-/* GStreamer
+/* GStreamer Speex Encoder
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* This library is free software; you can redistribute it and/or
* Boston, MA 02111-1307, USA.
*/
+/**
+ * SECTION:element-speexenc
+ * @see_also: speexdec, oggmux
+ *
+ * This element encodes audio as a Speex stream.
+ * <ulink url="http://www.speex.org/">Speex</ulink> is a royalty-free
+ * audio codec maintained by the <ulink url="http://www.xiph.org/">Xiph.org
+ * Foundation</ulink>.
+ *
+ * <refsect2>
+ * <title>Example pipelines</title>
+ * |[
+ * gst-launch audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
+ * ]| Encode an Ogg/Speex file.
+ * </refsect2>
+ */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
+#include <stdlib.h>
#include <string.h>
-
+#include <time.h>
+#include <math.h>
+#include <speex/speex.h>
+#include <speex/speex_stereo.h>
+
+#include <gst/gsttagsetter.h>
+#include <gst/tag/tag.h>
+#include <gst/audio/audio.h>
#include "gstspeexenc.h"
-/* elementfactory information */
-GstElementDetails gst_speexenc_details = {
- "speex audio encoder",
- "Codec/Encoder/Audio",
- ".speex",
- "Wim Taymans <wim.taymans@chello.be>",
-};
+GST_DEBUG_CATEGORY_STATIC (speexenc_debug);
+#define GST_CAT_DEFAULT speexenc_debug
+
+#define FORMAT_STR GST_AUDIO_NE(S16)
+
+static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
+ GST_PAD_SINK,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " FORMAT_STR ", "
+ "rate = (int) [ 6000, 48000 ], " "channels = (int) [ 1, 2 ]")
+ );
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+ GST_PAD_SRC,
+ GST_PAD_ALWAYS,
+ GST_STATIC_CAPS ("audio/x-speex, "
+ "rate = (int) [ 6000, 48000 ], " "channels = (int) [ 1, 2]")
+ );
+
+#define DEFAULT_QUALITY 8.0
+#define DEFAULT_BITRATE 0
+#define DEFAULT_MODE GST_SPEEX_ENC_MODE_AUTO
+#define DEFAULT_VBR FALSE
+#define DEFAULT_ABR 0
+#define DEFAULT_VAD FALSE
+#define DEFAULT_DTX FALSE
+#define DEFAULT_COMPLEXITY 3
+#define DEFAULT_NFRAMES 1
-/* SpeexEnc signals and args */
enum
{
- FRAME_ENCODED,
- /* FILL ME */
- LAST_SIGNAL
+ PROP_0,
+ PROP_QUALITY,
+ PROP_BITRATE,
+ PROP_MODE,
+ PROP_VBR,
+ PROP_ABR,
+ PROP_VAD,
+ PROP_DTX,
+ PROP_COMPLEXITY,
+ PROP_NFRAMES,
+ PROP_LAST_MESSAGE
};
-enum
+#define GST_TYPE_SPEEX_ENC_MODE (gst_speex_enc_mode_get_type())
+static GType
+gst_speex_enc_mode_get_type (void)
{
- ARG_0,
- /* FILL ME */
-};
+ static GType speex_enc_mode_type = 0;
+ static const GEnumValue speex_enc_modes[] = {
+ {GST_SPEEX_ENC_MODE_AUTO, "Auto", "auto"},
+ {GST_SPEEX_ENC_MODE_UWB, "Ultra Wide Band", "uwb"},
+ {GST_SPEEX_ENC_MODE_WB, "Wide Band", "wb"},
+ {GST_SPEEX_ENC_MODE_NB, "Narrow Band", "nb"},
+ {0, NULL, NULL},
+ };
+ if (G_UNLIKELY (speex_enc_mode_type == 0)) {
+ speex_enc_mode_type = g_enum_register_static ("GstSpeexEncMode",
+ speex_enc_modes);
+ }
+ return speex_enc_mode_type;
+}
+
+static void gst_speex_enc_finalize (GObject * object);
+
+static gboolean gst_speex_enc_setup (GstSpeexEnc * enc);
-static void gst_speexenc_base_init (gpointer g_class);
-static void gst_speexenc_class_init (GstSpeexEnc * klass);
-static void gst_speexenc_init (GstSpeexEnc * speexenc);
+static void gst_speex_enc_get_property (GObject * object, guint prop_id,
+ GValue * value, GParamSpec * pspec);
+static void gst_speex_enc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec);
-static void gst_speexenc_chain (GstPad * pad, GstData * _data);
-static GstPadLinkReturn gst_speexenc_sinkconnect (GstPad * pad,
- const GstCaps * caps);
+static GstFlowReturn gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf);
-static GstElementClass *parent_class = NULL;
-static guint gst_speexenc_signals[LAST_SIGNAL] = { 0 };
+static gboolean gst_speex_enc_start (GstAudioEncoder * enc);
+static gboolean gst_speex_enc_stop (GstAudioEncoder * enc);
+static gboolean gst_speex_enc_set_format (GstAudioEncoder * enc,
+ GstAudioInfo * info);
+static GstFlowReturn gst_speex_enc_handle_frame (GstAudioEncoder * enc,
