GST_DEBUG_CATEGORY_STATIC (speexenc_debug);
#define GST_CAT_DEFAULT speexenc_debug
+#define FORMAT_STR GST_AUDIO_NE(S16)
+
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "rate = (int) [ 6000, 48000 ], "
- "channels = (int) [ 1, 2 ], "
- "endianness = (int) BYTE_ORDER, "
- "signed = (boolean) TRUE, " "width = (int) 16, " "depth = (int) 16")
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " FORMAT_STR ", "
+ "rate = (int) [ 6000, 48000 ], " "channels = (int) [ 1, 2 ]")
);
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
return speex_enc_mode_type;
}
-#if 0
-static const GstFormat *
-gst_speex_enc_get_formats (GstPad * pad)
-{
- static const GstFormat src_formats[] = {
- GST_FORMAT_BYTES,
- GST_FORMAT_TIME,
- 0
- };
- static const GstFormat sink_formats[] = {
- GST_FORMAT_BYTES,
- GST_FORMAT_DEFAULT,
- GST_FORMAT_TIME,
- 0
- };
-
- return (GST_PAD_IS_SRC (pad) ? src_formats : sink_formats);
-}
-#endif
-
static void gst_speex_enc_finalize (GObject * object);
-static gboolean gst_speex_enc_sink_event (GstPad * pad, GstEvent * event);
-static GstFlowReturn gst_speex_enc_chain (GstPad * pad, GstBuffer * buf);
static gboolean gst_speex_enc_setup (GstSpeexEnc * enc);
static void gst_speex_enc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_speex_enc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
-static GstStateChangeReturn gst_speex_enc_change_state (GstElement * element,
- GstStateChange transition);
-static GstFlowReturn gst_speex_enc_encode (GstSpeexEnc * enc, gboolean flush);
+static GstFlowReturn gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf);
+
+static gboolean gst_speex_enc_start (GstAudioEncoder * enc);
+static gboolean gst_speex_enc_stop (GstAudioEncoder * enc);
+static gboolean gst_speex_enc_set_format (GstAudioEncoder * enc,
+ GstAudioInfo * info);
+static GstFlowReturn gst_speex_enc_handle_frame (GstAudioEncoder * enc,
+ GstBuffer * in_buf);
+static gboolean gst_speex_enc_sink_event (GstAudioEncoder * enc,
+ GstEvent * event);
+static GstFlowReturn
+gst_speex_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer);
#define gst_speex_enc_parent_class parent_class
-G_DEFINE_TYPE_WITH_CODE (GstSpeexEnc, gst_speex_enc, GST_TYPE_ELEMENT,
+G_DEFINE_TYPE_WITH_CODE (GstSpeexEnc, gst_speex_enc, GST_TYPE_AUDIO_ENCODER,
G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL);
G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL));
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
+ GstAudioEncoderClass *base_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
+ base_class = (GstAudioEncoderClass *) klass;
gobject_class->finalize = gst_speex_enc_finalize;
gobject_class->set_property = gst_speex_enc_set_property;
gobject_class->get_property = gst_speex_enc_get_property;
+ base_class->start = GST_DEBUG_FUNCPTR (gst_speex_enc_start);
+ base_class->stop = GST_DEBUG_FUNCPTR (gst_speex_enc_stop);
+ base_class->set_format = GST_DEBUG_FUNCPTR (gst_speex_enc_set_format);
+ base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_speex_enc_handle_frame);
+ base_class->event = GST_DEBUG_FUNCPTR (gst_speex_enc_sink_event);
+ base_class->pre_push = GST_DEBUG_FUNCPTR (gst_speex_enc_pre_push);
+
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QUALITY,
g_param_spec_float ("quality", "Quality", "Encoding quality",
0.0, 10.0, DEFAULT_QUALITY,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BITRATE,
g_param_spec_int ("bitrate", "Encoding Bit-rate",
"Specify an encoding bit-rate (in bps). (0 = automatic)",
0, G_MAXINT, DEFAULT_BITRATE,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MODE,
g_param_spec_enum ("mode", "Mode", "The encoding mode",
GST_TYPE_SPEEX_ENC_MODE, GST_SPEEX_ENC_MODE_AUTO,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_VBR,
g_param_spec_boolean ("vbr", "VBR",
"Enable variable bit-rate", DEFAULT_VBR,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ABR,
g_param_spec_int ("abr", "ABR",
"Enable average bit-rate (0 = disabled)",
0, G_MAXINT, DEFAULT_ABR,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_VAD,
