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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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lame: port to 0.11
This commit is contained in:
Wim Taymans
parent
036963bc89
commit
ee98ebc46a
2 changed files with 62 additions and 72 deletions
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@ -1,6 +1,6 @@
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/* GStreamer
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* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
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* Copyright (C) <2004> Wim Taymans <wim@fluendo.com>
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* Copyright (C) <2004> Wim Taymans <wim.taymans@gmail.com>
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* Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
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*
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* This library is free software; you can redistribute it and/or
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@ -319,18 +319,8 @@ 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_setup (GstLame * lame);
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static void
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gst_lame_add_interfaces (GType lame_type)
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{
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static const GInterfaceInfo tag_setter_info = { NULL, NULL, NULL };
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/* FIXME: remove support for the GstTagSetter interface in 0.11 */
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g_type_add_interface_static (lame_type, GST_TYPE_TAG_SETTER,
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&tag_setter_info);
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}
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GST_BOILERPLATE_FULL (GstLame, gst_lame, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER, gst_lame_add_interfaces);
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#define gst_lame_parent_class parent_class
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G_DEFINE_TYPE (GstLame, gst_lame, GST_TYPE_AUDIO_ENCODER);
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static void
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gst_lame_release_memory (GstLame * lame)
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@ -349,34 +339,32 @@ gst_lame_finalize (GObject * obj)
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G_OBJECT_CLASS (parent_class)->finalize (obj);
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}
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static void
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gst_lame_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_lame_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_lame_sink_template));
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gst_element_class_set_details_simple (element_class, "L.A.M.E. mp3 encoder",
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"Codec/Encoder/Audio",
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"High-quality free MP3 encoder (deprecated)",
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"Erik Walthinsen <omega@cse.ogi.edu>, " "Wim Taymans <wim@fluendo.com>");
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}
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static void
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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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gobject_class->set_property = gst_lame_set_property;
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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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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_lame_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_lame_sink_template));
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gst_element_class_set_details_simple (gstelement_class,
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"L.A.M.E. mp3 encoder", "Codec/Encoder/Audio",
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"High-quality free MP3 encoder (deprecated)",
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"Erik Walthinsen <omega@cse.ogi.edu>, "
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"Wim Taymans <wim.taymans@gmail.com>");
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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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@ -618,7 +606,7 @@ setup_failed:
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}
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static void
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gst_lame_init (GstLame * lame, GstLameClass * klass)
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gst_lame_init (GstLame * lame)
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{
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GST_DEBUG_OBJECT (lame, "starting initialization");
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@ -948,6 +936,7 @@ static GstFlowReturn
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gst_lame_flush_full (GstLame * lame, gboolean push)
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{
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GstBuffer *buf;
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guint8 *data;
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gint size;
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GstFlowReturn result = GST_FLOW_OK;
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@ -955,10 +944,11 @@ gst_lame_flush_full (GstLame * lame, gboolean push)
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return GST_FLOW_OK;
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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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data = gst_buffer_map (buf, NULL, NULL, GST_MAP_WRITE);
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size = lame_encode_flush (lame->lgf, data, 7200);
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gst_buffer_unmap (buf, data, size);
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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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@ -985,7 +975,7 @@ gst_lame_handle_frame (GstAudioEncoder * enc, GstBuffer * buf)
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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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gsize size;
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lame = GST_LAME (enc);
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@ -993,15 +983,14 @@ gst_lame_handle_frame (GstAudioEncoder * enc, GstBuffer * buf)
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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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data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
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num_samples = size / 2;
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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_buf = gst_buffer_new_and_alloc (mp3_buffer_size);
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mp3_data = GST_BUFFER_DATA (mp3_buf);
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mp3_buf = gst_buffer_new_allocate (NULL, mp3_buffer_size, 0);
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mp3_data = gst_buffer_map (mp3_buf, NULL, NULL, GST_MAP_WRITE);
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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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@ -1013,18 +1002,16 @@ gst_lame_handle_frame (GstAudioEncoder * enc, GstBuffer * buf)
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(short int *) data,
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num_samples / lame->num_channels, mp3_data, mp3_buffer_size);
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}
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gst_buffer_unmap (buf, data, size);
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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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if (mp3_size < 0) {
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g_warning ("error %d", mp3_size);
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}
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if (G_LIKELY (mp3_size > 0)) {
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GST_BUFFER_SIZE (mp3_buf) = mp3_size;
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gst_buffer_unmap (mp3_buf, mp3_data, mp3_size);
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result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
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} else {
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gst_buffer_unmap (mp3_buf, mp3_data, 0);
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if (mp3_size < 0) {
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/* eat error ? */
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g_warning ("error %d", mp3_size);