+ GstBuffer * in_buf);
+static gboolean gst_speex_enc_sink_event (GstAudioEncoder * enc,
+ GstEvent * event);
+static GstFlowReturn
+gst_speex_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer);
-GType
-gst_speexenc_get_type (void)
+#define gst_speex_enc_parent_class parent_class
+G_DEFINE_TYPE_WITH_CODE (GstSpeexEnc, gst_speex_enc, GST_TYPE_AUDIO_ENCODER,
+ G_IMPLEMENT_INTERFACE (GST_TYPE_TAG_SETTER, NULL);
+ G_IMPLEMENT_INTERFACE (GST_TYPE_PRESET, NULL));
+
+static void
+gst_speex_enc_class_init (GstSpeexEncClass * klass)
{
- static GType speexenc_type = 0;
-
- if (!speexenc_type) {
- static const GTypeInfo speexenc_info = {
- sizeof (GstSpeexEncClass),
- gst_speexenc_base_init,
- NULL,
- (GClassInitFunc) gst_speexenc_class_init,
- NULL,
- NULL,
- sizeof (GstSpeexEnc),
- 0,
- (GInstanceInitFunc) gst_speexenc_init,
- };
-
- speexenc_type =
- g_type_register_static (GST_TYPE_ELEMENT, "GstSpeexEnc", &speexenc_info,
- 0);
- }
- return speexenc_type;
+ GObjectClass *gobject_class;
+ GstElementClass *gstelement_class;
+ GstAudioEncoderClass *base_class;
+
+ gobject_class = (GObjectClass *) klass;
+ gstelement_class = (GstElementClass *) klass;
+ base_class = (GstAudioEncoderClass *) klass;
+
+ gobject_class->finalize = gst_speex_enc_finalize;
+ gobject_class->set_property = gst_speex_enc_set_property;
+ gobject_class->get_property = gst_speex_enc_get_property;
+
+ base_class->start = GST_DEBUG_FUNCPTR (gst_speex_enc_start);
+ base_class->stop = GST_DEBUG_FUNCPTR (gst_speex_enc_stop);
+ base_class->set_format = GST_DEBUG_FUNCPTR (gst_speex_enc_set_format);
+ base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_speex_enc_handle_frame);
+ base_class->event = GST_DEBUG_FUNCPTR (gst_speex_enc_sink_event);
+ base_class->pre_push = GST_DEBUG_FUNCPTR (gst_speex_enc_pre_push);
+
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_QUALITY,
+ g_param_spec_float ("quality", "Quality", "Encoding quality",
+ 0.0, 10.0, DEFAULT_QUALITY,
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_BITRATE,
+ g_param_spec_int ("bitrate", "Encoding Bit-rate",
+ "Specify an encoding bit-rate (in bps). (0 = automatic)",
+ 0, G_MAXINT, DEFAULT_BITRATE,
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (gobject_class, PROP_MODE,
+ g_param_spec_enum ("mode", "Mode", "The encoding mode",
+ GST_TYPE_SPEEX_ENC_MODE, GST_SPEEX_ENC_MODE_AUTO,
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_VBR,
+ g_param_spec_boolean ("vbr", "VBR",
+ "Enable variable bit-rate", DEFAULT_VBR,
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_ABR,
+ g_param_spec_int ("abr", "ABR",
+ "Enable average bit-rate (0 = disabled)",
+ 0, G_MAXINT, DEFAULT_ABR,
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_VAD,
+ g_param_spec_boolean ("vad", "VAD",
+ "Enable voice activity detection", DEFAULT_VAD,
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_DTX,
+ g_param_spec_boolean ("dtx", "DTX",
+ "Enable discontinuous transmission", DEFAULT_DTX,
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_COMPLEXITY,
+ g_param_spec_int ("complexity", "Complexity",
+ "Set encoding complexity",
+ 0, G_MAXINT, DEFAULT_COMPLEXITY,
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_NFRAMES,
+ g_param_spec_int ("nframes", "NFrames",
+ "Number of frames per buffer",
+ 0, G_MAXINT, DEFAULT_NFRAMES,
+ G_PARAM_CONSTRUCT | G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+ g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_LAST_MESSAGE,
+ g_param_spec_string ("last-message", "last-message",
+ "The last status message", NULL,
+ G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&src_factory));
+ gst_element_class_add_pad_template (gstelement_class,
+ gst_static_pad_template_get (&sink_factory));
+ gst_element_class_set_details_simple (gstelement_class, "Speex audio encoder",
+ "Codec/Encoder/Audio",
+ "Encodes audio in Speex format", "Wim Taymans <wim@fluendo.com>");
+
+ GST_DEBUG_CATEGORY_INIT (speexenc_debug, "speexenc", 0, "Speex encoder");