g_param_spec_boolean ("vad", "VAD",
"Enable voice activity detection", DEFAULT_VAD,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DTX,
g_param_spec_boolean ("dtx", "DTX",
"Enable discontinuous transmission", DEFAULT_DTX,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_COMPLEXITY,
g_param_spec_int ("complexity", "Complexity",
"Set encoding complexity",
0, G_MAXINT, DEFAULT_COMPLEXITY,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_NFRAMES,
g_param_spec_int ("nframes", "NFrames",
"Number of frames per buffer",
0, G_MAXINT, DEFAULT_NFRAMES,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LAST_MESSAGE,
g_param_spec_string ("last-message", "last-message",
"The last status message", NULL,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
- gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_speex_enc_change_state);
-
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (gstelement_class,
enc = GST_SPEEX_ENC (object);
g_free (enc->last_message);
- g_object_unref (enc->adapter);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
-static gboolean
-gst_speex_enc_sink_setcaps (GstPad * pad, GstCaps * caps)
+static void
+gst_speex_enc_init (GstSpeexEnc * enc)
{
- GstSpeexEnc *enc;
- GstStructure *structure;
-
- enc = GST_SPEEX_ENC (GST_PAD_PARENT (pad));
- enc->setup = FALSE;
-
- structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "channels", &enc->channels);
- gst_structure_get_int (structure, "rate", &enc->rate);
+ GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
- gst_speex_enc_setup (enc);
-
- return enc->setup;
+ /* arrange granulepos marking (and required perfect ts) */
+ gst_audio_encoder_set_mark_granule (benc, TRUE);
+ gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
}
-
-static GstCaps *
-gst_speex_enc_sink_getcaps (GstPad * pad, GstCaps * filter)
+static gboolean
+gst_speex_enc_start (GstAudioEncoder * benc)
{
- GstCaps *caps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
- GstCaps *peercaps = NULL;
- GstSpeexEnc *enc = GST_SPEEX_ENC (gst_pad_get_parent_element (pad));
-
- peercaps = gst_pad_peer_get_caps (enc->srcpad, filter);
-
- if (peercaps) {
- if (!gst_caps_is_empty (peercaps) && !gst_caps_is_any (peercaps)) {
- GstStructure *ps = gst_caps_get_structure (peercaps, 0);
- GstStructure *s = gst_caps_get_structure (caps, 0);
- gint rate, channels;
-
- if (gst_structure_get_int (ps, "rate", &rate)) {
- gst_structure_fixate_field_nearest_int (s, "rate", rate);
- }
-
- if (gst_structure_get_int (ps, "channels", &channels)) {
- gst_structure_fixate_field_nearest_int (s, "channels", channels);
- }
- }
- gst_caps_unref (peercaps);
- }
+ GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
- gst_object_unref (enc);
+ GST_DEBUG_OBJECT (enc, "start");
+ speex_bits_init (&enc->bits);
+ enc->tags = gst_tag_list_new_empty ();
+ enc->header_sent = FALSE;
- return caps;
+ return TRUE;
}
-
static gboolean
-gst_speex_enc_convert_src (GstPad * pad, GstFormat src_format, gint64 src_value,
- GstFormat * dest_format, gint64 * dest_value)
+gst_speex_enc_stop (GstAudioEncoder * benc)
{
- gboolean res = TRUE;
- GstSpeexEnc *enc;
- gint64 avg;
-
- enc = GST_SPEEX_ENC (GST_PAD_PARENT (pad));
-
- if (enc->samples_in == 0 || enc->bytes_out == 0 || enc->rate == 0)
- return FALSE;
+ GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
- avg = (enc->bytes_out * enc->rate) / (enc->samples_in);
-
- switch (src_format) {
- case GST_FORMAT_BYTES:
- switch (*dest_format) {
- case GST_FORMAT_TIME:
- *dest_value = src_value * GST_SECOND / avg;
- break;
- default:
- res = FALSE;
- }
- break;
- case GST_FORMAT_TIME:
- switch (*dest_format) {
- case GST_FORMAT_BYTES:
- *dest_value = src_value * avg / GST_SECOND;
- break;
- default:
- res = FALSE;
- }
- break;
- default:
- res = FALSE;
+ GST_DEBUG_OBJECT (enc, "stop");
+ enc->header_sent = FALSE;
+ if (enc->state) {
+ speex_encoder_destroy (enc->state);
+ enc->state = NULL;
}
- return res;
-}
+ speex_bits_destroy (&enc->bits);
+ gst_tag_list_free (enc->tags);