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@ -1032,7 +1019,6 @@ gst_lame_handle_frame (GstAudioEncoder * enc, GstBuffer * buf)
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result = GST_FLOW_OK;
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gst_buffer_unref (mp3_buf);
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}
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return result;
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}
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@ -190,8 +190,8 @@ static void gst_lamemp3enc_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_lamemp3enc_setup (GstLameMP3Enc * lame, GstTagList ** tags);
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GST_BOILERPLATE (GstLameMP3Enc, gst_lamemp3enc, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER);
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#define gst_lamemp3enc_parent_class parent_class
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G_DEFINE_TYPE (GstLameMP3Enc, gst_lamemp3enc, GST_TYPE_AUDIO_ENCODER);
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static void
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gst_lamemp3enc_release_memory (GstLameMP3Enc * lame)
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@ -210,34 +210,31 @@ gst_lamemp3enc_finalize (GObject * obj)
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G_OBJECT_CLASS (parent_class)->finalize (obj);
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}
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static void
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gst_lamemp3enc_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_lamemp3enc_src_template));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&gst_lamemp3enc_sink_template));
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gst_element_class_set_details_simple (element_class, "L.A.M.E. mp3 encoder",
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"Codec/Encoder/Audio",
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"High-quality free MP3 encoder",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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}
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static void
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gst_lamemp3enc_class_init (GstLameMP3EncClass * 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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gobject_class->set_property = gst_lamemp3enc_set_property;
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gobject_class->get_property = gst_lamemp3enc_get_property;
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gobject_class->finalize = gst_lamemp3enc_finalize;
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_lamemp3enc_src_template));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&gst_lamemp3enc_sink_template));
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gst_element_class_set_details_simple (gstelement_class,
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"L.A.M.E. mp3 encoder", "Codec/Encoder/Audio",
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"High-quality free MP3 encoder",
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"Sebastian Dröge <sebastian.droege@collabora.co.uk>");
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base_class->start = GST_DEBUG_FUNCPTR (gst_lamemp3enc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_lamemp3enc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_lamemp3enc_set_format);
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@ -279,7 +276,7 @@ gst_lamemp3enc_class_init (GstLameMP3EncClass * klass)
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}
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static void
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gst_lamemp3enc_init (GstLameMP3Enc * lame, GstLameMP3EncClass * klass)
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gst_lamemp3enc_init (GstLameMP3Enc * lame)
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{
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}
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@ -491,19 +488,25 @@ gst_lamemp3enc_flush_full (GstLameMP3Enc * lame, gboolean push)
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{
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GstBuffer *buf;
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gint size;
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guint8 *data;
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GstFlowReturn result = GST_FLOW_OK;
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if (!lame->lgf)
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return GST_FLOW_OK;
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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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data = gst_buffer_map (buf, NULL, NULL, GST_MAP_WRITE);
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size = lame_encode_flush (lame->lgf, data, 7200);
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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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if (size > 0) {
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gst_buffer_unmap (buf, data, size);
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if (push) {
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GST_DEBUG_OBJECT (lame, "pushing final packet of %u bytes", size);
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result =
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gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (lame), buf, -1);
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}
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} else {
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gst_buffer_unmap (buf, data, 0);
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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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@ -527,7 +530,7 @@ gst_lamemp3enc_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
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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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gsize size;
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lame = GST_LAMEMP3ENC (enc);
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@ -535,15 +538,14 @@ gst_lamemp3enc_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
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if (G_UNLIKELY (in_buf == NULL))
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return gst_lamemp3enc_flush_full (lame, TRUE);
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data = GST_BUFFER_DATA (in_buf);
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size = GST_BUFFER_SIZE (in_buf);
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data = gst_buffer_map (in_buf, &size, NULL, GST_MAP_READ);
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num_samples = size / 2;
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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_buf = gst_buffer_new_and_alloc (mp3_buffer_size);
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mp3_data = GST_BUFFER_DATA (mp3_buf);
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mp3_buf = gst_buffer_new_allocate (NULL, mp3_buffer_size, 0);
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mp3_data = gst_buffer_map (mp3_buf, NULL, NULL, GST_MAP_WRITE);
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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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@ -555,20 +557,22 @@ gst_lamemp3enc_handle_frame (GstAudioEncoder * enc, GstBuffer * in_buf)
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(short int *) data,
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num_samples / lame->num_channels, mp3_data, mp3_buffer_size);
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}
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gst_buffer_unmap (in_buf, data, size);
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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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if (G_LIKELY (mp3_size > 0)) {
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GST_BUFFER_SIZE (mp3_buf) = mp3_size;
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gst_buffer_unmap (mp3_buf, mp3_data, mp3_size);
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result = gst_audio_encoder_finish_frame (enc, mp3_buf, -1);
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} else {
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gst_buffer_unmap (mp3_buf, mp3_data, 0);
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if (mp3_size < 0) {
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/* eat error ? */
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g_warning ("error %d", mp3_size);
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}
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result = GST_FLOW_OK;
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gst_buffer_unref (mp3_buf);
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result = GST_FLOW_OK;
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}
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return result;
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