static GstStateChangeReturn
gst_amrnbenc_state_change (GstElement * element, GstStateChange transition)
{
GstAmrnbEnc *amrnbenc;
GstStateChangeReturn ret;
amrnbenc = GST_AMRNBENC (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (!(amrnbenc->handle = Encoder_Interface_init (0)))
return GST_STATE_CHANGE_FAILURE;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
amrnbenc->rate = 0;
amrnbenc->channels = 0;
amrnbenc->ts = 0;
gst_adapter_clear (amrnbenc->adapter);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_READY_TO_NULL:
Encoder_Interface_exit (amrnbenc->handle);
break;
default:
break;
}
return ret;
}
static GstFlowReturn
gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
{
GstAmrnbEnc *amrnbenc;
GstFlowReturn ret;
GstBuffer *out;
GstMapInfo in_map, out_map;
gsize out_size;
amrnbenc = GST_AMRNBENC (enc);
g_return_val_if_fail (amrnbenc->handle, GST_FLOW_FLUSHING);
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buffer == NULL)) {
GST_DEBUG_OBJECT (amrnbenc, "no data");
return GST_FLOW_OK;
}
gst_buffer_map (buffer, &in_map, GST_MAP_READ);
if (G_UNLIKELY (in_map.size < 320)) {
gst_buffer_unmap (buffer, &in_map);
GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data of %" G_GSIZE_FORMAT
" bytes", in_map.size);
return gst_audio_encoder_finish_frame (enc, NULL, -1);
}
/* get output, max size is 32 */
out = gst_buffer_new_and_alloc (32);
/* AMR encoder actually writes into the source data buffers it gets */
/* should be able to handle that with what we are given */
gst_buffer_map (out, &out_map, GST_MAP_WRITE);
/* encode */
out_size =
Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
(short *) in_map.data, out_map.data, 0);
gst_buffer_unmap (out, &out_map);
gst_buffer_resize (out, 0, out_size);
gst_buffer_unmap (buffer, &in_map);
GST_LOG_OBJECT (amrnbenc, "output data size %" G_GSIZE_FORMAT, out_size);
if (out_size) {
ret = gst_audio_encoder_finish_frame (enc, out, 160);
} else {
/* should not happen (without dtx or so at least) */
GST_WARNING_OBJECT (amrnbenc, "no encoded data; discarding input");
gst_buffer_unref (out);
ret = gst_audio_encoder_finish_frame (enc, NULL, -1);
}
return ret;
}
static gboolean
gst_amrnbenc_stop (GstAudioEncoder * enc)
{
GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
GST_DEBUG_OBJECT (amrnbenc, "stop");
Encoder_Interface_exit (amrnbenc->handle);
return TRUE;
}
static gboolean
gst_amrnbenc_start (GstAudioEncoder * enc)
{
GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);
GST_DEBUG_OBJECT (amrnbenc, "start");
if (!(amrnbenc->handle = Encoder_Interface_init (0)))
return FALSE;
return TRUE;
}
static void
gst_amrnbenc_finalize (GObject * object)
{
GstAmrnbEnc *amrnbenc;
amrnbenc = GST_AMRNBENC (object);
g_object_unref (G_OBJECT (amrnbenc->adapter));
amrnbenc->adapter = NULL;
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_amrnbenc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstAmrnbEnc *self = GST_AMRNBENC (object);
switch (prop_id) {
case PROP_BANDMODE:
g_value_set_enum (value, self->bandmode);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
return;
}
static gboolean
gst_amrnbenc_setcaps (GstPad * pad, GstCaps * caps)
{
GstStructure *structure;
GstAmrnbEnc *amrnbenc;
GstCaps *copy;
amrnbenc = GST_AMRNBENC (GST_PAD_PARENT (pad));
structure = gst_caps_get_structure (caps, 0);
/* get channel count */
gst_structure_get_int (structure, "channels", &amrnbenc->channels);
gst_structure_get_int (structure, "rate", &amrnbenc->rate);
