static GstStateChangeReturn
gst_siren_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstSirenEnc *enc = GST_SIREN_ENC (element);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
enc->discont = FALSE;
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_adapter_clear (enc->adapter);
break;
default:
break;
}
return ret;
}
static gboolean
gst_siren_enc_stop (GstAudioEncoder * enc)
{
GstSirenEnc *senc = GST_SIREN_ENC (enc);
GST_DEBUG_OBJECT (senc, "stop");
Siren7_CloseEncoder (senc->encoder);
return TRUE;
}
static gboolean
gst_siren_enc_start (GstAudioEncoder * enc)
{
GstSirenEnc *senc = GST_SIREN_ENC (enc);
GST_DEBUG_OBJECT (enc, "start");
senc->encoder = Siren7_NewEncoder (16000);
return TRUE;
}
static void
gst_siren_enc_finalize (GObject * object)
{
GstSirenEnc *enc = GST_SIREN_ENC (object);
GST_DEBUG_OBJECT (object, "Disposing");
Siren7_CloseEncoder (enc->encoder);
g_object_unref (enc->adapter);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_siren_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstSirenEnc *enc;
gboolean res;
GstCaps *outcaps;
enc = GST_SIREN_ENC (GST_PAD_PARENT (pad));
outcaps = gst_static_pad_template_get_caps (&srctemplate);
res = gst_pad_set_caps (enc->srcpad, outcaps);
gst_caps_unref (outcaps);
return res;
}
static gboolean
gst_siren_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
GstSirenEnc *enc;
gboolean res;
GstCaps *outcaps;
enc = GST_SIREN_ENC (benc);
outcaps = gst_static_pad_template_get_caps (&srctemplate);
res = gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), outcaps);
gst_caps_unref (outcaps);
/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (benc, 320);
gst_audio_encoder_set_frame_samples_max (benc, 320);
/* no remainder or flushing please */
gst_audio_encoder_set_hard_min (benc, TRUE);
gst_audio_encoder_set_drainable (benc, FALSE);
return res;
}
static gboolean
gst_siren_enc_sink_event (GstPad * pad, GstEvent * event)
{
GstSirenEnc *enc;
gboolean res;
enc = GST_SIREN_ENC (GST_PAD_PARENT (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
gst_adapter_clear (enc->adapter);
res = gst_pad_push_event (enc->srcpad, event);
break;
case GST_EVENT_FLUSH_STOP:
gst_adapter_clear (enc->adapter);
res = gst_pad_push_event (enc->srcpad, event);
break;
default:
res = gst_pad_push_event (enc->srcpad, event);
break;
}
return res;
}
static GstFlowReturn
gst_siren_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstSirenEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
guint8 *in_data, *out_data;
guint i, size, num_frames;
gint out_size, in_size;
gint encode_ret;
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
enc = GST_SIREN_ENC (benc);
size = GST_BUFFER_SIZE (buf);
GST_LOG_OBJECT (enc, "Received buffer of size %d", GST_BUFFER_SIZE (buf));
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
g_return_val_if_fail (size % 640 == 0, GST_FLOW_ERROR);
/* we need to process 640 input bytes to produce 40 output bytes */
/* calculate the amount of frames we will handle */
num_frames = size / 640;
/* this is the input/output size */
in_size = num_frames * 640;
out_size = num_frames * 40;
GST_LOG_OBJECT (enc, "we have %u frames, %u in, %u out", num_frames, in_size,
out_size);
/* get a buffer */
ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (benc),
-1, out_size, GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (benc)), &out_buf);
if (ret != GST_FLOW_OK)
goto alloc_failed;
/* get the input data for all the frames */
in_data = GST_BUFFER_DATA (buf);
out_data = GST_BUFFER_DATA (out_buf);
for (i = 0; i < num_frames; i++) {
GST_LOG_OBJECT (enc, "Encoding frame %u/%u", i, num_frames);
/* encode 640 input bytes to 40 output bytes */
encode_ret = Siren7_EncodeFrame (enc->encoder, in_data, out_data);
if (encode_ret != 0)
goto encode_error;
/* move to next frame */
out_data += 40;
in_data += 640;
}
GST_LOG_OBJECT (enc, "Finished encoding");
/* we encode all we get, pass it along */
ret = gst_audio_encoder_finish_frame (benc, out_buf, -1);
done:
return ret;
/* ERRORS */
alloc_failed:
{
GST_DEBUG_OBJECT (enc, "failed to pad_alloc buffer: %d (%s)", ret,
gst_flow_get_name (ret));
goto done;
}
encode_error:
{
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
("Error encoding frame: %d", encode_ret));
ret = GST_FLOW_ERROR;
gst_buffer_unref (out_buf);
goto done;
}
}
static GstFlowReturn
gst_siren_enc_chain (GstPad * pad, GstBuffer * buf)
{
GstSirenEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
guint8 *in_data, *out_data;
guint8 *to_free = NULL;
guint i, size, num_frames;
gint out_size, in_size;
gint encode_ret;
gboolean discont;
GstClockTime timestamp;
guint64 distance;
GstCaps *outcaps;
enc = GST_SIREN_ENC (GST_PAD_PARENT (pad));
discont = GST_BUFFER_IS_DISCONT (buf);