+ enc->tags = NULL;
+ g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL);
+ enc->headers = NULL;
-static gboolean
-gst_speex_enc_convert_sink (GstPad * pad, GstFormat src_format,
- gint64 src_value, GstFormat * dest_format, gint64 * dest_value)
-{
- gboolean res = TRUE;
- guint scale = 1;
- gint bytes_per_sample;
- GstSpeexEnc *enc;
+ gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
- enc = GST_SPEEX_ENC (GST_PAD_PARENT (pad));
-
- bytes_per_sample = enc->channels * 2;
-
- switch (src_format) {
- case GST_FORMAT_BYTES:
- switch (*dest_format) {
- case GST_FORMAT_DEFAULT:
- if (bytes_per_sample == 0)
- return FALSE;
- *dest_value = src_value / bytes_per_sample;
- break;
- case GST_FORMAT_TIME:
- {
- gint byterate = bytes_per_sample * enc->rate;
-
- if (byterate == 0)
- return FALSE;
- *dest_value = src_value * GST_SECOND / byterate;
- break;
- }
- default:
- res = FALSE;
- }
- break;
- case GST_FORMAT_DEFAULT:
- switch (*dest_format) {
- case GST_FORMAT_BYTES:
- *dest_value = src_value * bytes_per_sample;
- break;
- case GST_FORMAT_TIME:
- if (enc->rate == 0)
- return FALSE;
- *dest_value = src_value * GST_SECOND / enc->rate;
- break;
- default:
- res = FALSE;
- }
- break;
- case GST_FORMAT_TIME:
- switch (*dest_format) {
- case GST_FORMAT_BYTES:
- scale = bytes_per_sample;
- /* fallthrough */
- case GST_FORMAT_DEFAULT:
- *dest_value = src_value * scale * enc->rate / GST_SECOND;
- break;
- default:
- res = FALSE;
- }
- break;
- default:
- res = FALSE;
- }
- return res;
+ return TRUE;
}
static gint64
return 34 * GST_MSECOND;
}
-static const GstQueryType *
-gst_speex_enc_get_query_types (GstPad * pad)
-{
- static const GstQueryType gst_speex_enc_src_query_types[] = {
- GST_QUERY_POSITION,
- GST_QUERY_DURATION,
- GST_QUERY_CONVERT,
- GST_QUERY_LATENCY,
- 0
- };
-
- return gst_speex_enc_src_query_types;
-}
-
static gboolean
-gst_speex_enc_src_query (GstPad * pad, GstQuery * query)
+gst_speex_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
- gboolean res = TRUE;
GstSpeexEnc *enc;
- enc = GST_SPEEX_ENC (gst_pad_get_parent (pad));
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_POSITION:
- {
- GstFormat fmt, req_fmt;
- gint64 pos, val;
-
- gst_query_parse_position (query, &req_fmt, NULL);
- if ((res = gst_pad_query_peer_position (enc->sinkpad, &req_fmt, &val))) {
- gst_query_set_position (query, req_fmt, val);
- break;
- }
-
- fmt = GST_FORMAT_TIME;
- if (!(res = gst_pad_query_peer_position (enc->sinkpad, &fmt, &pos)))
- break;
-
- if ((res =
- gst_pad_query_peer_convert (enc->sinkpad, fmt, pos, &req_fmt,
- &val)))
- gst_query_set_position (query, req_fmt, val);
+ enc = GST_SPEEX_ENC (benc);
- break;
- }
- case GST_QUERY_DURATION:
- {
- GstFormat fmt, req_fmt;
- gint64 dur, val;
-
- gst_query_parse_duration (query, &req_fmt, NULL);
- if ((res = gst_pad_query_peer_duration (enc->sinkpad, &req_fmt, &val))) {
- gst_query_set_duration (query, req_fmt, val);
- break;
- }
-
- fmt = GST_FORMAT_TIME;
- if (!(res = gst_pad_query_peer_duration (enc->sinkpad, &fmt, &dur)))
- break;
+ enc->channels = GST_AUDIO_INFO_CHANNELS (info);
+ enc->rate = GST_AUDIO_INFO_RATE (info);
- if ((res =
- gst_pad_query_peer_convert (enc->sinkpad, fmt, dur, &req_fmt,
- &val))) {
- gst_query_set_duration (query, req_fmt, val);
- }
- break;
- }
- case GST_QUERY_CONVERT:
- {
- GstFormat src_fmt, dest_fmt;
- gint64 src_val, dest_val;
-
- gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
- if (!(res = gst_speex_enc_convert_src (pad, src_fmt, src_val, &dest_fmt,
- &dest_val)))
- goto error;
- gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
- break;
- }
- case GST_QUERY_LATENCY:
- {
- gboolean live;
- GstClockTime min_latency, max_latency;
- gint64 latency;
-
- if ((res = gst_pad_peer_query (enc->sinkpad, query))) {
- gst_query_parse_latency (query, &live, &min_latency, &max_latency);
- GST_LOG_OBJECT (pad, "Upstream latency: %" GST_PTR_FORMAT, query);
-
- latency = gst_speex_enc_get_latency (enc);
-
- /* add our latency */
- min_latency += latency;
