--- /dev/null
+/* GStreamer
+ * Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
+ * Copyright (C) <2004> Wim Taymans <wim@fluendo.com>
+ * Copyright (C) <2005> Thomas Vander Stichele <thomas at apestaart dot org>
+ * Copyright (C) <2009> Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
+ * Boston, MA 02111-1307, USA.
+ */
+
+/**
+ * SECTION:element-lamemp3enc
+ * @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
+ * a free format, there are licensing and patent issues to take into
+ * consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink>
+ * for a royalty free (and often higher quality) alternative.
+ *
+ * <refsect2>
+ * <title>Output sample rate</title>
+ * If no fixed output sample rate is negotiated on the element's src pad,
+ * the element will choose an optimal sample rate to resample to internally.
+ * For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will
+ * get resampled to 32 KHz. Use filter caps on the src pad to force a
+ * particular sample rate.
+ * </refsect2>
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3
+ * ]| Encode a test sine signal to MP3.
+ * |[
+ * gst-launch -v alsasrc ! audioconvert ! lamemp3enc bitrate=192 ! filesink location=alsasrc.mp3
+ * ]| Record from a sound card using ALSA and encode to MP3
+ * |[
+ * gst-launch -v filesrc location=music.wav ! decodebin ! audioconvert ! audioresample ! lamemp3enc bitrate=192 ! id3v2mux ! filesink location=music.mp3
+ * ]| Transcode from a .wav file to MP3 (the id3v2mux element is optional)
+ * |[
+ * gst-launch -v cdda://5 ! audioconvert ! lamemp3enc bitrate=192 ! filesink location=track5.mp3
+ * ]| Encode Audio CD track 5 to MP3
+ * |[
+ * gst-launch -v audiotestsrc num-buffers=10 ! audio/x-raw-int,rate=44100,channels=1 ! lamemp3enc bitrate=48 ! filesink location=test.mp3
+ * ]| Encode to a fixed sample rate
+ * </refsect2>
+ *
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <string.h>
+#include "gstlamemp3enc.h"
+#include <gst/gst-i18n-plugin.h>
+
+GST_DEBUG_CATEGORY_STATIC (debug);
+#define GST_CAT_DEFAULT debug
+
+/* elementfactory information */
+
+/* LAMEMP3ENC can do MPEG-1, MPEG-2, and MPEG-2.5, so it has 9 possible
+ * sample rates it supports */
+static GstStaticPadTemplate gst_lamemp3enc_sink_template =
+GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw-int, "
+ "endianness = (int) " G_STRINGIFY (G_BYTE_ORDER) ", "
+ "signed = (boolean) true, "
+ "width = (int) 16, "
+ "depth = (int) 16, "
+ "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
+ "channels = (int) [ 1, 2 ]")
+ );
+
+static GstStaticPadTemplate gst_lamemp3enc_src_template =
+GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/mpeg, "
+ "mpegversion = (int) 1, "
+ "layer = (int) 3, "
+ "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }, "
+ "channels = (int) [ 1, 2 ]")
+ );
+
+/********** Define useful types for non-programmatic interfaces **********/
+enum
+{
+ LAMEMP3ENC_TARGET_QUALITY = 0,
+ LAMEMP3ENC_TARGET_BITRATE
+};
+
+#define GST_TYPE_LAMEMP3ENC_TARGET (gst_lamemp3enc_target_get_type())
+static GType
+gst_lamemp3enc_target_get_type (void)
+{
+ static GType lame_target_type = 0;
+ static GEnumValue lame_targets[] = {
+ {LAMEMP3ENC_TARGET_QUALITY, "Quality", "quality"},
