/**
* SECTION:element-lame
- * @see_also: lamemp3enc, mad, vorbisenc
+ * @see_also: lame, mad, vorbisenc
*
* This element encodes raw integer audio into an MPEG-1 layer 3 (MP3) stream.
* Note that <ulink url="http://en.wikipedia.org/wiki/MP3">MP3</ulink> is not
*
* <refsect2>
* <title>Note</title>
- * This element is deprecated, use the lamemp3enc element instead
+ * This element is deprecated, use the lame element instead
* which provides a much simpler interface and results in better MP3 files.
* </refsect2>
*
static void gst_lame_class_init (GstLameClass * klass);
static void gst_lame_init (GstLame * gst_lame);
+static gboolean gst_lame_start (GstAudioEncoder * enc);
+static gboolean gst_lame_stop (GstAudioEncoder * enc);
+static gboolean gst_lame_set_format (GstAudioEncoder * enc,
+ GstAudioInfo * info);
+static GstFlowReturn gst_lame_handle_frame (GstAudioEncoder * enc,
+ GstBuffer * in_buf);
+static void gst_lame_flush (GstAudioEncoder * enc);
+
static void gst_lame_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_lame_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
-static gboolean gst_lame_sink_event (GstPad * pad, GstEvent * event);
-static GstFlowReturn gst_lame_chain (GstPad * pad, GstBuffer * buf);
static gboolean gst_lame_setup (GstLame * lame);
-static GstStateChangeReturn gst_lame_change_state (GstElement * element,
- GstStateChange transition);
static GstElementClass *parent_class = NULL;
};
gst_lame_type =
- g_type_register_static (GST_TYPE_ELEMENT, "GstLame", &gst_lame_info, 0);
+ g_type_register_static (GST_TYPE_AUDIO_ENCODER, "GstLame",
+ &gst_lame_info, 0);
g_type_add_interface_static (gst_lame_type, GST_TYPE_TAG_SETTER,
&tag_setter_info);
g_type_add_interface_static (gst_lame_type, GST_TYPE_PRESET, &preset_info);
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
+ GstAudioEncoderClass *base_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
+ base_class = (GstAudioEncoderClass *) klass;
parent_class = g_type_class_peek_parent (klass);
gobject_class->get_property = gst_lame_get_property;
gobject_class->finalize = gst_lame_finalize;
+ base_class->start = GST_DEBUG_FUNCPTR (gst_lame_start);
+ base_class->stop = GST_DEBUG_FUNCPTR (gst_lame_stop);
+ base_class->set_format = GST_DEBUG_FUNCPTR (gst_lame_set_format);
+ base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_lame_handle_frame);
+ base_class->flush = GST_DEBUG_FUNCPTR (gst_lame_flush);
+
g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BITRATE,
g_param_spec_int ("bitrate", "Bitrate (kb/s)",
"Bitrate in kbit/sec (8, 16, 24, 32, 40, 48, 56, 64, 80, 96, "
GST_TYPE_LAME_PRESET, gst_lame_default_settings.preset,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
#endif
-
- gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_lame_change_state);
-}
-
-static gboolean
-gst_lame_src_setcaps (GstPad * pad, GstCaps * caps)
-{
- GST_DEBUG_OBJECT (pad, "caps: %" GST_PTR_FORMAT, caps);
- return TRUE;
}
static gboolean
-gst_lame_sink_setcaps (GstPad * pad, GstCaps * caps)
+gst_lame_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
{
GstLame *lame;
gint out_samplerate;
gint version;
- GstStructure *structure;
GstCaps *othercaps;
+ GstClockTime latency;
+
+ lame = GST_LAME (enc);
- lame = GST_LAME (GST_PAD_PARENT (pad));
- structure = gst_caps_get_structure (caps, 0);
