static GstFlowReturn
gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length)
{
GstGSMDec *gsmdec = GST_GSMDEC (dec);
guint size;
size = gst_adapter_available (adapter);
/* if input format is TIME each buffer should be self-contained and
* the data is presumably packetised, and we should start with a clean
* slate/state at the beginning of each buffer (for wav49 case) */
if (dec->input_segment.format == GST_FORMAT_TIME) {
*offset = 0;
*length = size;
gsmdec->needed = 33;
return GST_FLOW_OK;
}
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
if (size < gsmdec->needed)
return GST_FLOW_EOS;
*offset = 0;
*length = gsmdec->needed;
/* WAV49 requires alternating 33 and 32 bytes of input */
if (gsmdec->use_wav49) {
gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33);
}
return GST_FLOW_OK;
}
static gboolean
gst_gsmdec_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstGSMDec *gsmdec;
GstCaps *srccaps;
GstStructure *s;
gboolean ret = FALSE;
gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
s = gst_caps_get_structure (caps, 0);
if (s == NULL)
goto wrong_caps;
/* figure out if we deal with plain or MSGSM */
if (gst_structure_has_name (s, "audio/x-gsm"))
gsmdec->use_wav49 = 0;
else if (gst_structure_has_name (s, "audio/ms-gsm"))
gsmdec->use_wav49 = 1;
else
goto wrong_caps;
if (!gst_structure_get_int (s, "rate", &gsmdec->rate)) {
GST_WARNING_OBJECT (gsmdec, "missing sample rate parameter from sink caps");
goto beach;
}
/* MSGSM needs different framing */
gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49);
gsmdec->duration = gst_util_uint64_scale (ENCODED_SAMPLES,
GST_SECOND, gsmdec->rate);
/* Setting up src caps based on the input sample rate. */
srccaps = gst_caps_new_simple ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"rate", G_TYPE_INT, gsmdec->rate, "channels", G_TYPE_INT, 1, NULL);
ret = gst_pad_set_caps (gsmdec->srcpad, srccaps);
gst_caps_unref (srccaps);
gst_object_unref (gsmdec);
return ret;
/* ERRORS */
wrong_caps:
GST_ERROR_OBJECT (gsmdec, "invalid caps received");
beach:
gst_object_unref (gsmdec);
return ret;
}
static GstFlowReturn
gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{
GstGSMDec *gsmdec;
gsm_signal *out_data;
gsm_byte *data;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
GstMapInfo map, omap;
gsize outsize;
guint frames, i, errors = 0;
/* no fancy draining */
if (G_UNLIKELY (!buffer))
return GST_FLOW_OK;
gsmdec = GST_GSMDEC (dec);
gst_buffer_map (buffer, &map, GST_MAP_READ);
frames = gst_gsmdec_get_frame_count (gsmdec, map.size);
/* always the same amount of output samples (20ms worth per frame) */
outsize = ENCODED_SAMPLES * frames * sizeof (gsm_signal);
outbuf = gst_buffer_new_and_alloc (outsize);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
out_data = (gsm_signal *) omap.data;
data = (gsm_byte *) map.data;
for (i = 0; i < frames; ++i) {
/* now encode frame into the output buffer */
if (gsm_decode (gsmdec->state, data, out_data) < 0) {
/* invalid frame */
GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL),
("tried to decode an invalid frame"), ret);
memset (out_data, 0, ENCODED_SAMPLES * sizeof (gsm_signal));
++errors;
}
out_data += ENCODED_SAMPLES;
data += gsmdec->needed;
if (gsmdec->use_wav49)
gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33);
}
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unmap (buffer, &map);
if (errors == frames) {
gst_buffer_unref (outbuf);
outbuf = NULL;
}
gst_audio_decoder_finish_frame (dec, outbuf, 1);
return ret;
}
static gboolean
gst_gsmdec_stop (GstAudioDecoder * dec)
{
GstGSMDec *gsmdec = GST_GSMDEC (dec);
GST_DEBUG_OBJECT (dec, "stop");
gsm_destroy (gsmdec->state);
return TRUE;
}
static gboolean
gst_gsmdec_start (GstAudioDecoder * dec)
{
GstGSMDec *gsmdec = GST_GSMDEC (dec);
GST_DEBUG_OBJECT (dec, "start");
gsmdec->state = gsm_create ();
return TRUE;
}
static void
gst_gsmdec_finalize (GObject * object)
{
GstGSMDec *gsmdec;
gsmdec = GST_GSMDEC (object);
g_object_unref (gsmdec->adapter);
gsm_destroy (gsmdec->state);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
{
GstGSMDec *gsmdec;
GstStructure *s;
gboolean ret = FALSE;
gint rate;
GstAudioInfo info;
gsmdec = GST_GSMDEC (dec);
s = gst_caps_get_structure (caps, 0);
if (s == NULL)
goto wrong_caps;
/* figure out if we deal with plain or MSGSM */
if (gst_structure_has_name (s, "audio/x-gsm"))
gsmdec->use_wav49 = 0;
else if (gst_structure_has_name (s, "audio/ms-gsm"))
gsmdec->use_wav49 = 1;
else
goto wrong_caps;
gsmdec->needed = 33;
if (!gst_structure_get_int (s, "rate", &rate)) {
GST_WARNING_OBJECT (gsmdec, "missing sample rate parameter from sink caps");
goto beach;
}
/* MSGSM needs different framing */
gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49);