+ {LAMEMP3ENC_TARGET_BITRATE, "Bitrate", "bitrate"},
+ {0, NULL, NULL}
+ };
+
+ if (!lame_target_type) {
+ lame_target_type =
+ g_enum_register_static ("GstLameMP3EncTarget", lame_targets);
+ }
+ return lame_target_type;
+}
+
+enum
+{
+ LAMEMP3ENC_ENCODING_ENGINE_QUALITY_FAST = 0,
+ LAMEMP3ENC_ENCODING_ENGINE_QUALITY_STANDARD,
+ LAMEMP3ENC_ENCODING_ENGINE_QUALITY_HIGH
+};
+
+#define GST_TYPE_LAMEMP3ENC_ENCODING_ENGINE_QUALITY (gst_lamemp3enc_encoding_engine_quality_get_type())
+static GType
+gst_lamemp3enc_encoding_engine_quality_get_type (void)
+{
+ static GType lame_encoding_engine_quality_type = 0;
+ static GEnumValue lame_encoding_engine_quality[] = {
+ {0, "Fast", "fast"},
+ {1, "Standard", "standard"},
+ {2, "High", "high"},
+ {0, NULL, NULL}
+ };
+
+ if (!lame_encoding_engine_quality_type) {
+ lame_encoding_engine_quality_type =
+ g_enum_register_static ("GstLameMP3EncEncodingEngineQuality",
+ lame_encoding_engine_quality);
+ }
+ return lame_encoding_engine_quality_type;
+}
+
+/********** Standard stuff for signals and arguments **********/
+
+enum
+{
+ ARG_0,
+ ARG_TARGET,
+ ARG_BITRATE,
+ ARG_CBR,
+ ARG_VBR_QUALITY,
+ ARG_FAST_VBR,
+ ARG_ENCODING_ENGINE_QUALITY,
+ ARG_MONO
+};
+
+#define DEFAULT_TARGET LAMEMP3ENC_TARGET_QUALITY
+#define DEFAULT_BITRATE 128
+#define DEFAULT_CBR FALSE
+#define DEFAULT_VBR_QUALITY 4
+#define DEFAULT_FAST_VBR FALSE
+#define DEFAULT_ENCODING_ENGINE_QUALITY LAMEMP3ENC_ENCODING_ENGINE_QUALITY_STANDARD
+#define DEFAULT_MONO FALSE
+
+static void gst_lamemp3enc_base_init (gpointer g_class);
+static void gst_lamemp3enc_class_init (GstLameMP3EncClass * klass);
+static void gst_lamemp3enc_init (GstLameMP3Enc * gst_lame);
+
+static void gst_lamemp3enc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
+static void gst_lamemp3enc_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static gboolean gst_lamemp3enc_sink_event (GstPad * pad, GstEvent * event);
+static GstFlowReturn gst_lamemp3enc_chain (GstPad * pad, GstBuffer * buf);
+static gboolean gst_lamemp3enc_setup (GstLameMP3Enc * lame);
+static GstStateChangeReturn gst_lamemp3enc_change_state (GstElement * element,
+ GstStateChange transition);
+
+static GstElementClass *parent_class = NULL;
+
+GType
+gst_lamemp3enc_get_type (void)
+{
+ static GType gst_lamemp3enc_type = 0;
+
+ if (!gst_lamemp3enc_type) {
+ static const GTypeInfo gst_lamemp3enc_info = {
+ sizeof (GstLameMP3EncClass),
+ gst_lamemp3enc_base_init,
+ NULL,
+ (GClassInitFunc) gst_lamemp3enc_class_init,
+ NULL,
+ NULL,
+ sizeof (GstLameMP3Enc),
+ 0,
+ (GInstanceInitFunc) gst_lamemp3enc_init,
+ };
+
+ gst_lamemp3enc_type =
+ g_type_register_static (GST_TYPE_ELEMENT, "GstLameMP3Enc",
+ &gst_lamemp3enc_info, 0);
+ }
+ return gst_lamemp3enc_type;
+}
+
+static void
+gst_lamemp3enc_release_memory (GstLameMP3Enc * lame)
+{
+ if (lame->lgf) {
+ lame_close (lame->lgf);
+ lame->lgf = NULL;
+ }
+}
+
+static void
+gst_lamemp3enc_finalize (GObject * obj)
+{
+ gst_lamemp3enc_release_memory (GST_LAMEMP3ENC (obj));
+
+ G_OBJECT_CLASS (parent_class)->finalize (obj);
+}
+
+static void
+gst_lamemp3enc_base_init (gpointer g_class)
+{
+ GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_lamemp3enc_src_template));
+ gst_element_class_add_pad_template (element_class,