+ /* parameters already parsed for us */
+ lame->samplerate = GST_AUDIO_INFO_RATE (info);
+ lame->num_channels = GST_AUDIO_INFO_CHANNELS (info);
- if (!gst_structure_get_int (structure, "rate", &lame->samplerate))
- goto no_rate;
- if (!gst_structure_get_int (structure, "channels", &lame->num_channels))
- goto no_channels;
+ /* but we might be asked to reconfigure, so reset */
+ gst_lame_release_memory (lame);
GST_DEBUG_OBJECT (lame, "setting up lame");
if (!gst_lame_setup (lame))
goto setup_failed;
-
out_samplerate = lame_get_out_samplerate (lame->lgf);
if (out_samplerate == 0)
goto zero_output_rate;
"rate", G_TYPE_INT, out_samplerate, NULL);
/* and use these caps */
- gst_pad_set_caps (lame->srcpad, othercaps);
+ gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (lame), othercaps);
gst_caps_unref (othercaps);
+ /* base class feedback:
+ * - we will handle buffers, just hand us all available
+ * - report latency */
+ latency = gst_util_uint64_scale_int (lame_get_framesize (lame->lgf),
+ GST_SECOND, lame->samplerate);
+ gst_audio_encoder_set_latency (enc, latency, latency);
+
return TRUE;
-no_rate:
- {
- GST_ERROR_OBJECT (lame, "input caps have no sample rate field");
- return FALSE;
- }
-no_channels:
- {
- GST_ERROR_OBJECT (lame, "input caps have no channels field");
- return FALSE;
- }
zero_output_rate:
{
GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS, (NULL),
{
GST_DEBUG_OBJECT (lame, "starting initialization");
- lame->sinkpad =
- gst_pad_new_from_static_template (&gst_lame_sink_template, "sink");
- gst_pad_set_event_function (lame->sinkpad,
- GST_DEBUG_FUNCPTR (gst_lame_sink_event));
- gst_pad_set_chain_function (lame->sinkpad,
- GST_DEBUG_FUNCPTR (gst_lame_chain));
- gst_pad_set_setcaps_function (lame->sinkpad,
- GST_DEBUG_FUNCPTR (gst_lame_sink_setcaps));
- gst_element_add_pad (GST_ELEMENT (lame), lame->sinkpad);
-
- lame->srcpad =
- gst_pad_new_from_static_template (&gst_lame_src_template, "src");
- gst_pad_set_setcaps_function (lame->srcpad,
- GST_DEBUG_FUNCPTR (gst_lame_src_setcaps));
- gst_element_add_pad (GST_ELEMENT (lame), lame->srcpad);
-
- lame->samplerate = 44100;
- lame->num_channels = 2;
- lame->setup = FALSE;
-
/* Set default settings */
lame->bitrate = gst_lame_default_settings.bitrate;
lame->compression_ratio = gst_lame_default_settings.compression_ratio;
GST_DEBUG_OBJECT (lame, "done initializing");
}
+static gboolean
+gst_lame_start (GstAudioEncoder * enc)
+{
+ GstLame *lame = GST_LAME (enc);
+
+ GST_DEBUG_OBJECT (lame, "start");
+ return TRUE;
+}
+
+static gboolean
+gst_lame_stop (GstAudioEncoder * enc)
+{
+ GstLame *lame = GST_LAME (enc);
+
+ GST_DEBUG_OBJECT (lame, "stop");
+
+ gst_lame_release_memory (lame);
+ return TRUE;
+}
+
+
/* <php-emulation-mode>three underscores for ___rate is really really really
* private as opposed to one underscore<php-emulation-mode> */
/* call this MACRO outside of the NULL state so that we have a higher chance
}
}
-static gboolean
-gst_lame_sink_event (GstPad * pad, GstEvent * event)
+static GstFlowReturn
+gst_lame_flush_full (GstLame * lame, gboolean push)
{
- gboolean ret;
- GstLame *lame;
-
- lame = GST_LAME (gst_pad_get_parent (pad));
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_EOS:{
- GST_DEBUG_OBJECT (lame, "handling EOS event");
-
- if (lame->lgf != NULL) {
- GstBuffer *buf;
- gint size;
-