}
-static GstStaticPadTemplate speexenc_sink_template =
-GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "endianness = (int) BYTE_ORDER, "
- "signed = (boolean) true, "
- "width = (int) 16, "
- "depth = (int) 16, "
- "rate = (int) [ 1000, 48000 ], " "channels = (int) 1")
- );
+static void
+gst_speex_enc_finalize (GObject * object)
+{
+ GstSpeexEnc *enc;
-static GstStaticPadTemplate speexenc_src_template =
-GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-speex, "
- "rate = (int) [ 1000, 48000 ], " "channels = (int) 1")
- );
+ enc = GST_SPEEX_ENC (object);
+
+ g_free (enc->last_message);
+
+ G_OBJECT_CLASS (parent_class)->finalize (object);
+}
static void
-gst_speexenc_base_init (gpointer g_class)
+gst_speex_enc_init (GstSpeexEnc * enc)
+{
+ GstAudioEncoder *benc = GST_AUDIO_ENCODER (enc);
+
+ /* arrange granulepos marking (and required perfect ts) */
+ gst_audio_encoder_set_mark_granule (benc, TRUE);
+ gst_audio_encoder_set_perfect_timestamp (benc, TRUE);
+}
+
+static gboolean
+gst_speex_enc_start (GstAudioEncoder * benc)
{
- GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+ GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&speexenc_sink_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&speexenc_src_template));
+ GST_DEBUG_OBJECT (enc, "start");
+ speex_bits_init (&enc->bits);
+ enc->tags = gst_tag_list_new_empty ();
+ enc->header_sent = FALSE;
- gst_element_class_set_details (element_class, &gst_speexenc_details);
+ return TRUE;
}
-static void
-gst_speexenc_class_init (GstSpeexEnc * klass)
+static gboolean
+gst_speex_enc_stop (GstAudioEncoder * benc)
{
- GObjectClass *gobject_class;
- GstElementClass *gstelement_class;
+ GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
- gobject_class = (GObjectClass *) klass;
- gstelement_class = (GstElementClass *) klass;
+ GST_DEBUG_OBJECT (enc, "stop");
+ enc->header_sent = FALSE;
+ if (enc->state) {
+ speex_encoder_destroy (enc->state);
+ enc->state = NULL;
+ }
+ speex_bits_destroy (&enc->bits);
+ gst_tag_list_free (enc->tags);
+ enc->tags = NULL;
+ g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL);
+ enc->headers = NULL;
+
+ gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
+
+ return TRUE;
+}
- parent_class = g_type_class_ref (GST_TYPE_ELEMENT);
+static gint64
+gst_speex_enc_get_latency (GstSpeexEnc * enc)
+{
+ /* See the Speex manual section "Latency and algorithmic delay" */
+ if (enc->rate == 8000)
+ return 30 * GST_MSECOND;
+ else
+ return 34 * GST_MSECOND;
+}
+
+static gboolean
+gst_speex_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
+{
+ GstSpeexEnc *enc;
+
+ enc = GST_SPEEX_ENC (benc);
+
+ enc->channels = GST_AUDIO_INFO_CHANNELS (info);
+ enc->rate = GST_AUDIO_INFO_RATE (info);
+
+ /* handle reconfigure */
+ if (enc->state) {
+ speex_encoder_destroy (enc->state);
+ enc->state = NULL;
+ }
+
+ if (!gst_speex_enc_setup (enc))
+ return FALSE;
+
+ /* feedback to base class */
+ gst_audio_encoder_set_latency (benc,
+ gst_speex_enc_get_latency (enc), gst_speex_enc_get_latency (enc));
+ gst_audio_encoder_set_lookahead (benc, enc->lookahead);
+
+ if (enc->nframes == 0) {
+ /* as many frames as available input allows */
+ gst_audio_encoder_set_frame_samples_min (benc, enc->frame_size);
+ gst_audio_encoder_set_frame_samples_max (benc, enc->frame_size);
+ gst_audio_encoder_set_frame_max (benc, 0);
+ } else {
+ /* exactly as many frames as configured */
+ gst_audio_encoder_set_frame_samples_min (benc,
+ enc->frame_size * enc->nframes);
+ gst_audio_encoder_set_frame_samples_max (benc,
+ enc->frame_size * enc->nframes);
+ gst_audio_encoder_set_frame_max (benc, 1);
+ }
- gst_speexenc_signals[FRAME_ENCODED] =
- g_signal_new ("frame-encoded", G_TYPE_FROM_CLASS (klass),
- G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSpeexEncClass, frame_encoded),
- NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0);
+ return TRUE;
}
+static GstBuffer *
+gst_speex_enc_create_metadata_buffer (GstSpeexEnc * enc)