+ gst_static_pad_template_get (&gst_lamemp3enc_sink_template));
+ gst_element_class_set_details_simple (element_class, "L.A.M.E. mp3 encoder",
+ "Codec/Encoder/Audio",
+ "High-quality free MP3 encoder",
+ "Sebastian Dröge <sebastian.droege@collabora.co.uk>");
+}
+
+static void
+gst_lamemp3enc_class_init (GstLameMP3EncClass * klass)
+{
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+
+ parent_class = g_type_class_peek_parent (klass);
+
+ gobject_class->set_property = gst_lamemp3enc_set_property;
+ gobject_class->get_property = gst_lamemp3enc_get_property;
+ gobject_class->finalize = gst_lamemp3enc_finalize;
+
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_TARGET,
+ g_param_spec_enum ("target", "Target",
+ "Optimize for quality or bitrate", GST_TYPE_LAMEMP3ENC_TARGET,
+ DEFAULT_TARGET, G_PARAM_READWRITE));
+ 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, "
+ "112, 128, 160, 192, 224, 256 or 320)",
+ 8, 320, DEFAULT_BITRATE, G_PARAM_READWRITE));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_CBR,
+ g_param_spec_boolean ("cbr", "CBR", "Enforce constant bitrate encoding",
+ DEFAULT_CBR, G_PARAM_READWRITE));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_VBR_QUALITY,
+ g_param_spec_float ("vbr-quality", "VBR Quality",
+ "VBR Quality from 0 to 10, 0 being the best", 0.0, 9.999,
+ DEFAULT_VBR_QUALITY, G_PARAM_READWRITE));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_FAST_VBR,
+ g_param_spec_boolean ("fast-vbr", "Fast VBR", "Use fast VBR encoding",
+ DEFAULT_FAST_VBR, G_PARAM_READWRITE));
+ g_object_class_install_property (G_OBJECT_CLASS (klass),
+ ARG_ENCODING_ENGINE_QUALITY, g_param_spec_enum ("encoding-engine-quality",
+ "Encoding Engine Quality", "Quality/speed of the encoding engine",
+ GST_TYPE_LAMEMP3ENC_ENCODING_ENGINE_QUALITY,
+ DEFAULT_ENCODING_ENGINE_QUALITY, G_PARAM_READWRITE));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), ARG_MONO,
+ g_param_spec_boolean ("mono", "Mono", "Enforce mono encoding",
+ DEFAULT_MONO, G_PARAM_READWRITE));
+
+ gstelement_class->change_state =
+ GST_DEBUG_FUNCPTR (gst_lamemp3enc_change_state);
+}
+
+static gboolean
+gst_lamemp3enc_src_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GST_DEBUG_OBJECT (pad, "caps: %" GST_PTR_FORMAT, caps);
+ return TRUE;
+}
+
+static gboolean
+gst_lamemp3enc_sink_setcaps (GstPad * pad, GstCaps * caps)
+{
+ GstLameMP3Enc *lame;
+ gint out_samplerate;
+ gint version;
+ GstStructure *structure;
+ GstCaps *othercaps;
+
+ lame = GST_LAMEMP3ENC (GST_PAD_PARENT (pad));
+ structure = gst_caps_get_structure (caps, 0);
+
+ 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;
+
+ GST_DEBUG_OBJECT (lame, "setting up lame");
+ if (!gst_lamemp3enc_setup (lame))
+ goto setup_failed;
+
+
+ out_samplerate = lame_get_out_samplerate (lame->lgf);
+ if (out_samplerate == 0)
+ goto zero_output_rate;
+ if (out_samplerate != lame->samplerate) {
+ GST_WARNING_OBJECT (lame,
+ "output samplerate %d is different from incoming samplerate %d",
+ out_samplerate, lame->samplerate);
+ }
+
+ version = lame_get_version (lame->lgf);
+ if (version == 0)
+ version = 2;
+ else if (version == 1)
+ version = 1;
+ else if (version == 2)
+ version = 3;
+
+ othercaps =
+ gst_caps_new_simple ("audio/mpeg",