- if (max_latency != -1)
- max_latency += latency;
-
- gst_query_set_latency (query, live, min_latency, max_latency);
- GST_LOG_OBJECT (pad, "Adjusted latency: %" GST_PTR_FORMAT, query);
- }
- break;
- }
- default:
- res = gst_pad_peer_query (enc->sinkpad, query);
- break;
+ /* handle reconfigure */
+ if (enc->state) {
+ speex_encoder_destroy (enc->state);
+ enc->state = NULL;
}
-error:
-
- gst_object_unref (enc);
-
- return res;
-}
+ if (!gst_speex_enc_setup (enc))
+ return FALSE;
-static gboolean
-gst_speex_enc_sink_query (GstPad * pad, GstQuery * query)
-{
- gboolean res = TRUE;
+ /* feedback to base class */
+ gst_audio_encoder_set_latency (benc,
+ gst_speex_enc_get_latency (enc), gst_speex_enc_get_latency (enc));
+ gst_audio_encoder_set_lookahead (benc, enc->lookahead);
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_CONVERT:
- {
- GstFormat src_fmt, dest_fmt;
- gint64 src_val, dest_val;
-
- gst_query_parse_convert (query, &src_fmt, &src_val, &dest_fmt, &dest_val);
- if (!(res =
- gst_speex_enc_convert_sink (pad, src_fmt, src_val, &dest_fmt,
- &dest_val)))
- goto error;
- gst_query_set_convert (query, src_fmt, src_val, dest_fmt, dest_val);
- break;
- }
- default:
- res = gst_pad_query_default (pad, query);
- break;
+ if (enc->nframes == 0) {
+ /* as many frames as available input allows */
+ gst_audio_encoder_set_frame_samples_min (benc, enc->frame_size);
+ gst_audio_encoder_set_frame_samples_max (benc, enc->frame_size);
+ gst_audio_encoder_set_frame_max (benc, 0);
+ } else {
+ /* exactly as many frames as configured */
+ gst_audio_encoder_set_frame_samples_min (benc,
+ enc->frame_size * enc->nframes);
+ gst_audio_encoder_set_frame_samples_max (benc,
+ enc->frame_size * enc->nframes);
+ gst_audio_encoder_set_frame_max (benc, 1);
}
-error:
- return res;
-}
-
-static void
-gst_speex_enc_init (GstSpeexEnc * enc)
-{
- enc->sinkpad = gst_pad_new_from_static_template (&sink_factory, "sink");
- gst_element_add_pad (GST_ELEMENT (enc), enc->sinkpad);
- gst_pad_set_event_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_sink_event));
- gst_pad_set_chain_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_chain));
- gst_pad_set_getcaps_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_sink_getcaps));
- gst_pad_set_query_function (enc->sinkpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_sink_query));
-
- enc->srcpad = gst_pad_new_from_static_template (&src_factory, "src");
- gst_pad_set_query_function (enc->srcpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_src_query));
- gst_pad_set_query_type_function (enc->srcpad,
- GST_DEBUG_FUNCPTR (gst_speex_enc_get_query_types));
- gst_element_add_pad (GST_ELEMENT (enc), enc->srcpad);
-
- enc->channels = -1;
- enc->rate = -1;
-
- enc->quality = DEFAULT_QUALITY;
- enc->bitrate = DEFAULT_BITRATE;
- enc->mode = DEFAULT_MODE;
- enc->vbr = DEFAULT_VBR;
- enc->abr = DEFAULT_ABR;
- enc->vad = DEFAULT_VAD;
- enc->dtx = DEFAULT_DTX;
- enc->complexity = DEFAULT_COMPLEXITY;
- enc->nframes = DEFAULT_NFRAMES;
-
- enc->setup = FALSE;
- enc->header_sent = FALSE;
-
- enc->adapter = gst_adapter_new ();
+ return TRUE;
}
static GstBuffer *
gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (enc)));
if (merged_tags == NULL)
- merged_tags = gst_tag_list_new ();
+ merged_tags = gst_tag_list_new_empty ();
GST_DEBUG_OBJECT (enc, "merged tags = %" GST_PTR_FORMAT, merged_tags);
comments = gst_tag_list_to_vorbiscomment_buffer (merged_tags, NULL,
0, "Encoded with GStreamer Speexenc");
gst_tag_list_free (merged_tags);
- GST_BUFFER_OFFSET (comments) = enc->bytes_out;
+ GST_BUFFER_OFFSET (comments) = 0;
GST_BUFFER_OFFSET_END (comments) = 0;
return comments;
static gboolean
gst_speex_enc_setup (GstSpeexEnc * enc)
{
- enc->setup = FALSE;
-
switch (enc->mode) {
case GST_SPEEX_ENC_MODE_UWB:
GST_LOG_OBJECT (enc, "configuring for requested UWB mode");
GST_LOG_OBJECT (enc, "we have frame size %d, lookahead %d", enc->frame_size,
enc->lookahead);
- enc->setup = TRUE;
-
return TRUE;
}
-/* prepare a buffer for transmission */