+{
+ const GstTagList *user_tags;
+ GstTagList *merged_tags;
+ GstBuffer *comments = NULL;
+
+ user_tags = gst_tag_setter_get_tag_list (GST_TAG_SETTER (enc));
+
+ GST_DEBUG_OBJECT (enc, "upstream tags = %" GST_PTR_FORMAT, enc->tags);
+ GST_DEBUG_OBJECT (enc, "user-set tags = %" GST_PTR_FORMAT, user_tags);
+
+ /* gst_tag_list_merge() will handle NULL for either or both lists fine */
+ merged_tags = gst_tag_list_merge (user_tags, enc->tags,
+ gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (enc)));
+
+ if (merged_tags == NULL)
+ merged_tags = gst_tag_list_new_empty ();
+
+ GST_DEBUG_OBJECT (enc, "merged tags = %" GST_PTR_FORMAT, merged_tags);
+ comments = gst_tag_list_to_vorbiscomment_buffer (merged_tags, NULL,
+ 0, "Encoded with GStreamer Speexenc");
+ gst_tag_list_free (merged_tags);
+
+ GST_BUFFER_OFFSET (comments) = 0;
+ GST_BUFFER_OFFSET_END (comments) = 0;
+
+ return comments;
+}
static void
-gst_speexenc_init (GstSpeexEnc * speexenc)
+gst_speex_enc_set_last_msg (GstSpeexEnc * enc, const gchar * msg)
{
- /* create the sink and src pads */
- speexenc->sinkpad =
- gst_pad_new_from_template (gst_static_pad_template_get
- (&speexenc_sink_template), "sink");
- gst_element_add_pad (GST_ELEMENT (speexenc), speexenc->sinkpad);
- gst_pad_set_chain_function (speexenc->sinkpad, gst_speexenc_chain);
- gst_pad_set_link_function (speexenc->sinkpad, gst_speexenc_sinkconnect);
-
- speexenc->srcpad =
- gst_pad_new_from_template (gst_static_pad_template_get
- (&speexenc_src_template), "src");
- gst_element_add_pad (GST_ELEMENT (speexenc), speexenc->srcpad);
-
- speex_bits_init (&speexenc->bits);
- speexenc->mode = (SpeexMode *) & speex_nb_mode;
- speexenc->bufsize = 0;
- speexenc->packet_count = 0;
- speexenc->n_packets = 20;
+ g_free (enc->last_message);
+ enc->last_message = g_strdup (msg);
+ GST_WARNING_OBJECT (enc, "%s", msg);
+ g_object_notify (G_OBJECT (enc), "last-message");
}
-static GstPadLinkReturn
-gst_speexenc_sinkconnect (GstPad * pad, const GstCaps * caps)
+static gboolean
+gst_speex_enc_setup (GstSpeexEnc * enc)
{
- GstSpeexEnc *speexenc;
- GstStructure *structure;
+ switch (enc->mode) {
+ case GST_SPEEX_ENC_MODE_UWB:
+ GST_LOG_OBJECT (enc, "configuring for requested UWB mode");
+ enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_UWB);
+ break;
+ case GST_SPEEX_ENC_MODE_WB:
+ GST_LOG_OBJECT (enc, "configuring for requested WB mode");
+ enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_WB);
+ break;
+ case GST_SPEEX_ENC_MODE_NB:
+ GST_LOG_OBJECT (enc, "configuring for requested NB mode");
+ enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_NB);
+ break;
+ case GST_SPEEX_ENC_MODE_AUTO:
+ /* fall through */
+ GST_LOG_OBJECT (enc, "finding best mode");
+ default:
+ break;
+ }
- speexenc = GST_SPEEXENC (gst_pad_get_parent (pad));
+ if (enc->rate > 25000) {
+ if (enc->mode == GST_SPEEX_ENC_MODE_AUTO) {
+ GST_LOG_OBJECT (enc, "selected UWB mode for samplerate %d", enc->rate);
+ enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_UWB);
+ } else {
+ if (enc->speex_mode != speex_lib_get_mode (SPEEX_MODEID_UWB)) {
+ gst_speex_enc_set_last_msg (enc,
+ "Warning: suggest to use ultra wide band mode for this rate");
+ }
+ }
+ } else if (enc->rate > 12500) {
+ if (enc->mode == GST_SPEEX_ENC_MODE_AUTO) {
+ GST_LOG_OBJECT (enc, "selected WB mode for samplerate %d", enc->rate);
+ enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_WB);
+ } else {
+ if (enc->speex_mode != speex_lib_get_mode (SPEEX_MODEID_WB)) {
+ gst_speex_enc_set_last_msg (enc,
+ "Warning: suggest to use wide band mode for this rate");
+ }
+ }
+ } else {
+ if (enc->mode == GST_SPEEX_ENC_MODE_AUTO) {
+ GST_LOG_OBJECT (enc, "selected NB mode for samplerate %d", enc->rate);
+ enc->speex_mode = speex_lib_get_mode (SPEEX_MODEID_NB);
+ } else {
+ if (enc->speex_mode != speex_lib_get_mode (SPEEX_MODEID_NB)) {
+ gst_speex_enc_set_last_msg (enc,
+ "Warning: suggest to use narrow band mode for this rate");
+ }
+ }