/* Setting up src caps based on the input sample rate. */
gst_audio_info_init (&info);
gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16, rate, 1, NULL);
ret = gst_audio_decoder_set_output_format (dec, &info);
return ret;
/* ERRORS */
wrong_caps:
GST_ERROR_OBJECT (gsmdec, "invalid caps received");
beach:
return ret;
}
static gboolean
gst_gsmdec_sink_event (GstPad * pad, GstEvent * event)
{
gboolean res;
GstGSMDec *gsmdec;
gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_FLUSH_START:
res = gst_pad_push_event (gsmdec->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&gsmdec->segment, GST_FORMAT_UNDEFINED);
res = gst_pad_push_event (gsmdec->srcpad, event);
break;
case GST_EVENT_NEWSEGMENT:
{
gboolean update;
GstFormat format;
gdouble rate, arate;
gint64 start, stop, time;
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
/* now configure the values */
gst_segment_set_newsegment_full (&gsmdec->segment, update,
rate, arate, format, start, stop, time);
/* and forward */
res = gst_pad_push_event (gsmdec->srcpad, event);
break;
}
case GST_EVENT_EOS:
default:
res = gst_pad_push_event (gsmdec->srcpad, event);
break;
}
gst_object_unref (gsmdec);
return res;
}
static GstFlowReturn
gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
{
GstGSMDec *gsmdec;
gsm_byte *data;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
GstMapInfo map, omap;
/* no fancy draining */
if (G_UNLIKELY (!buffer))
return GST_FLOW_OK;
gsmdec = GST_GSMDEC (dec);
/* always the same amount of output samples */
outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
/* now encode frame into the output buffer */
gst_buffer_map (buffer, &map, GST_MAP_READ);
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
data = (gsm_byte *) map.data;
if (gsm_decode (gsmdec->state, data, (gsm_signal *) omap.data) < 0) {
/* invalid frame */
GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL),
("tried to decode an invalid frame"), ret);
gst_buffer_unmap (outbuf, &omap);
gst_buffer_unref (outbuf);
outbuf = NULL;
} else {
gst_buffer_unmap (outbuf, &omap);
}
gst_buffer_unmap (buffer, &map);
gst_audio_decoder_finish_frame (dec, outbuf, 1);
return ret;
}
static GstFlowReturn
gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
gint * offset, gint * length)
{
GstGSMDec *gsmdec = GST_GSMDEC (dec);
guint size;
size = gst_adapter_available (adapter);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
/* WAV49 requires alternating 33 and 32 bytes of input */
if (gsmdec->use_wav49) {
gsmdec->needed = (gsmdec->needed == 33 ? 32 : 33);
}
if (size < gsmdec->needed)
return GST_FLOW_EOS;
*offset = 0;
*length = gsmdec->needed;
return GST_FLOW_OK;
}
static GstFlowReturn
gst_gsmdec_chain (GstPad * pad, GstBuffer * buf)
{
GstGSMDec *gsmdec;
gsm_byte *data;
GstFlowReturn ret = GST_FLOW_OK;
GstClockTime timestamp;
gint needed;
gsmdec = GST_GSMDEC (gst_pad_get_parent (pad));
timestamp = GST_BUFFER_TIMESTAMP (buf);
if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (gsmdec->adapter);
gsmdec->next_ts = GST_CLOCK_TIME_NONE;
/* FIXME, do some good offset */
gsmdec->next_of = 0;
}
gst_adapter_push (gsmdec->adapter, buf);
needed = 33;
/* do we have enough bytes to read a frame */
while (gst_adapter_available (gsmdec->adapter) >= needed) {
GstBuffer *outbuf;
/* always the same amount of output samples */
outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
/* If we are not given any timestamp, interpolate from last seen
* timestamp (if any). */
if (timestamp == GST_CLOCK_TIME_NONE)
timestamp = gsmdec->next_ts;
GST_BUFFER_TIMESTAMP (outbuf) = timestamp;
/* interpolate in the next run */
if (timestamp != GST_CLOCK_TIME_NONE)
gsmdec->next_ts = timestamp + gsmdec->duration;
timestamp = GST_CLOCK_TIME_NONE;
GST_BUFFER_DURATION (outbuf) = gsmdec->duration;
GST_BUFFER_OFFSET (outbuf) = gsmdec->next_of;
if (gsmdec->next_of != -1)
gsmdec->next_of += ENCODED_SAMPLES;
GST_BUFFER_OFFSET_END (outbuf) = gsmdec->next_of;
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmdec->srcpad));
/* now encode frame into the output buffer */
data = (gsm_byte *) gst_adapter_peek (gsmdec->adapter, needed);
if (gsm_decode (gsmdec->state, data,
(gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
/* invalid frame */
GST_WARNING_OBJECT (gsmdec, "tried to decode an invalid frame, skipping");
}
gst_adapter_flush (gsmdec->adapter, needed);
/* WAV49 requires alternating 33 and 32 bytes of input */
if (gsmdec->use_wav49)
needed = (needed == 33 ? 32 : 33);
GST_DEBUG_OBJECT (gsmdec, "Pushing buffer of size %d ts %" GST_TIME_FORMAT,
GST_BUFFER_SIZE (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
/* push */
ret = gst_pad_push (gsmdec->srcpad, outbuf);
}
gst_object_unref (gsmdec);
return ret;
}
