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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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lame: port to audioencoder
This commit is contained in:
Mark Nauwelaerts
parent
7961d3f2e3
commit
e33c98bc24
2 changed files with 102 additions and 230 deletions
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@ -21,7 +21,7 @@
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/**
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* SECTION:element-lame
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* @see_also: lamemp3enc, mad, vorbisenc
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* @see_also: lame, mad, vorbisenc
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*
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* This element encodes raw integer audio into an MPEG-1 layer 3 (MP3) stream.
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* Note that <ulink url="http://en.wikipedia.org/wiki/MP3">MP3</ulink> is not
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@ -31,7 +31,7 @@
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*
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* <refsect2>
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* <title>Note</title>
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* This element is deprecated, use the lamemp3enc element instead
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* This element is deprecated, use the lame element instead
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* which provides a much simpler interface and results in better MP3 files.
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* </refsect2>
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*
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@ -309,15 +309,19 @@ static void gst_lame_base_init (gpointer g_class);
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static void gst_lame_class_init (GstLameClass * klass);
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static void gst_lame_init (GstLame * gst_lame);
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static gboolean gst_lame_start (GstAudioEncoder * enc);
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static gboolean gst_lame_stop (GstAudioEncoder * enc);
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static gboolean gst_lame_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_lame_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static void gst_lame_flush (GstAudioEncoder * enc);
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static void gst_lame_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_lame_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_lame_sink_event (GstPad * pad, GstEvent * event);
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static GstFlowReturn gst_lame_chain (GstPad * pad, GstBuffer * buf);
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static gboolean gst_lame_setup (GstLame * lame);
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static GstStateChangeReturn gst_lame_change_state (GstElement * element,
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GstStateChange transition);
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static GstElementClass *parent_class = NULL;
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@ -352,7 +356,8 @@ gst_lame_get_type (void)
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};
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gst_lame_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstLame", &gst_lame_info, 0);
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g_type_register_static (GST_TYPE_AUDIO_ENCODER, "GstLame",
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&gst_lame_info, 0);
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g_type_add_interface_static (gst_lame_type, GST_TYPE_TAG_SETTER,
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&tag_setter_info);
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g_type_add_interface_static (gst_lame_type, GST_TYPE_PRESET, &preset_info);
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@ -397,9 +402,11 @@ gst_lame_class_init (GstLameClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstAudioEncoderClass *base_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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base_class = (GstAudioEncoderClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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@ -407,6 +414,12 @@ gst_lame_class_init (GstLameClass * klass)
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gobject_class->get_property = gst_lame_get_property;
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gobject_class->finalize = gst_lame_finalize;
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base_class->start = GST_DEBUG_FUNCPTR (gst_lame_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_lame_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_lame_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_lame_handle_frame);
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base_class->flush = GST_DEBUG_FUNCPTR (gst_lame_flush);
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g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_BITRATE,
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g_param_spec_int ("bitrate", "Bitrate (kb/s)",
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"Bitrate in kbit/sec (8, 16, 24, 32, 40, 48, 56, 64, 80, 96, "
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@ -565,39 +578,30 @@ gst_lame_class_init (GstLameClass * klass)
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GST_TYPE_LAME_PRESET, gst_lame_default_settings.preset,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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#endif
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gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_lame_change_state);
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}
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static gboolean
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gst_lame_src_setcaps (GstPad * pad, GstCaps * caps)
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{
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GST_DEBUG_OBJECT (pad, "caps: %" GST_PTR_FORMAT, caps);
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return TRUE;
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}
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static gboolean
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gst_lame_sink_setcaps (GstPad * pad, GstCaps * caps)
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gst_lame_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
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{
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GstLame *lame;
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gint out_samplerate;
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gint version;
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GstStructure *structure;
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GstCaps *othercaps;
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GstClockTime latency;
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lame = GST_LAME (GST_PAD_PARENT (pad));
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structure = gst_caps_get_structure (caps, 0);
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lame = GST_LAME (enc);