-static GstBuffer *
-gst_speex_enc_buffer_from_data (GstSpeexEnc * enc, guchar * data,
- gint data_len, guint64 granulepos)
-{
- GstBuffer *outbuf;
-
- outbuf = gst_buffer_new_and_alloc (data_len);
- gst_buffer_fill (outbuf, 0, data, data_len);
- GST_BUFFER_OFFSET (outbuf) = enc->bytes_out;
- GST_BUFFER_OFFSET_END (outbuf) = granulepos;
-
- GST_LOG_OBJECT (enc, "encoded buffer of %d bytes", data_len);
- return outbuf;
-}
-
-
-/* push out the buffer and do internal bookkeeping */
+/* push out the buffer */
static GstFlowReturn
gst_speex_enc_push_buffer (GstSpeexEnc * enc, GstBuffer * buffer)
{
- guint size;
-
- size = gst_buffer_get_size (buffer);
- enc->bytes_out += size;
-
- GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u", size);
-
- return gst_pad_push (enc->srcpad, buffer);
-}
-
-static GstCaps *
-gst_speex_enc_set_header_on_caps (GstCaps * caps, GstBuffer * buf1,
- GstBuffer * buf2)
-{
- GstStructure *structure = NULL;
- GstBuffer *buf;
- GValue array = { 0 };
- GValue value = { 0 };
-
- caps = gst_caps_make_writable (caps);
- structure = gst_caps_get_structure (caps, 0);
-
- g_assert (gst_buffer_is_writable (buf1));
- g_assert (gst_buffer_is_writable (buf2));
-
- /* mark buffers */
- GST_BUFFER_FLAG_SET (buf1, GST_BUFFER_FLAG_IN_CAPS);
- GST_BUFFER_FLAG_SET (buf2, GST_BUFFER_FLAG_IN_CAPS);
-
- /* put buffers in a fixed list */
- g_value_init (&array, GST_TYPE_ARRAY);
- g_value_init (&value, GST_TYPE_BUFFER);
- buf = gst_buffer_copy (buf1);
- gst_value_set_buffer (&value, buf);
- gst_buffer_unref (buf);
- gst_value_array_append_value (&array, &value);
- g_value_unset (&value);
- g_value_init (&value, GST_TYPE_BUFFER);
- buf = gst_buffer_copy (buf2);
- gst_value_set_buffer (&value, buf);
- gst_buffer_unref (buf);
- gst_value_array_append_value (&array, &value);
- gst_structure_set_value (structure, "streamheader", &array);
- g_value_unset (&value);
- g_value_unset (&array);
+ GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u",
+ gst_buffer_get_size (buffer));
- return caps;
+ return gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), buffer);
}
-
static gboolean
-gst_speex_enc_sink_event (GstPad * pad, GstEvent * event)
+gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
{
- gboolean res = TRUE;
GstSpeexEnc *enc;
- enc = GST_SPEEX_ENC (gst_pad_get_parent (pad));
+ enc = GST_SPEEX_ENC (benc);
switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_CAPS:
- {
- GstCaps *caps;
-
- gst_event_parse_caps (event, &caps);
- res = gst_speex_enc_sink_setcaps (pad, caps);
- gst_event_unref (event);
- break;
- }
- case GST_EVENT_EOS:
- if (enc->setup)
- gst_speex_enc_encode (enc, TRUE);
- res = gst_pad_event_default (pad, event);
- break;
case GST_EVENT_TAG:
{
if (enc->tags) {
} else {
g_assert_not_reached ();
}
- res = gst_pad_event_default (pad, event);
break;
}
default:
- res = gst_pad_event_default (pad, event);
break;
}
- gst_object_unref (enc);
-
- return res;
+ /* we only peeked, let base class handle it */
+ return FALSE;
}
static GstFlowReturn
-gst_speex_enc_encode (GstSpeexEnc * enc, gboolean flush)
+gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
{
gint frame_size = enc->frame_size;
- gint bytes = frame_size * 2 * enc->channels;
+ gint bytes = frame_size * 2 * enc->channels, samples;
+ gint outsize, written, dtx_ret = 0;
+ guint8 *data, *data0 = NULL, *bdata, *outdata;
+ gsize bsize, size;
+ GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
- if (flush && gst_adapter_available (enc->adapter) % bytes != 0) {
- guint diff = gst_adapter_available (enc->adapter) % bytes;
- GstBuffer *buf = gst_buffer_new_and_alloc (diff);
- gst_buffer_memset (buf, 0, 0, diff);
- gst_adapter_push (enc->adapter, buf);
- }
+ if (G_LIKELY (buf)) {
+ bdata = gst_buffer_map (buf, &bsize, NULL, GST_MAP_READ);
- while (gst_adapter_available (enc->adapter) >= bytes) {
- gint16 *data;
- gint outsize, written, dtx_ret;
- GstBuffer *outbuf;
- gchar *outdata;
+ if (G_UNLIKELY (bsize % bytes)) {
+ GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