+ "mpegversion", G_TYPE_INT, 1,
+ "mpegaudioversion", G_TYPE_INT, version,
+ "layer", G_TYPE_INT, 3,
+ "channels", G_TYPE_INT, lame->mono ? 1 : lame->num_channels,
+ "rate", G_TYPE_INT, out_samplerate, NULL);
+
+ /* and use these caps */
+ gst_pad_set_caps (lame->srcpad, othercaps);
+ gst_caps_unref (othercaps);
+
+ 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),
+ ("LAMEMP3ENC decided on a zero sample rate"));
+ return FALSE;
+ }
+setup_failed:
+ {
+ GST_ELEMENT_ERROR (lame, LIBRARY, SETTINGS,
+ (_("Failed to configure LAMEMP3ENC encoder. Check your encoding parameters.")), (NULL));
+ return FALSE;
+ }
+}
+
+static void
+gst_lamemp3enc_init (GstLameMP3Enc * lame)
+{
+ GST_DEBUG_OBJECT (lame, "starting initialization");
+
+ lame->sinkpad =
+ gst_pad_new_from_static_template (&gst_lamemp3enc_sink_template, "sink");
+ gst_pad_set_event_function (lame->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_lamemp3enc_sink_event));
+ gst_pad_set_chain_function (lame->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_lamemp3enc_chain));
+ gst_pad_set_setcaps_function (lame->sinkpad,
+ GST_DEBUG_FUNCPTR (gst_lamemp3enc_sink_setcaps));
+ gst_element_add_pad (GST_ELEMENT (lame), lame->sinkpad);
+
+ lame->srcpad =
+ gst_pad_new_from_static_template (&gst_lamemp3enc_src_template, "src");
+ gst_pad_set_setcaps_function (lame->srcpad,
+ GST_DEBUG_FUNCPTR (gst_lamemp3enc_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->target = DEFAULT_TARGET;
+ lame->bitrate = DEFAULT_BITRATE;
+ lame->cbr = DEFAULT_CBR;
+ lame->vbr_quality = DEFAULT_VBR_QUALITY;
+ lame->fast_vbr = DEFAULT_FAST_VBR;
+ lame->encoding_engine_quality = DEFAULT_ENCODING_ENGINE_QUALITY;
+ lame->mono = DEFAULT_MONO;
+
+ GST_DEBUG_OBJECT (lame, "done initializing");
+}
+
+/* <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
+ * of actually having a pipeline and bus to get the message through */
+
+#define CHECK_AND_FIXUP_BITRATE(obj,param,rate) \
+G_STMT_START { \
+ gint ___rate = rate; \
+ gint maxrate = 320; \
+ gint multiplier = 64; \
+ if (rate == 0) { \
+ ___rate = rate; \
+ } else if (rate <= 64) { \
+ maxrate = 64; multiplier = 8; \
+ if ((rate % 8) != 0) ___rate = GST_ROUND_UP_8 (rate); \
+ } else if (rate <= 128) { \
+ maxrate = 128; multiplier = 16; \
+ if ((rate % 16) != 0) ___rate = GST_ROUND_UP_16 (rate); \
+ } else if (rate <= 256) { \
+ maxrate = 256; multiplier = 32; \
+ if ((rate % 32) != 0) ___rate = GST_ROUND_UP_32 (rate); \
+ } else if (rate <= 320) { \
+ maxrate = 320; multiplier = 64; \
+ if ((rate % 64) != 0) ___rate = GST_ROUND_UP_64 (rate); \
+ } \
+ if (___rate != rate) { \
+ GST_ELEMENT_WARNING (obj, LIBRARY, SETTINGS, \
+ (_("The requested bitrate %d kbit/s for property '%s' " \
+ "is not allowed. " \
+ "The bitrate was changed to %d kbit/s."), rate, \
+ param, ___rate), \
+ ("A bitrate below %d should be a multiple of %d.", \
+ maxrate, multiplier)); \
+ rate = ___rate; \
+ } \
+} G_STMT_END
+
+static void
+gst_lamemp3enc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstLameMP3Enc *lame;
+
+ lame = GST_LAMEMP3ENC (object);
+
+ switch (prop_id) {
+ case ARG_TARGET:
+ lame->target = g_value_get_enum (value);
+ break;
+ case ARG_BITRATE:
+ lame->bitrate = g_value_get_int (value);