+ }
- structure = gst_caps_get_structure (caps, 0);
- gst_structure_get_int (structure, "rate", &speexenc->rate);
- if (gst_pad_try_set_caps (speexenc->srcpad,
- gst_caps_new_simple ("audio/x-speex",
- "rate", G_TYPE_INT, speexenc->rate,
- "channels", G_TYPE_INT, 1, NULL))) {
- speex_init_header (&speexenc->header, speexenc->rate, 1, speexenc->mode);
- speexenc->header.frames_per_packet = speexenc->n_packets;
-
- speexenc->state = speex_encoder_init (speexenc->mode);
- speex_encoder_ctl (speexenc->state, SPEEX_GET_FRAME_SIZE,
- &speexenc->frame_size);
-
- return GST_PAD_LINK_OK;
+ if (enc->rate != 8000 && enc->rate != 16000 && enc->rate != 32000) {
+ gst_speex_enc_set_last_msg (enc,
+ "Warning: speex is optimized for 8, 16 and 32 KHz");
}
- return GST_PAD_LINK_REFUSED;
+ speex_init_header (&enc->header, enc->rate, 1, enc->speex_mode);
+ enc->header.frames_per_packet = enc->nframes;
+ enc->header.vbr = enc->vbr;
+ enc->header.nb_channels = enc->channels;
+
+ /*Initialize Speex encoder */
+ enc->state = speex_encoder_init (enc->speex_mode);
+
+ speex_encoder_ctl (enc->state, SPEEX_GET_FRAME_SIZE, &enc->frame_size);
+ speex_encoder_ctl (enc->state, SPEEX_SET_COMPLEXITY, &enc->complexity);
+ speex_encoder_ctl (enc->state, SPEEX_SET_SAMPLING_RATE, &enc->rate);
+
+ if (enc->vbr)
+ speex_encoder_ctl (enc->state, SPEEX_SET_VBR_QUALITY, &enc->quality);
+ else {
+ gint tmp = floor (enc->quality);
+
+ speex_encoder_ctl (enc->state, SPEEX_SET_QUALITY, &tmp);
+ }
+ if (enc->bitrate) {
+ if (enc->quality >= 0.0 && enc->vbr) {
+ gst_speex_enc_set_last_msg (enc,
+ "Warning: bitrate option is overriding quality");
+ }
+ speex_encoder_ctl (enc->state, SPEEX_SET_BITRATE, &enc->bitrate);
+ }
+ if (enc->vbr) {
+ gint tmp = 1;
+
+ speex_encoder_ctl (enc->state, SPEEX_SET_VBR, &tmp);
+ } else if (enc->vad) {
+ gint tmp = 1;
+
+ speex_encoder_ctl (enc->state, SPEEX_SET_VAD, &tmp);
+ }
+
+ if (enc->dtx) {
+ gint tmp = 1;
+
+ speex_encoder_ctl (enc->state, SPEEX_SET_DTX, &tmp);
+ }
+
+ if (enc->dtx && !(enc->vbr || enc->abr || enc->vad)) {
+ gst_speex_enc_set_last_msg (enc,
+ "Warning: dtx is useless without vad, vbr or abr");
+ } else if ((enc->vbr || enc->abr) && (enc->vad)) {
+ gst_speex_enc_set_last_msg (enc,
+ "Warning: vad is already implied by vbr or abr");
+ }
+
+ if (enc->abr) {
+ speex_encoder_ctl (enc->state, SPEEX_SET_ABR, &enc->abr);
+ }
+
+ speex_encoder_ctl (enc->state, SPEEX_GET_LOOKAHEAD, &enc->lookahead);
+
+ GST_LOG_OBJECT (enc, "we have frame size %d, lookahead %d", enc->frame_size,
+ enc->lookahead);
+
+ return TRUE;
}
-static void
-gst_speexenc_chain (GstPad * pad, GstData * _data)
+/* push out the buffer */
+static GstFlowReturn
+gst_speex_enc_push_buffer (GstSpeexEnc * enc, GstBuffer * buffer)
{
- GstBuffer *buf = GST_BUFFER (_data);
- GstSpeexEnc *speexenc;
+ GST_DEBUG_OBJECT (enc, "pushing output buffer of size %u",
+ gst_buffer_get_size (buffer));
+
+ return gst_pad_push (GST_AUDIO_ENCODER_SRC_PAD (enc), buffer);
+}
+
+static gboolean
+gst_speex_enc_sink_event (GstAudioEncoder * benc, GstEvent * event)
+{
+ GstSpeexEnc *enc;
+
+ enc = GST_SPEEX_ENC (benc);
+
+ switch (GST_EVENT_TYPE (event)) {
+ case GST_EVENT_TAG:
+ {
+ if (enc->tags) {
+ GstTagList *list;
+
+ gst_event_parse_tag (event, &list);
+ gst_tag_list_insert (enc->tags, list,
+ gst_tag_setter_get_tag_merge_mode (GST_TAG_SETTER (enc)));
+ } else {
+ g_assert_not_reached ();
+ }
+ break;
+ }
+ default:
+ break;
+ }
+
+ /* we only peeked, let base class handle it */
+ return FALSE;
+}
+
+static GstFlowReturn
+gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf)
+{
+ gint frame_size = enc->frame_size;
+ gint bytes = frame_size * 2 * enc->channels, samples;
+ gint outsize, written, dtx_ret = 0;
+ guint8 *data, *data0 = NULL, *bdata, *outdata;
+ gsize bsize, size;
GstBuffer *outbuf;
- gint16 *data;
- guint8 *header_data;
- gint size;
- float input[1000];
- gint frame_size;
- gint i;
-
- g_return_if_fail (pad != NULL);