- data = (gint16 *) gst_adapter_take (enc->adapter, bytes);
+ size = ((bsize / bytes) + 1) * bytes;
+ data0 = data = g_malloc0 (size);
+ memcpy (data, bdata, bsize);
+ gst_buffer_unmap (buf, bdata, bsize);
+ bdata = NULL;
+ } else {
+ data = bdata;
+ size = bsize;
+ }
+ } else {
+ GST_DEBUG_OBJECT (enc, "nothing to drain");
+ goto done;
+ }
- enc->samples_in += frame_size;
+ samples = size / (2 * enc->channels);
+ speex_bits_reset (&enc->bits);
+ /* FIXME what about dropped samples if DTS enabled ?? */
+
+ while (size) {
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", frame_size, bytes);
if (enc->channels == 2) {
- speex_encode_stereo_int (data, frame_size, &enc->bits);
+ speex_encode_stereo_int ((gint16 *) data, frame_size, &enc->bits);
}
- dtx_ret = speex_encode_int (enc->state, data, &enc->bits);
-
- g_free (data);
+ dtx_ret += speex_encode_int (enc->state, (gint16 *) data, &enc->bits);
- enc->frameno++;
- enc->frameno_out++;
+ data += bytes;
+ size -= bytes;
+ }
- if ((enc->frameno % enc->nframes) != 0)
- continue;
+ speex_bits_insert_terminator (&enc->bits);
+ outsize = speex_bits_nbytes (&enc->bits);
- speex_bits_insert_terminator (&enc->bits);
- outsize = speex_bits_nbytes (&enc->bits);
+ if (bdata)
+ gst_buffer_unmap (buf, bdata, bsize);
#if 0
- ret = gst_pad_alloc_buffer_and_set_caps (enc->srcpad,
- GST_BUFFER_OFFSET_NONE, outsize, GST_PAD_CAPS (enc->srcpad), &outbuf);
- if ((GST_FLOW_OK != ret))
- goto done;
+ ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
+ GST_BUFFER_OFFSET_NONE, outsize,
+ GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
+
+ if ((GST_FLOW_OK != ret))
+ goto done;
#endif
- outbuf = gst_buffer_new_allocate (NULL, outsize, 0);
+ outbuf = gst_buffer_new_allocate (NULL, outsize, 0);
+ outdata = gst_buffer_map (outbuf, NULL, NULL, GST_MAP_WRITE);
- outdata = gst_buffer_map (outbuf, NULL, NULL, GST_MAP_WRITE);
- written = speex_bits_write (&enc->bits, outdata, outsize);
+ written = speex_bits_write (&enc->bits, (gchar *) outdata, outsize);
- if (G_UNLIKELY (written != outsize)) {
- GST_ERROR_OBJECT (enc, "short write: %d < %d bytes", written, outsize);
- }
- gst_buffer_unmap (outbuf, outdata, written);
-
- speex_bits_reset (&enc->bits);
-
- if (!dtx_ret)
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
-
- GST_BUFFER_TIMESTAMP (outbuf) = enc->start_ts +
- gst_util_uint64_scale_int ((enc->frameno_out -
- enc->nframes) * frame_size - enc->lookahead, GST_SECOND, enc->rate);
- GST_BUFFER_DURATION (outbuf) =
- gst_util_uint64_scale_int (frame_size * enc->nframes, GST_SECOND,
- enc->rate);
- /* set gp time and granulepos; see gst-plugins-base/ext/ogg/README */
- GST_BUFFER_OFFSET_END (outbuf) = enc->granulepos_offset +
- ((enc->frameno_out) * frame_size - enc->lookahead);
- GST_BUFFER_OFFSET (outbuf) =
- gst_util_uint64_scale_int (GST_BUFFER_OFFSET_END (outbuf), GST_SECOND,
- enc->rate);
-
- ret = gst_speex_enc_push_buffer (enc, outbuf);
-
- if ((GST_FLOW_OK != ret) && (GST_FLOW_NOT_LINKED != ret))
- goto done;
+ if (G_UNLIKELY (written < outsize)) {
+ GST_ERROR_OBJECT (enc, "short write: %d < %d bytes", written, outsize);
+ } else if (G_UNLIKELY (written > outsize)) {
+ GST_ERROR_OBJECT (enc, "overrun: %d > %d bytes", written, outsize);
+ written = outsize;
}
+ gst_buffer_unmap (outbuf, outdata, written);
-done:
+ if (!dtx_ret)
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
+
+ ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
+ outbuf, samples);
+done:
+ g_free (data0);
return ret;
}
+/*
+ * (really really) FIXME: move into core (dixit tpm)
+ */
+/**
+ * _gst_caps_set_buffer_array:
+ * @caps: a #GstCaps
+ * @field: field in caps to set
+ * @buf: header buffers
+ *
+ * Adds given buffers to an array of buffers set as the given @field
+ * on the given @caps. List of buffer arguments must be NULL-terminated.
+ *
+ * Returns: input caps with a streamheader field added, or NULL if some error
+ */
+static GstCaps *
+_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
+ GstBuffer * buf, ...)