+ break;
+ case ARG_CBR:
+ lame->cbr = g_value_get_boolean (value);
+ break;
+ case ARG_VBR_QUALITY:
+ lame->vbr_quality = g_value_get_float (value);
+ break;
+ case ARG_FAST_VBR:
+ lame->fast_vbr = g_value_get_boolean (value);
+ break;
+ case ARG_ENCODING_ENGINE_QUALITY:
+ lame->encoding_engine_quality = g_value_get_enum (value);
+ break;
+ case ARG_MONO:
+ lame->mono = g_value_get_boolean (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static void
+gst_lamemp3enc_get_property (GObject * object, guint prop_id, GValue * value,
+ GParamSpec * pspec)
+{
+ GstLameMP3Enc *lame;
+
+ lame = GST_LAMEMP3ENC (object);
+
+ switch (prop_id) {
+ case ARG_TARGET:
+ g_value_set_enum (value, lame->target);
+ break;
+ case ARG_BITRATE:
+ g_value_set_int (value, lame->bitrate);
+ break;
+ case ARG_CBR:
+ g_value_set_boolean (value, lame->cbr);
+ break;
+ case ARG_VBR_QUALITY:
+ g_value_set_float (value, lame->vbr_quality);
+ break;
+ case ARG_FAST_VBR:
+ g_value_set_boolean (value, lame->fast_vbr);
+ break;
+ case ARG_ENCODING_ENGINE_QUALITY:
+ g_value_set_enum (value, lame->encoding_engine_quality);
+ break;
+ case ARG_MONO:
+ g_value_set_boolean (value, lame->mono);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
+}
+
+static gboolean
+gst_lamemp3enc_sink_event (GstPad * pad, GstEvent * event)
+{
+ gboolean ret;
+ GstLameMP3Enc *lame;
+
+ lame = GST_LAMEMP3ENC (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;
+
+ GST_DEBUG_OBJECT (lame, "handling FLUSH stop event");
+
+ /* clear buffers */
+ mp3_buffer_size = 7200;
+ mp3_data = g_malloc (mp3_buffer_size);
+ lame_encode_flush (lame->lgf, mp3_data, mp3_buffer_size);
+
+ 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;
+ }
+ gst_object_unref (lame);
+ return ret;
+}
+
+static GstFlowReturn
+gst_lamemp3enc_chain (GstPad * pad, GstBuffer * buf)
+{
+ GstLameMP3Enc *lame;
+ guchar *mp3_data;
+ gint mp3_buffer_size, mp3_size;
+ gint64 duration;
+ GstFlowReturn result;
+ gint num_samples;
+ guint8 *data;
+ guint size;
+
+ lame = GST_LAMEMP3ENC (GST_PAD_PARENT (pad));
+
+ GST_LOG_OBJECT (lame, "entered chain");
+
+ if (!lame->setup)
+ goto not_setup;
+
+ data = GST_BUFFER_DATA (buf);
+ size = GST_BUFFER_SIZE (buf);
+
+ num_samples = size / 2;
+
+ /* allocate space for output */
+ mp3_buffer_size = 1.25 * num_samples + 7200;
+ mp3_data = g_malloc (mp3_buffer_size);
+
+ /* lame seems to be too stupid to get mono interleaved going */
+ if (lame->num_channels == 1) {
+ mp3_size = lame_encode_buffer (lame->lgf,
+ (short int *) data,
+ (short int *) data, num_samples, mp3_data, mp3_buffer_size);
+ } else {
+ mp3_size = lame_encode_buffer_interleaved (lame->lgf,
+ (short int *) data,
+ num_samples / lame->num_channels, mp3_data, mp3_buffer_size);
+ }
+
+ 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;
+ } else {
+ g_free (mp3_data);
+ result = GST_FLOW_OK;
+ }
+
+ 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 */
+static gboolean
+gst_lamemp3enc_setup (GstLameMP3Enc * lame)
+{
+
+#define CHECK_ERROR(command) G_STMT_START {\
+ if ((command) < 0) { \
+ GST_ERROR_OBJECT (lame, "setup failed: " G_STRINGIFY (command)); \
+ return FALSE; \
+ } \
+}G_STMT_END
+
+ int retval;
+ GstCaps *allowed_caps;
+
+ GST_DEBUG_OBJECT (lame, "starting setup");