- g_return_if_fail (GST_IS_PAD (pad));
- g_return_if_fail (buf != NULL);
-
- speexenc = GST_SPEEXENC (GST_OBJECT_PARENT (pad));
-
- if (!GST_PAD_CAPS (speexenc->srcpad)) {
-
- if (!gst_pad_try_set_caps (speexenc->srcpad,
- gst_caps_new_simple ("audio/x-speex",
- "rate", G_TYPE_INT, speexenc->rate,
- "channels", G_TYPE_INT, 1, NULL))) {
- GST_ELEMENT_ERROR (speexenc, CORE, NEGOTIATION, (NULL), (NULL));
- return;
+ GstFlowReturn ret = GST_FLOW_OK;
+
+ if (G_LIKELY (buf)) {
+ bdata = gst_buffer_map (buf, &bsize, NULL, GST_MAP_READ);
+
+ if (G_UNLIKELY (bsize % bytes)) {
+ GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
+
+ size = ((bsize / bytes) + 1) * bytes;
+ data0 = data = g_malloc0 (size);
+ memcpy (data, bdata, bsize);
+ gst_buffer_unmap (buf, bdata, bsize);
+ bdata = NULL;
+ } else {
+ data = bdata;
+ size = bsize;
}
+ } else {
+ GST_DEBUG_OBJECT (enc, "nothing to drain");
+ goto done;
}
- if (speexenc->packet_count == 0) {
- header_data = speex_header_to_packet (&speexenc->header, &size);
+ samples = size / (2 * enc->channels);
+ speex_bits_reset (&enc->bits);
+
+ /* FIXME what about dropped samples if DTS enabled ?? */
- outbuf = gst_buffer_new ();
- GST_BUFFER_DATA (outbuf) = header_data;
- GST_BUFFER_SIZE (outbuf) = size;
+ while (size) {
+ GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", frame_size, bytes);
- gst_pad_push (speexenc->srcpad, GST_DATA (outbuf));
+ if (enc->channels == 2) {
+ speex_encode_stereo_int ((gint16 *) data, frame_size, &enc->bits);
+ }
+ dtx_ret += speex_encode_int (enc->state, (gint16 *) data, &enc->bits);
+
+ data += bytes;
+ size -= bytes;
}
- data = (gint16 *) GST_BUFFER_DATA (buf);
- size = GST_BUFFER_SIZE (buf) / sizeof (gint16);
+ speex_bits_insert_terminator (&enc->bits);
+ outsize = speex_bits_nbytes (&enc->bits);
- frame_size = speexenc->frame_size;
+ if (bdata)
+ gst_buffer_unmap (buf, bdata, bsize);
- if (speexenc->bufsize && (speexenc->bufsize + size >= frame_size)) {
- memcpy (speexenc->buffer + speexenc->bufsize, data,
- (frame_size - speexenc->bufsize) * sizeof (gint16));
+#if 0
+ ret = gst_pad_alloc_buffer_and_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc),
+ GST_BUFFER_OFFSET_NONE, outsize,
+ GST_PAD_CAPS (GST_AUDIO_ENCODER_SRC_PAD (enc)), &outbuf);
- for (i = 0; i < frame_size; i++)
- input[i] = speexenc->buffer[i];
+ if ((GST_FLOW_OK != ret))
+ goto done;
+#endif
+ outbuf = gst_buffer_new_allocate (NULL, outsize, 0);
+ outdata = gst_buffer_map (outbuf, NULL, NULL, GST_MAP_WRITE);
- speex_encode (speexenc->state, input, &speexenc->bits);
- speexenc->packet_count++;
+ written = speex_bits_write (&enc->bits, (gchar *) outdata, outsize);
- if (speexenc->packet_count % speexenc->n_packets == 0) {
- GstBuffer *outbuf;
+ if (G_UNLIKELY (written < outsize)) {
+ GST_ERROR_OBJECT (enc, "short write: %d < %d bytes", written, outsize);
+ } else if (G_UNLIKELY (written > outsize)) {
+ GST_ERROR_OBJECT (enc, "overrun: %d > %d bytes", written, outsize);
+ written = outsize;
+ }
+ gst_buffer_unmap (outbuf, outdata, written);
- outbuf = gst_buffer_new_and_alloc (frame_size * speexenc->n_packets);
- GST_BUFFER_SIZE (outbuf) = speex_bits_write (&speexenc->bits,
- GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf));
- GST_BUFFER_TIMESTAMP (outbuf) = speexenc->next_ts;
- speex_bits_reset (&speexenc->bits);
+ if (!dtx_ret)
+ GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
- gst_pad_push (speexenc->srcpad, GST_DATA (outbuf));
- speexenc->next_ts += frame_size * GST_SECOND / speexenc->rate;
- }
+ ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
+ outbuf, samples);
+
+done:
+ g_free (data0);
+ return ret;
+}
+
+/*
+ * (really really) FIXME: move into core (dixit tpm)
+ */
+/**
+ * _gst_caps_set_buffer_array:
+ * @caps: a #GstCaps
+ * @field: field in caps to set
+ * @buf: header buffers
+ *
+ * Adds given buffers to an array of buffers set as the given @field
+ * on the given @caps. List of buffer arguments must be NULL-terminated.
