (!gst_buffer_map (out, &infoout, GST_MAP_READWRITE))
{
gst_buffer_unmap (buf, &infobuf);
gst_buffer_unref (out);
return NULL;
}
success = gst_audio_iec61937_payload (infobuf.data, infobuf.size,
infoout.data, infoout.size, &sink->ringbuffer->spec);
if (!success) {
gst_buffer_unmap (out, &infoout);
gst_buffer_unmap (buf, &infobuf);
gst_buffer_unref (out);
return NULL;
}
gst_buffer_copy_into (out, buf, GST_BUFFER_COPY_ALL, 0, -1);
/* Fix endianness */
_swab ((gchar *) infoout.data, (gchar *) infoout.data, infobuf.size);
gst_buffer_unmap (out, &infoout);
gst_buffer_unmap (buf, &infobuf);
return out;
}
default:
return gst_buffer_ref (buf);
}
}
static void
gst_directsound_sink_set_volume (GstDirectSoundSink * dsoundsink,
gdouble dvolume, gboolean store)
......
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