+{
+ GstStructure *structure = NULL;
+ va_list va;
+ GValue array = { 0 };
+ GValue value = { 0 };
+
+ g_return_val_if_fail (caps != NULL, NULL);
+ g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
+ g_return_val_if_fail (field != NULL, NULL);
+
+ caps = gst_caps_make_writable (caps);
+ structure = gst_caps_get_structure (caps, 0);
+
+ g_value_init (&array, GST_TYPE_ARRAY);
+
+ va_start (va, buf);
+ /* put buffers in a fixed list */
+ while (buf) {
+ g_assert (gst_buffer_is_writable (buf));
+
+ /* mark buffer */
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+
+ g_value_init (&value, GST_TYPE_BUFFER);
+ buf = gst_buffer_copy (buf);
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+ gst_value_set_buffer (&value, buf);
+ gst_buffer_unref (buf);
+ gst_value_array_append_value (&array, &value);
+ g_value_unset (&value);
+
+ buf = va_arg (va, GstBuffer *);
+ }
+
+ gst_structure_set_value (structure, field, &array);
+ g_value_unset (&array);
+
+ return caps;
+}
+
static GstFlowReturn
-gst_speex_enc_chain (GstPad * pad, GstBuffer * buf)
+gst_speex_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstSpeexEnc *enc;
GstFlowReturn ret = GST_FLOW_OK;
- enc = GST_SPEEX_ENC (GST_PAD_PARENT (pad));
-
- if (!enc->setup)
- goto not_setup;
+ enc = GST_SPEEX_ENC (benc);
if (!enc->header_sent) {
/* Speex streams begin with two headers; the initial header (with
/* create header buffer */
data = (guint8 *) speex_header_to_packet (&enc->header, &data_len);
- buf1 = gst_speex_enc_buffer_from_data (enc, data, data_len, 0);
- free (data);
+ buf1 = gst_buffer_new_wrapped (data, data_len);
+ GST_BUFFER_OFFSET_END (buf1) = 0;
+ GST_BUFFER_OFFSET (buf1) = 0;
/* create comment buffer */
buf2 = gst_speex_enc_create_metadata_buffer (enc);
/* mark and put on caps */
- caps = gst_pad_get_caps (enc->srcpad, NULL);
- caps = gst_speex_enc_set_header_on_caps (caps, buf1, buf2);
-
- gst_caps_set_simple (caps,
- "rate", G_TYPE_INT, enc->rate,
+ caps = gst_caps_new_simple ("audio/x-speex", "rate", G_TYPE_INT, enc->rate,
"channels", G_TYPE_INT, enc->channels, NULL);
+ caps = _gst_caps_set_buffer_array (caps, "streamheader", buf1, buf2, NULL);
/* negotiate with these caps */
GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
- gst_pad_set_caps (enc->srcpad, caps);
+
+ gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps);
gst_caps_unref (caps);
/* push out buffers */
- ret = gst_speex_enc_push_buffer (enc, buf1);
-
- if (ret != GST_FLOW_OK) {
- gst_buffer_unref (buf2);
- goto done;
- }
-
- ret = gst_speex_enc_push_buffer (enc, buf2);
-
- if (ret != GST_FLOW_OK)
- goto done;
-
- speex_bits_reset (&enc->bits);
+ /* store buffers for later pre_push sending */
+ g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL);
+ enc->headers = NULL;
+ GST_DEBUG_OBJECT (enc, "storing header buffers");
+ enc->headers = g_slist_prepend (enc->headers, buf2);
+ enc->headers = g_slist_prepend (enc->headers, buf1);
enc->header_sent = TRUE;
}
- /* Save the timestamp of the first buffer. This will be later
- * used as offset for all following buffers */
- if (enc->start_ts == GST_CLOCK_TIME_NONE) {
- if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
- enc->start_ts = GST_BUFFER_TIMESTAMP (buf);
- enc->granulepos_offset = gst_util_uint64_scale
- (GST_BUFFER_TIMESTAMP (buf), enc->rate, GST_SECOND);
- } else {
- enc->start_ts = 0;
- enc->granulepos_offset = 0;
- }
- }
-
- /* Check if we have a continous stream, if not drop some samples or the buffer or
- * insert some silence samples */
- if (enc->next_ts != GST_CLOCK_TIME_NONE &&
- GST_BUFFER_TIMESTAMP (buf) < enc->next_ts) {
- guint64 diff = enc->next_ts - GST_BUFFER_TIMESTAMP (buf);
- guint64 diff_bytes;
-
- GST_WARNING_OBJECT (enc, "Buffer is older than previous "
- "timestamp + duration (%" GST_TIME_FORMAT "< %" GST_TIME_FORMAT
- "), cannot handle. Clipping buffer.",
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)),
- GST_TIME_ARGS (enc->next_ts));
-