+
+ /* check if we're already setup; if we are, we might want to check
+ * if this initialization is compatible with the previous one */
+ /* FIXME: do this */
+ if (lame->setup) {
+ GST_WARNING_OBJECT (lame, "already setup");
+ lame->setup = FALSE;
+ }
+
+ lame->lgf = lame_init ();
+
+ if (lame->lgf == NULL)
+ return FALSE;
+
+ /* copy the parameters over */
+ 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);
+
+ if (allowed_caps != NULL) {
+ GstStructure *structure;
+ gint samplerate;
+
+ structure = gst_caps_get_structure (allowed_caps, 0);
+
+ if (gst_structure_get_int (structure, "rate", &samplerate)) {
+ GST_DEBUG_OBJECT (lame, "Setting sample rate to %d as fixed in src caps",
+ samplerate);
+ lame_set_out_samplerate (lame->lgf, samplerate);
+ } else {
+ GST_DEBUG_OBJECT (lame, "Letting lame choose sample rate");
+ lame_set_out_samplerate (lame->lgf, 0);
+ }
+ gst_caps_unref (allowed_caps);
+ allowed_caps = NULL;
+ } else {
+ GST_DEBUG_OBJECT (lame, "No peer yet, letting lame choose sample rate");
+ lame_set_out_samplerate (lame->lgf, 0);
+ }
+
+ CHECK_ERROR (lame_set_num_channels (lame->lgf, lame->num_channels));
+
+ if (lame->target == LAMEMP3ENC_TARGET_QUALITY) {
+ if (lame->fast_vbr)
+ CHECK_ERROR (lame_set_VBR (lame->lgf, vbr_rh));
+ else
+ CHECK_ERROR (lame_set_VBR (lame->lgf, vbr_mtrh));
+ CHECK_ERROR (lame_set_VBR_quality (lame->lgf, lame->vbr_quality));
+ } else {
+ CHECK_AND_FIXUP_BITRATE (lame, "bitrate", lame->bitrate);
+ if (lame->cbr) {
+ CHECK_ERROR (lame_set_VBR (lame->lgf, vbr_off));
+ CHECK_ERROR (lame_set_brate (lame->lgf, lame->bitrate));
+ } else {
+ CHECK_ERROR (lame_set_VBR (lame->lgf, vbr_abr));
+ CHECK_ERROR (lame_set_VBR_mean_bitrate_kbps (lame->lgf, lame->bitrate));
+ }
+ }
+
+ if (lame->encoding_engine_quality == LAMEMP3ENC_ENCODING_ENGINE_QUALITY_FAST)
+ CHECK_ERROR (lame_set_quality (lame->lgf, 7));
+ else if (lame->encoding_engine_quality ==
+ LAMEMP3ENC_ENCODING_ENGINE_QUALITY_HIGH)
+ CHECK_ERROR (lame_set_quality (lame->lgf, 2));
+ /* else default */
+
+ if (lame->mono)
+ CHECK_ERROR (lame_set_mode (lame->lgf, MONO));
+
+ /* initialize the lame encoder */
+ if ((retval = lame_init_params (lame->lgf)) >= 0) {
+ lame->setup = TRUE;
+ /* FIXME: it would be nice to print out the mode here */
+ GST_INFO ("lame encoder setup (%d Hz, %d channels)",
+ lame->bitrate, lame->samplerate, lame->num_channels);
+ } else {
+ GST_ERROR_OBJECT (lame, "lame_init_params returned %d", retval);
+ }
+
+ GST_DEBUG_OBJECT (lame, "done with setup");
+
+ return lame->setup;
+#undef CHECK_ERROR
+}
+
+static GstStateChangeReturn
+gst_lamemp3enc_change_state (GstElement * element, GstStateChange transition)
+{
+ GstLameMP3Enc *lame;
+ GstStateChangeReturn result;
+
+ lame = GST_LAMEMP3ENC (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_lamemp3enc_release_memory (lame);
+ break;
+ default:
+ break;
+ }
+
+ return result;
+}
+
+gboolean
+gst_lamemp3enc_register (GstPlugin * plugin)
+{
+ GST_DEBUG_CATEGORY_INIT (debug, "lamemp3enc", 0, "lame mp3 encoder");
+
+ if (!gst_element_register (plugin, "lamemp3enc", GST_RANK_NONE,
+ GST_TYPE_LAMEMP3ENC))
+ return FALSE;
+
+ return TRUE;
+}
projects / platform / upstream / gstreamer.git / commitdiff