/* this is not wrong but will sound bad */
if (amrnbenc->channels != 1) {
g_warning ("amrnbdec is only optimized for mono channels");
}
if (amrnbenc->rate != 8000) {
g_warning ("amrnbdec is only optimized for 8000 Hz samplerate");
}
/* create reverse caps */
copy = gst_caps_new_simple ("audio/AMR",
"channels", G_TYPE_INT, amrnbenc->channels,
"rate", G_TYPE_INT, amrnbenc->rate, NULL);
/* precalc duration as it's constant now */
amrnbenc->duration = gst_util_uint64_scale_int (160, GST_SECOND,
amrnbenc->rate * amrnbenc->channels);
gst_pad_set_caps (amrnbenc->srcpad, copy);
gst_caps_unref (copy);
return TRUE;
}
static gboolean
gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstAmrnbEnc *amrnbenc;
GstCaps *copy;
amrnbenc = GST_AMRNBENC (enc);
/* parameters already parsed for us */
amrnbenc->rate = GST_AUDIO_INFO_RATE (info);
amrnbenc->channels = GST_AUDIO_INFO_CHANNELS (info);
/* we do not really accept other input, but anyway ... */
/* this is not wrong but will sound bad */
if (amrnbenc->channels != 1) {
g_warning ("amrnbdec is only optimized for mono channels");
}
if (amrnbenc->rate != 8000) {
g_warning ("amrnbdec is only optimized for 8000 Hz samplerate");
}
/* create reverse caps */
copy = gst_caps_new_simple ("audio/AMR",
"channels", G_TYPE_INT, amrnbenc->channels,
"rate", G_TYPE_INT, amrnbenc->rate, NULL);
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (amrnbenc), copy);
gst_caps_unref (copy);
/* report needs to base class: hand one frame at a time */
gst_audio_encoder_set_frame_samples_min (enc, 160);
gst_audio_encoder_set_frame_samples_max (enc, 160);
gst_audio_encoder_set_frame_max (enc, 1);
return TRUE;
}
static GstFlowReturn
gst_amrnbenc_chain (GstPad * pad, GstBuffer * buffer)
{
GstAmrnbEnc *amrnbenc;
GstFlowReturn ret;
amrnbenc = GST_AMRNBENC (GST_PAD_PARENT (pad));
g_return_val_if_fail (amrnbenc->handle, GST_FLOW_WRONG_STATE);
if (amrnbenc->rate == 0 || amrnbenc->channels == 0)
goto not_negotiated;
/* discontinuity clears adapter, FIXME, maybe we can set some
* encoder flag to mask the discont. */
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
gst_adapter_clear (amrnbenc->adapter);
amrnbenc->ts = 0;
}
/* take latest timestamp, FIXME timestamp is the one of the
* first buffer in the adapter. */
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer))
amrnbenc->ts = GST_BUFFER_TIMESTAMP (buffer);
ret = GST_FLOW_OK;
gst_adapter_push (amrnbenc->adapter, buffer);
/* Collect samples until we have enough for an output frame */
while (gst_adapter_available (amrnbenc->adapter) >= 320) {
GstBuffer *out;
guint8 *data;
gint outsize;
/* get output, max size is 32 */
out = gst_buffer_new_and_alloc (32);
GST_BUFFER_DURATION (out) = amrnbenc->duration;
GST_BUFFER_TIMESTAMP (out) = amrnbenc->ts;
if (amrnbenc->ts != -1)
amrnbenc->ts += amrnbenc->duration;
gst_buffer_set_caps (out, GST_PAD_CAPS (amrnbenc->srcpad));
/* The AMR encoder actually writes into the source data buffers it gets */
data = gst_adapter_take (amrnbenc->adapter, 320);
/* encode */
outsize =
Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
(short *) data, (guint8 *) GST_BUFFER_DATA (out), 0);
g_free (data);
GST_BUFFER_SIZE (out) = outsize;
/* play */
if ((ret = gst_pad_push (amrnbenc->srcpad, out)) != GST_FLOW_OK)
break;
}
return ret;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (amrnbenc, STREAM, TYPE_NOT_FOUND,
(NULL), ("unknown type"));
return GST_FLOW_NOT_NEGOTIATED;
}
}