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if (!gst_structure_get_int (structure, "rate", &lame->samplerate))
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goto no_rate;
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if (!gst_structure_get_int (structure, "channels", &lame->num_channels))
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goto no_channels;
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/* parameters already parsed for us */
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lame->samplerate = GST_AUDIO_INFO_RATE (info);
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lame->num_channels = GST_AUDIO_INFO_CHANNELS (info);
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/* but we might be asked to reconfigure, so reset */
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gst_lame_release_memory (lame);
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GST_DEBUG_OBJECT (lame, "setting up lame");
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if (!gst_lame_setup (lame))
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goto setup_failed;
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out_samplerate = lame_get_out_samplerate (lame->lgf);
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if (out_samplerate == 0)
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goto zero_output_rate;
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@ -624,21 +628,18 @@ gst_lame_sink_setcaps (GstPad * pad, GstCaps * caps)
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"rate", G_TYPE_INT, out_samplerate, NULL);
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/* and use these caps */
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gst_pad_set_caps (lame->srcpad, othercaps);
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gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (lame), othercaps);
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gst_caps_unref (othercaps);
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/* base class feedback:
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* - we will handle buffers, just hand us all available
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* - report latency */
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latency = gst_util_uint64_scale_int (lame_get_framesize (lame->lgf),
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GST_SECOND, lame->samplerate);
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gst_audio_encoder_set_latency (enc, latency, latency);
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return TRUE;
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no_rate:
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{
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GST_ERROR_OBJECT (lame, "input caps have no sample rate field");
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return FALSE;
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}
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no_channels:
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{
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GST_ERROR_OBJECT (lame, "input caps have no channels field");
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return FALSE;
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}
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zero_output_rate:
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{
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GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS, (NULL),
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@ -658,26 +659,6 @@ gst_lame_init (GstLame * lame)
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{
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GST_DEBUG_OBJECT (lame, "starting initialization");
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lame->sinkpad =
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gst_pad_new_from_static_template (&gst_lame_sink_template, "sink");
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gst_pad_set_event_function (lame->sinkpad,
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GST_DEBUG_FUNCPTR (gst_lame_sink_event));
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gst_pad_set_chain_function (lame->sinkpad,
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GST_DEBUG_FUNCPTR (gst_lame_chain));
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gst_pad_set_setcaps_function (lame->sinkpad,
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GST_DEBUG_FUNCPTR (gst_lame_sink_setcaps));
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gst_element_add_pad (GST_ELEMENT (lame), lame->sinkpad);
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lame->srcpad =
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gst_pad_new_from_static_template (&gst_lame_src_template, "src");
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gst_pad_set_setcaps_function (lame->srcpad,
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GST_DEBUG_FUNCPTR (gst_lame_src_setcaps));
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gst_element_add_pad (GST_ELEMENT (lame), lame->srcpad);
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lame->samplerate = 44100;
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lame->num_channels = 2;
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lame->setup = FALSE;
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/* Set default settings */
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lame->bitrate = gst_lame_default_settings.bitrate;
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lame->compression_ratio = gst_lame_default_settings.compression_ratio;
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@ -714,6 +695,27 @@ gst_lame_init (GstLame * lame)
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GST_DEBUG_OBJECT (lame, "done initializing");
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}
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static gboolean
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gst_lame_start (GstAudioEncoder * enc)
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{
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GstLame *lame = GST_LAME (enc);
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GST_DEBUG_OBJECT (lame, "start");
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return TRUE;
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}
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static gboolean
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gst_lame_stop (GstAudioEncoder * enc)
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{
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GstLame *lame = GST_LAME (enc);
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GST_DEBUG_OBJECT (lame, "stop");
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gst_lame_release_memory (lame);
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return TRUE;
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}
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/* <php-emulation-mode>three underscores for ___rate is really really really
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* private as opposed to one underscore<php-emulation-mode> */
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/* call this MACRO outside of the NULL state so that we have a higher chance
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@ -979,108 +981,54 @@ gst_lame_get_property (GObject * object, guint prop_id, GValue * value,