+ *
+ * Returns: input caps with a streamheader field added, or NULL if some error
+ */
+static GstCaps *
+_gst_caps_set_buffer_array (GstCaps * caps, const gchar * field,
+ GstBuffer * buf, ...)
+{
+ GstStructure *structure = NULL;
+ va_list va;
+ GValue array = { 0 };
+ GValue value = { 0 };
+
+ g_return_val_if_fail (caps != NULL, NULL);
+ g_return_val_if_fail (gst_caps_is_fixed (caps), NULL);
+ g_return_val_if_fail (field != NULL, NULL);
+
+ caps = gst_caps_make_writable (caps);
+ structure = gst_caps_get_structure (caps, 0);
- size -= (speexenc->frame_size - speexenc->bufsize);
- data += (speexenc->frame_size - speexenc->bufsize);
+ g_value_init (&array, GST_TYPE_ARRAY);
- speexenc->bufsize = 0;
+ va_start (va, buf);
+ /* put buffers in a fixed list */
+ while (buf) {
+ g_assert (gst_buffer_is_writable (buf));
+
+ /* mark buffer */
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+
+ g_value_init (&value, GST_TYPE_BUFFER);
+ buf = gst_buffer_copy (buf);
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+ gst_value_set_buffer (&value, buf);
+ gst_buffer_unref (buf);
+ gst_value_array_append_value (&array, &value);
+ g_value_unset (&value);
+
+ buf = va_arg (va, GstBuffer *);
}
- while (size >= frame_size) {
- for (i = 0; i < frame_size; i++)
- input[i] = data[i];
+ gst_structure_set_value (structure, field, &array);
+ g_value_unset (&array);
- speex_encode (speexenc->state, input, &speexenc->bits);
- speexenc->packet_count++;
+ return caps;
+}
- if (speexenc->packet_count % speexenc->n_packets == 0) {
- GstBuffer *outbuf;
+static GstFlowReturn
+gst_speex_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
+{
+ GstSpeexEnc *enc;
+ GstFlowReturn ret = GST_FLOW_OK;
+
+ enc = GST_SPEEX_ENC (benc);
+
+ if (!enc->header_sent) {
+ /* Speex streams begin with two headers; the initial header (with
+ most of the codec setup parameters) which is mandated by the Ogg
+ bitstream spec. The second header holds any comment fields.