- buf = gst_buffer_new_and_alloc (7200);
- size = lame_encode_flush (lame->lgf, GST_BUFFER_DATA (buf), 7200);
-
- if (size > 0 && lame->last_flow == GST_FLOW_OK) {
- gint64 duration;
-
- duration = gst_util_uint64_scale (size, 8 * GST_SECOND,
- 1000 * lame->bitrate);
-
- if (lame->last_ts == GST_CLOCK_TIME_NONE) {
- lame->last_ts = lame->eos_ts;
- lame->last_duration = duration;
- } else {
- lame->last_duration += duration;
- }
-
- GST_BUFFER_TIMESTAMP (buf) = lame->last_ts;
- GST_BUFFER_DURATION (buf) = lame->last_duration;
- lame->last_ts = GST_CLOCK_TIME_NONE;
- GST_BUFFER_SIZE (buf) = size;
- GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
- gst_buffer_set_caps (buf, GST_PAD_CAPS (lame->srcpad));
- gst_pad_push (lame->srcpad, buf);
- } else {
- GST_DEBUG_OBJECT (lame, "no final packet (size=%d, last_flow=%s)",
- size, gst_flow_get_name (lame->last_flow));
- gst_buffer_unref (buf);
- }
- }
-
- ret = gst_pad_event_default (pad, event);
- break;
- }
- case GST_EVENT_FLUSH_START:
- GST_DEBUG_OBJECT (lame, "handling FLUSH start event");
- /* forward event */
- ret = gst_pad_push_event (lame->srcpad, event);
- break;
- case GST_EVENT_FLUSH_STOP:
- {
- guchar *mp3_data = NULL;
- gint mp3_buffer_size;
+ GstBuffer *buf;
+ gint size;
+ GstFlowReturn result = GST_FLOW_OK;
- GST_DEBUG_OBJECT (lame, "handling FLUSH stop event");
+ if (!lame->lgf)
+ return GST_FLOW_OK;
- if (lame->lgf) {
- /* clear buffers if we already have lame set up */
- mp3_buffer_size = 7200;
- mp3_data = g_malloc (mp3_buffer_size);
- lame_encode_flush (lame->lgf, mp3_data, mp3_buffer_size);
- g_free (mp3_data);
- }
+ buf = gst_buffer_new_and_alloc (7200);
+ size = lame_encode_flush (lame->lgf, GST_BUFFER_DATA (buf), 7200);
- ret = gst_pad_push_event (lame->srcpad, event);
- break;
- }
- case GST_EVENT_TAG:
- GST_DEBUG_OBJECT (lame, "ignoring TAG event, passing it on");
- ret = gst_pad_push_event (lame->srcpad, event);
- break;
- default:
- ret = gst_pad_event_default (pad, event);
- break;
+ if (size > 0 && push) {
+ GST_BUFFER_SIZE (buf) = size;
+ GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
+ result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame), buf, -1);
+ } else {
+ GST_DEBUG_OBJECT (lame, "no final packet (size=%d, push=%d)", size, push);
+ gst_buffer_unref (buf);
+ result = GST_FLOW_OK;
}
- gst_object_unref (lame);
- return ret;
+ return result;
+}
+
+static void
+gst_lame_flush (GstAudioEncoder * enc)
+{
+ gst_lame_flush_full (GST_LAME (enc), FALSE);
}
static GstFlowReturn
-gst_lame_chain (GstPad * pad, GstBuffer * buf)
+gst_lame_handle_frame (GstAudioEncoder * enc, GstBuffer * buf)
{
GstLame *lame;
guchar *mp3_data;
+ GstBuffer *mp3_buf;
gint mp3_buffer_size, mp3_size;
- gint64 duration;
GstFlowReturn result;
gint num_samples;
guint8 *data;
guint size;
- lame = GST_LAME (GST_PAD_PARENT (pad));
-
- GST_LOG_OBJECT (lame, "entered chain");
+ lame = GST_LAME (enc);
- if (!lame->setup)
- goto not_setup;
+ /* squeeze remaining and push */
+ if (G_UNLIKELY (buf == NULL))
+ return gst_lame_flush_full (lame, TRUE);
data = GST_BUFFER_DATA (buf);
size = GST_BUFFER_SIZE (buf);
/* allocate space for output */
mp3_buffer_size = 1.25 * num_samples + 7200;
- mp3_data = g_malloc (mp3_buffer_size);
+ mp3_buf = gst_buffer_new_and_alloc (mp3_buffer_size);
+ mp3_data = GST_BUFFER_DATA (mp3_buf);