- diff_bytes = GST_CLOCK_TIME_TO_FRAMES (diff, enc->rate) * enc->channels * 2;
- if (diff_bytes >= gst_buffer_get_size (buf)) {
- gst_buffer_unref (buf);
- return GST_FLOW_OK;
- }
- buf = gst_buffer_make_writable (buf);
- gst_buffer_resize (buf, diff_bytes, -1);
+ GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf,
+ buf ? gst_buffer_get_size (buf) : 0);
- GST_BUFFER_TIMESTAMP (buf) += diff;
- if (GST_BUFFER_DURATION_IS_VALID (buf))
- GST_BUFFER_DURATION (buf) -= diff;
- }
-
- if (enc->next_ts != GST_CLOCK_TIME_NONE
- && GST_BUFFER_TIMESTAMP_IS_VALID (buf)) {
- guint64 max_diff =
- gst_util_uint64_scale (enc->frame_size, GST_SECOND, enc->rate);
+ ret = gst_speex_enc_encode (enc, buf);
- if (GST_BUFFER_TIMESTAMP (buf) != enc->next_ts &&
- GST_BUFFER_TIMESTAMP (buf) - enc->next_ts > max_diff) {
- GST_WARNING_OBJECT (enc,
- "Discontinuity detected: %" G_GUINT64_FORMAT " > %" G_GUINT64_FORMAT,
- GST_BUFFER_TIMESTAMP (buf) - enc->next_ts, max_diff);
+ return ret;
+}
- gst_speex_enc_encode (enc, TRUE);
+static GstFlowReturn
+gst_speex_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer)
+{
+ GstSpeexEnc *enc;
+ GstFlowReturn ret = GST_FLOW_OK;
- enc->frameno_out = 0;
- enc->start_ts = GST_BUFFER_TIMESTAMP (buf);
- enc->granulepos_offset = gst_util_uint64_scale
- (GST_BUFFER_TIMESTAMP (buf), enc->rate, GST_SECOND);
+ enc = GST_SPEEX_ENC (benc);
+
+ /* FIXME 0.11 ? get rid of this special ogg stuff and have it
+ * put and use 'codec data' in caps like anything else,
+ * with all the usual out-of-band advantage etc */
+ if (G_UNLIKELY (enc->headers)) {
+ GSList *header = enc->headers;
+
+ /* try to push all of these, if we lose one, might as well lose all */
+ while (header) {
+ if (ret == GST_FLOW_OK)
+ ret = gst_speex_enc_push_buffer (enc, header->data);
+ else
+ gst_speex_enc_push_buffer (enc, header->data);
+ header = g_slist_next (header);
}
- }
-
- if (GST_BUFFER_TIMESTAMP_IS_VALID (buf)
- && GST_BUFFER_DURATION_IS_VALID (buf))
- enc->next_ts = GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf);
- else
- enc->next_ts = GST_CLOCK_TIME_NONE;
- GST_DEBUG_OBJECT (enc, "received buffer of %u bytes",
- gst_buffer_get_size (buf));
-
- /* push buffer to adapter */
- gst_adapter_push (enc->adapter, buf);
- buf = NULL;
-
- ret = gst_speex_enc_encode (enc, FALSE);
-
-done:
-
- if (buf)
- gst_buffer_unref (buf);
-
- return ret;
-
- /* ERRORS */
-not_setup:
- {
- GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
- ("encoder not initialized (input is not audio?)"));
- ret = GST_FLOW_NOT_NEGOTIATED;
- goto done;
+ g_slist_free (enc->headers);
+ enc->headers = NULL;
}
+ return ret;
}
-
static void
gst_speex_enc_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
break;
}
}
-
-static GstStateChangeReturn
-gst_speex_enc_change_state (GstElement * element, GstStateChange transition)
-{
- GstSpeexEnc *enc = GST_SPEEX_ENC (element);
- GstStateChangeReturn res;
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- enc->tags = gst_tag_list_new ();
- break;
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- speex_bits_init (&enc->bits);
- enc->frameno = 0;
- enc->frameno_out = 0;
- enc->samples_in = 0;
- enc->start_ts = GST_CLOCK_TIME_NONE;
- enc->next_ts = GST_CLOCK_TIME_NONE;
- enc->granulepos_offset = 0;
- break;
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- /* fall through */
- default:
- break;
- }
-
- res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
- if (res == GST_STATE_CHANGE_FAILURE)
- return res;
-
- switch (transition) {
- case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
- break;
- case GST_STATE_CHANGE_PAUSED_TO_READY:
- enc->setup = FALSE;
- enc->header_sent = FALSE;
- if (enc->state) {
- speex_encoder_destroy (enc->state);
- enc->state = NULL;
- }
- speex_bits_destroy (&enc->bits);
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
- gst_tag_list_free (enc->tags);
- enc->tags = NULL;
- default:
- break;
- }
-
- return res;
-}
projects / platform / upstream / gstreamer.git / blobdiff