if (discont) {
GST_DEBUG_OBJECT (enc, "received DISCONT, flush adapter");
gst_adapter_clear (enc->adapter);
enc->discont = TRUE;
}
gst_adapter_push (enc->adapter, buf);
size = gst_adapter_available (enc->adapter);
GST_LOG_OBJECT (enc, "Received buffer of size %d with adapter of size : %d",
GST_BUFFER_SIZE (buf), size);
/* we need to process 640 input bytes to produce 40 output bytes */
/* calculate the amount of frames we will handle */
num_frames = size / 640;
/* no frames, wait some more */
if (num_frames == 0)
goto done;
/* this is the input/output size */
in_size = num_frames * 640;
out_size = num_frames * 40;
GST_LOG_OBJECT (enc, "we have %u frames, %u in, %u out", num_frames, in_size,
out_size);
/* set output caps when needed */
if ((outcaps = GST_PAD_CAPS (enc->srcpad)) == NULL) {
outcaps = gst_static_pad_template_get_caps (&srctemplate);
gst_pad_set_caps (enc->srcpad, outcaps);
gst_caps_unref (outcaps);
}
/* get a buffer */
ret = gst_pad_alloc_buffer_and_set_caps (enc->srcpad, -1,
out_size, outcaps, &out_buf);
if (ret != GST_FLOW_OK)
goto alloc_failed;
/* get the timestamp for the output buffer */
timestamp = gst_adapter_prev_timestamp (enc->adapter, &distance);
/* add the amount of time taken by the distance */
if (timestamp != -1)
timestamp += gst_util_uint64_scale_int (distance / 2, GST_SECOND, 16000);
GST_LOG_OBJECT (enc,
"timestamp %" GST_TIME_FORMAT ", distance %" G_GUINT64_FORMAT,
GST_TIME_ARGS (timestamp), distance);
/* get the input data for all the frames */
to_free = in_data = gst_adapter_take (enc->adapter, in_size);
out_data = GST_BUFFER_DATA (out_buf);
for (i = 0; i < num_frames; i++) {
GST_LOG_OBJECT (enc, "Encoding frame %u/%u", i, num_frames);
/* encode 640 input bytes to 40 output bytes */
encode_ret = Siren7_EncodeFrame (enc->encoder, in_data, out_data);
if (encode_ret != 0)
goto encode_error;
/* move to next frame */
out_data += 40;
in_data += 640;
}
GST_LOG_OBJECT (enc, "Finished encoding");
/* mark discont */
if (enc->discont) {
GST_BUFFER_FLAG_SET (out_buf, GST_BUFFER_FLAG_DISCONT);
enc->discont = FALSE;
}
GST_BUFFER_TIMESTAMP (out_buf) = timestamp;
GST_BUFFER_DURATION (out_buf) = num_frames * FRAME_DURATION;
ret = gst_pad_push (enc->srcpad, out_buf);
done:
if (to_free)
g_free (to_free);
return ret;
/* ERRORS */
alloc_failed:
{
GST_DEBUG_OBJECT (enc, "failed to pad_alloc buffer: %d (%s)", ret,
gst_flow_get_name (ret));
goto done;
}
encode_error:
{
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
("Error encoding frame: %d", encode_ret));
ret = GST_FLOW_ERROR;
gst_buffer_unref (out_buf);
goto done;
}
}
static GstFlowReturn
gst_siren_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstSirenEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out_buf;
guint8 *in_data, *out_data;
guint i, size, num_frames;
gint out_size;
#ifndef GST_DISABLE_GST_DEBUG
gint in_size;
#endif
gint encode_ret;
GstMapInfo inmap, outmap;
g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
enc = GST_SIREN_ENC (benc);
size = gst_buffer_get_size (buf);
GST_LOG_OBJECT (enc, "Received buffer of size %d", size);
g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
g_return_val_if_fail (size % 640 == 0, GST_FLOW_ERROR);
/* we need to process 640 input bytes to produce 40 output bytes */
/* calculate the amount of frames we will handle */
num_frames = size / 640;
/* this is the input/output size */
#ifndef GST_DISABLE_GST_DEBUG
in_size = num_frames * 640;
#endif
out_size = num_frames * 40;
GST_LOG_OBJECT (enc, "we have %u frames, %u in, %u out", num_frames, in_size,
out_size);
/* get a buffer */
out_buf = gst_audio_encoder_allocate_output_buffer (benc, out_size);
if (out_buf == NULL)
goto alloc_failed;
/* get the input data for all the frames */
gst_buffer_map (buf, &inmap, GST_MAP_READ);
gst_buffer_map (out_buf, &outmap, GST_MAP_READ);
in_data = inmap.data;
out_data = outmap.data;
for (i = 0; i < num_frames; i++) {
GST_LOG_OBJECT (enc, "Encoding frame %u/%u", i, num_frames);
/* encode 640 input bytes to 40 output bytes */
encode_ret = Siren7_EncodeFrame (enc->encoder, in_data, out_data);
if (encode_ret != 0)
goto encode_error;
/* move to next frame */
out_data += 40;
in_data += 640;
}
gst_buffer_unmap (buf, &inmap);
gst_buffer_unmap (out_buf, &outmap);
GST_LOG_OBJECT (enc, "Finished encoding");
/* we encode all we get, pass it along */
ret = gst_audio_encoder_finish_frame (benc, out_buf, -1);
done:
return ret;
/* ERRORS */
alloc_failed:
{
GST_DEBUG_OBJECT (enc, "failed to pad_alloc buffer: %d (%s)", ret,
gst_flow_get_name (ret));
goto done;
}
encode_error:
{
GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
("Error encoding frame: %d", encode_ret));
ret = GST_FLOW_ERROR;
gst_buffer_unref (out_buf);
goto done;
}
}