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}
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}
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static gboolean
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gst_lame_sink_event (GstPad * pad, GstEvent * event)
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static GstFlowReturn
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gst_lame_flush_full (GstLame * lame, gboolean push)
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{
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gboolean ret;
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GstLame *lame;
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GstBuffer *buf;
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gint size;
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GstFlowReturn result = GST_FLOW_OK;
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lame = GST_LAME (gst_pad_get_parent (pad));
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if (!lame->lgf)
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return GST_FLOW_OK;
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switch (GST_EVENT_TYPE (event)) {
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case GST_EVENT_EOS:{
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GST_DEBUG_OBJECT (lame, "handling EOS event");
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buf = gst_buffer_new_and_alloc (7200);
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size = lame_encode_flush (lame->lgf, GST_BUFFER_DATA (buf), 7200);
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if (lame->lgf != NULL) {
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GstBuffer *buf;
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gint size;
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buf = gst_buffer_new_and_alloc (7200);
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size = lame_encode_flush (lame->lgf, GST_BUFFER_DATA (buf), 7200);
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if (size > 0 && lame->last_flow == GST_FLOW_OK) {
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gint64 duration;
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duration = gst_util_uint64_scale (size, 8 * GST_SECOND,
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1000 * lame->bitrate);
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if (lame->last_ts == GST_CLOCK_TIME_NONE) {
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lame->last_ts = lame->eos_ts;
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lame->last_duration = duration;
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} else {
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lame->last_duration += duration;
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}
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GST_BUFFER_TIMESTAMP (buf) = lame->last_ts;
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GST_BUFFER_DURATION (buf) = lame->last_duration;
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lame->last_ts = GST_CLOCK_TIME_NONE;
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GST_BUFFER_SIZE (buf) = size;
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GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
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gst_buffer_set_caps (buf, GST_PAD_CAPS (lame->srcpad));
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gst_pad_push (lame->srcpad, buf);
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} else {
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GST_DEBUG_OBJECT (lame, "no final packet (size=%d, last_flow=%s)",
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size, gst_flow_get_name (lame->last_flow));
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gst_buffer_unref (buf);
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}
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}
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ret = gst_pad_event_default (pad, event);
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break;
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}
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case GST_EVENT_FLUSH_START:
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GST_DEBUG_OBJECT (lame, "handling FLUSH start event");
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/* forward event */
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ret = gst_pad_push_event (lame->srcpad, event);
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break;
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case GST_EVENT_FLUSH_STOP:
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{
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guchar *mp3_data = NULL;
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gint mp3_buffer_size;
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GST_DEBUG_OBJECT (lame, "handling FLUSH stop event");
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if (lame->lgf) {
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/* clear buffers if we already have lame set up */
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mp3_buffer_size = 7200;
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mp3_data = g_malloc (mp3_buffer_size);
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lame_encode_flush (lame->lgf, mp3_data, mp3_buffer_size);
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g_free (mp3_data);
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}
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ret = gst_pad_push_event (lame->srcpad, event);
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break;
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}
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case GST_EVENT_TAG:
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GST_DEBUG_OBJECT (lame, "ignoring TAG event, passing it on");
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ret = gst_pad_push_event (lame->srcpad, event);
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break;
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default:
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ret = gst_pad_event_default (pad, event);
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break;
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if (size > 0 && push) {
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GST_BUFFER_SIZE (buf) = size;
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GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
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result = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame), buf, -1);
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} else {
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GST_DEBUG_OBJECT (lame, "no final packet (size=%d, push=%d)", size, push);
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gst_buffer_unref (buf);
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result = GST_FLOW_OK;
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}
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gst_object_unref (lame);
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return ret;
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return result;
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}
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static void
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gst_lame_flush (GstAudioEncoder * enc)
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{