+ We merely need to make the headers, then pass them to libspeex
+ one at a time; libspeex handles the additional Ogg bitstream
+ constraints */
+ GstBuffer *buf1, *buf2;
+ GstCaps *caps;
+ guchar *data;
+ gint data_len;
+
+ /* create header buffer */
+ data = (guint8 *) speex_header_to_packet (&enc->header, &data_len);
+ buf1 = gst_buffer_new_wrapped (data, data_len);
+ GST_BUFFER_OFFSET_END (buf1) = 0;
+ GST_BUFFER_OFFSET (buf1) = 0;
+
+ /* create comment buffer */
+ buf2 = gst_speex_enc_create_metadata_buffer (enc);
+
+ /* mark and put on caps */
+ caps = gst_caps_new_simple ("audio/x-speex", "rate", G_TYPE_INT, enc->rate,
+ "channels", G_TYPE_INT, enc->channels, NULL);
+ caps = _gst_caps_set_buffer_array (caps, "streamheader", buf1, buf2, NULL);
+
+ /* negotiate with these caps */
+ GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
+
+ gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps);
+ gst_caps_unref (caps);
+
+ /* push out buffers */
+ /* store buffers for later pre_push sending */
+ g_slist_foreach (enc->headers, (GFunc) gst_buffer_unref, NULL);
+ enc->headers = NULL;
+ GST_DEBUG_OBJECT (enc, "storing header buffers");
+ enc->headers = g_slist_prepend (enc->headers, buf2);
+ enc->headers = g_slist_prepend (enc->headers, buf1);
+
+ enc->header_sent = TRUE;
+ }
+
+ GST_DEBUG_OBJECT (enc, "received buffer %p of %u bytes", buf,
+ buf ? gst_buffer_get_size (buf) : 0);
- outbuf = gst_buffer_new_and_alloc (frame_size * speexenc->n_packets);
- GST_BUFFER_SIZE (outbuf) = speex_bits_write (&speexenc->bits,
- GST_BUFFER_DATA (outbuf), GST_BUFFER_SIZE (outbuf));
- GST_BUFFER_TIMESTAMP (outbuf) = speexenc->next_ts;
- speex_bits_reset (&speexenc->bits);
+ ret = gst_speex_enc_encode (enc, buf);
+
+ return ret;
+}
- gst_pad_push (speexenc->srcpad, GST_DATA (outbuf));
- speexenc->next_ts += frame_size * GST_SECOND / speexenc->rate;
+static GstFlowReturn
+gst_speex_enc_pre_push (GstAudioEncoder * benc, GstBuffer ** buffer)
+{
+ GstSpeexEnc *enc;
+ GstFlowReturn ret = GST_FLOW_OK;
+
+ enc = GST_SPEEX_ENC (benc);
+
+ /* FIXME 0.11 ? get rid of this special ogg stuff and have it
+ * put and use 'codec data' in caps like anything else,
+ * with all the usual out-of-band advantage etc */
+ if (G_UNLIKELY (enc->headers)) {
+ GSList *header = enc->headers;
+
+ /* try to push all of these, if we lose one, might as well lose all */
+ while (header) {
+ if (ret == GST_FLOW_OK)
+ ret = gst_speex_enc_push_buffer (enc, header->data);
+ else
+ gst_speex_enc_push_buffer (enc, header->data);
+ header = g_slist_next (header);
}
- size -= frame_size;
- data += frame_size;
+ g_slist_free (enc->headers);
+ enc->headers = NULL;
}
- if (size) {
- memcpy (speexenc->buffer + speexenc->bufsize, data, size * sizeof (gint16));
- speexenc->bufsize += size;
+ return ret;
+}
+
+static void
+gst_speex_enc_get_property (GObject * object, guint prop_id, GValue * value,
+ GParamSpec * pspec)
+{
+ GstSpeexEnc *enc;
+
+ enc = GST_SPEEX_ENC (object);
+
+ switch (prop_id) {
+ case PROP_QUALITY:
+ g_value_set_float (value, enc->quality);
+ break;
+ case PROP_BITRATE:
+ g_value_set_int (value, enc->bitrate);
+ break;
+ case PROP_MODE:
+ g_value_set_enum (value, enc->mode);
+ break;
+ case PROP_VBR:
+ g_value_set_boolean (value, enc->vbr);
+ break;
+ case PROP_ABR:
+ g_value_set_int (value, enc->abr);
+ break;
+ case PROP_VAD:
+ g_value_set_boolean (value, enc->vad);
+ break;
+ case PROP_DTX:
+ g_value_set_boolean (value, enc->dtx);
+ break;
+ case PROP_COMPLEXITY:
+ g_value_set_int (value, enc->complexity);
+ break;
+ case PROP_NFRAMES:
+ g_value_set_int (value, enc->nframes);
+ break;
+ case PROP_LAST_MESSAGE:
+ g_value_set_string (value, enc->last_message);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
}
+}
- gst_buffer_unref (buf);
+static void
+gst_speex_enc_set_property (GObject * object, guint prop_id,
+ const GValue * value, GParamSpec * pspec)
+{
+ GstSpeexEnc *enc;
+
+ enc = GST_SPEEX_ENC (object);
+
+ switch (prop_id) {
+ case PROP_QUALITY:
+ enc->quality = g_value_get_float (value);
+ break;
+ case PROP_BITRATE:
+ enc->bitrate = g_value_get_int (value);
+ break;
+ case PROP_MODE:
+ enc->mode = g_value_get_enum (value);
+ break;
+ case PROP_VBR:
+ enc->vbr = g_value_get_boolean (value);
+ break;
+ case PROP_ABR:
+ enc->abr = g_value_get_int (value);
+ break;
+ case PROP_VAD:
+ enc->vad = g_value_get_boolean (value);
+ break;
+ case PROP_DTX:
+ enc->dtx = g_value_get_boolean (value);
+ break;
+ case PROP_COMPLEXITY:
+ enc->complexity = g_value_get_int (value);
+ break;
+ case PROP_NFRAMES:
+ enc->nframes = g_value_get_int (value);
+ break;
+ default:
+ G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+ break;
+ }
}
projects / platform / upstream / gstreamer.git / blobdiff