/* lame seems to be too stupid to get mono interleaved going */
if (lame->num_channels == 1) {
GST_LOG_OBJECT (lame, "encoded %d bytes of audio to %d bytes of mp3",
size, mp3_size);
- duration = gst_util_uint64_scale_int (size, GST_SECOND,
- 2 * lame->samplerate * lame->num_channels);
-
- if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE &&
- GST_BUFFER_DURATION (buf) != duration) {
- GST_DEBUG_OBJECT (lame, "incoming buffer had incorrect duration %"
- GST_TIME_FORMAT ", outgoing buffer will have correct duration %"
- GST_TIME_FORMAT,
- GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_TIME_ARGS (duration));
- }
-
- if (lame->last_ts == GST_CLOCK_TIME_NONE) {
- lame->last_ts = GST_BUFFER_TIMESTAMP (buf);
- lame->last_offs = GST_BUFFER_OFFSET (buf);
- lame->last_duration = duration;
- } else {
- lame->last_duration += duration;
- }
-
- gst_buffer_unref (buf);
-
if (mp3_size < 0) {
g_warning ("error %d", mp3_size);
}
- if (mp3_size > 0) {
- GstBuffer *outbuf;
-
- outbuf = gst_buffer_new ();
- GST_BUFFER_DATA (outbuf) = mp3_data;
- GST_BUFFER_MALLOCDATA (outbuf) = mp3_data;
- GST_BUFFER_SIZE (outbuf) = mp3_size;
- GST_BUFFER_TIMESTAMP (outbuf) = lame->last_ts;
- GST_BUFFER_OFFSET (outbuf) = lame->last_offs;
- GST_BUFFER_DURATION (outbuf) = lame->last_duration;
- gst_buffer_set_caps (outbuf, GST_PAD_CAPS (lame->srcpad));
-
- result = gst_pad_push (lame->srcpad, outbuf);
- lame->last_flow = result;
- if (result != GST_FLOW_OK) {
- GST_DEBUG_OBJECT (lame, "flow return: %s", gst_flow_get_name (result));
- }
-
- if (GST_CLOCK_TIME_IS_VALID (lame->last_ts))
- lame->eos_ts = lame->last_ts + lame->last_duration;
- else
- lame->eos_ts = GST_CLOCK_TIME_NONE;
- lame->last_ts = GST_CLOCK_TIME_NONE;
+ if (G_LIKELY (mp3_size > 0)) {
+ GST_BUFFER_SIZE (mp3_buf) = mp3_size;
+ result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
} else {
- g_free (mp3_data);
+ if (mp3_size < 0) {
+ /* eat error ? */
+ g_warning ("error %d", mp3_size);
+ }
result = GST_FLOW_OK;
+ gst_buffer_unref (mp3_buf);
}
return result;
-
- /* ERRORS */
-not_setup:
- {
- gst_buffer_unref (buf);
- GST_ELEMENT_ERROR (lame, CORE, NEGOTIATION, (NULL),
- ("encoder not initialized (input is not audio?)"));
- return GST_FLOW_ERROR;
- }
}
/* set up the encoder state */
lame_set_in_samplerate (lame->lgf, lame->samplerate);
/* let lame choose default samplerate unless outgoing sample rate is fixed */
- allowed_caps = gst_pad_get_allowed_caps (lame->srcpad);
+ allowed_caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (lame));
if (allowed_caps != NULL) {
GstStructure *structure;
#undef CHECK_ERROR
}
-static GstStateChangeReturn
-gst_lame_change_state (GstElement * element, GstStateChange transition)
-{
- GstLame *lame;
- GstStateChangeReturn result;
-
- lame = GST_LAME (element);
-
- switch (transition) {
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- lame->last_flow = GST_FLOW_OK;
- lame->last_ts = GST_CLOCK_TIME_NONE;
- lame->eos_ts = GST_CLOCK_TIME_NONE;
- break;
- default:
- break;
- }
-
- result = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
- switch (transition) {
- case GST_STATE_CHANGE_READY_TO_NULL:
- gst_lame_release_memory (lame);
- break;
- default:
- break;
- }
-
- return result;
-}
-
static gboolean
gst_lame_get_default_settings (void)
{
projects / platform / upstream / gst-plugins-good.git / commitdiff