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gst_lame_flush_full (GST_LAME (enc), FALSE);
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}
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static GstFlowReturn
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gst_lame_chain (GstPad * pad, GstBuffer * buf)
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gst_lame_handle_frame (GstAudioEncoder * enc, GstBuffer * buf)
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{
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GstLame *lame;
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guchar *mp3_data;
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GstBuffer *mp3_buf;
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gint mp3_buffer_size, mp3_size;
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gint64 duration;
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GstFlowReturn result;
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gint num_samples;
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guint8 *data;
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guint size;
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lame = GST_LAME (GST_PAD_PARENT (pad));
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lame = GST_LAME (enc);
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GST_LOG_OBJECT (lame, "entered chain");
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if (!lame->setup)
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goto not_setup;
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/* squeeze remaining and push */
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if (G_UNLIKELY (buf == NULL))
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return gst_lame_flush_full (lame, TRUE);
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data = GST_BUFFER_DATA (buf);
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size = GST_BUFFER_SIZE (buf);
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@ -1089,7 +1037,8 @@ gst_lame_chain (GstPad * pad, GstBuffer * buf)
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/* allocate space for output */
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mp3_buffer_size = 1.25 * num_samples + 7200;
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mp3_data = g_malloc (mp3_buffer_size);
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mp3_buf = gst_buffer_new_and_alloc (mp3_buffer_size);
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mp3_data = GST_BUFFER_DATA (mp3_buf);
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/* lame seems to be too stupid to get mono interleaved going */
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if (lame->num_channels == 1) {
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@ -1105,69 +1054,23 @@ gst_lame_chain (GstPad * pad, GstBuffer * buf)
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GST_LOG_OBJECT (lame, "encoded %d bytes of audio to %d bytes of mp3",
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size, mp3_size);
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duration = gst_util_uint64_scale_int (size, GST_SECOND,
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2 * lame->samplerate * lame->num_channels);
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if (GST_BUFFER_DURATION (buf) != GST_CLOCK_TIME_NONE &&
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GST_BUFFER_DURATION (buf) != duration) {
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GST_DEBUG_OBJECT (lame, "incoming buffer had incorrect duration %"
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GST_TIME_FORMAT ", outgoing buffer will have correct duration %"
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GST_TIME_FORMAT,
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GST_TIME_ARGS (GST_BUFFER_DURATION (buf)), GST_TIME_ARGS (duration));
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}
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if (lame->last_ts == GST_CLOCK_TIME_NONE) {
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lame->last_ts = GST_BUFFER_TIMESTAMP (buf);
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lame->last_offs = GST_BUFFER_OFFSET (buf);
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lame->last_duration = duration;
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} else {
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lame->last_duration += duration;
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}
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gst_buffer_unref (buf);
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if (mp3_size < 0) {
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g_warning ("error %d", mp3_size);
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}
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if (mp3_size > 0) {
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GstBuffer *outbuf;
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outbuf = gst_buffer_new ();
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GST_BUFFER_DATA (outbuf) = mp3_data;
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GST_BUFFER_MALLOCDATA (outbuf) = mp3_data;
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GST_BUFFER_SIZE (outbuf) = mp3_size;
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GST_BUFFER_TIMESTAMP (outbuf) = lame->last_ts;
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GST_BUFFER_OFFSET (outbuf) = lame->last_offs;
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GST_BUFFER_DURATION (outbuf) = lame->last_duration;
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gst_buffer_set_caps (outbuf, GST_PAD_CAPS (lame->srcpad));
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result = gst_pad_push (lame->srcpad, outbuf);
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lame->last_flow = result;
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if (result != GST_FLOW_OK) {
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GST_DEBUG_OBJECT (lame, "flow return: %s", gst_flow_get_name (result));
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}
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if (GST_CLOCK_TIME_IS_VALID (lame->last_ts))
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lame->eos_ts = lame->last_ts + lame->last_duration;
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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 */
|
||||
|
|
@ -1204,7 +1107,7 @@ gst_lame_setup (GstLame * lame)
|
|||
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;
|
||||
|
|
@ -1294,37 +1197,6 @@ gst_lame_setup (GstLame * lame)
|
|||
#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)
|
||||
{
|
||||
|
|
|
|||
|
|
@ -27,6 +27,7 @@
|
|||
G_BEGIN_DECLS
|
||||
|
||||
#include <lame/lame.h>
|
||||
#include <gst/audio/gstaudioencoder.h>
|
||||
|
||||
#define GST_TYPE_LAME \
|
||||
(gst_lame_get_type())
|
||||
|
|
@ -48,10 +49,9 @@ typedef struct _GstLameClass GstLameClass;
|
|||
* Opaque data structure.
|
||||
*/
|
||||
struct _GstLame {
|
||||
GstElement element;
|
||||
GstAudioEncoder element;
|
||||
|
||||
/*< private >*/
|
||||
GstPad *srcpad, *sinkpad;
|
||||
|
||||
gint samplerate;
|
||||
gint num_channels;
|
||||
|
|
@ -100,7 +100,7 @@ struct _GstLame {
|
|||
};
|
||||
|
||||
struct _GstLameClass {
|
||||
GstElementClass parent_class;
|
||||
GstAudioEncoderClass parent_class;
|
||||
};
|
||||
|
||||
GType gst_lame_get_type(void);
|
||||
|
|
|
|||
Loading…
Reference in a new issue