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gstreamer/subprojects/gst-plugins-good/ext/speex/gstspeexenc.c

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/* GStreamer Speex Encoder
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
*
* 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., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-speexenc
* @title: speexenc
* @see_also: speexdec, oggmux
*
* This element encodes audio as a Speex stream.
* [Speex](http://www.speex.org/) is a royalty-free
* audio codec maintained by the [Xiph.org Foundation](http://www.xiph.org/).
*
* ## Example pipelines
* |[
* gst-launch-1.0 audiotestsrc num-buffers=100 ! speexenc ! oggmux ! filesink location=beep.ogg
* ]| Encode an Ogg/Speex file.
*
*/
#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 "gstspeexelements.h"
#include "gstspeexenc.h"
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 ", "
"layout = (string) interleaved, "
"rate = (int) [ 6000, 48000 ], "
"channels = (int) 1; "
"audio/x-raw, "
"format = (string) " FORMAT_STR ", "
"layout = (string) interleaved, "
"rate = (int) [ 6000, 48000 ], "
"channels = (int) 2, " "channel-mask = (bitmask) 0x3")
);
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
enum
{
PROP_0,
PROP_QUALITY,
PROP_BITRATE,
PROP_MODE,
PROP_VBR,
PROP_ABR,
PROP_VAD,
PROP_DTX,
PROP_COMPLEXITY,
PROP_NFRAMES,
PROP_LAST_MESSAGE
};
#define GST_TYPE_SPEEX_ENC_MODE (gst_speex_enc_mode_get_type())
static GType
gst_speex_enc_mode_get_type (void)
{
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_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 GstFlowReturn gst_speex_enc_encode (GstSpeexEnc * enc, GstBuffer * buf);
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);
#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));
GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (speexenc, "speexenc",
GST_RANK_PRIMARY, GST_TYPE_SPEEX_ENC, speex_element_init (plugin));
static void
gst_speex_enc_class_init (GstSpeexEncClass * klass)
{
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->sink_event = GST_DEBUG_FUNCPTR (gst_speex_enc_sink_event);
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_static_pad_template (gstelement_class, &src_factory);
gst_element_class_add_static_pad_template (gstelement_class, &sink_factory);
gst_element_class_set_static_metadata (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");
gst_type_mark_as_plugin_api (GST_TYPE_SPEEX_ENC_MODE, 0);
}
static void
gst_speex_enc_finalize (GObject * object)
{
GstSpeexEnc *enc;
enc = GST_SPEEX_ENC (object);
g_free (enc->last_message);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
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);
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (enc));
}
static gboolean
gst_speex_enc_start (GstAudioEncoder * benc)
{
GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
GST_DEBUG_OBJECT (enc, "start");
speex_bits_init (&enc->bits);
enc->tags = gst_tag_list_new_empty ();
enc->header_sent = FALSE;
enc->encoded_samples = 0;
return TRUE;
}
static gboolean
gst_speex_enc_stop (GstAudioEncoder * benc)
{
GstSpeexEnc *enc = GST_SPEEX_ENC (benc);
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);
speex_bits_set_bit_buffer (&enc->bits, NULL, 0);
gst_tag_list_unref (enc->tags);
enc->tags = NULL;
gst_tag_setter_reset_tags (GST_TAG_SETTER (enc));
return TRUE;
}
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);
}
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_unref (merged_tags);
GST_BUFFER_OFFSET (comments) = 0;
GST_BUFFER_OFFSET_END (comments) = 0;
return comments;
}
static void
gst_speex_enc_set_last_msg (GstSpeexEnc * enc, const gchar * msg)
{
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 gboolean
gst_speex_enc_setup (GstSpeexEnc * enc)
{
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;
}
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");
}
}
}
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");
}
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 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;
}
case GST_EVENT_SEGMENT:
enc->encoded_samples = 0;
break;
default:
break;
}
/* we only peeked, let base class handle it */
return GST_AUDIO_ENCODER_CLASS (parent_class)->sink_event (benc, event);
}
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;
GstMapInfo map;
guint8 *data, *data0 = NULL, *bdata;
gsize bsize, size;
GstBuffer *outbuf;
GstFlowReturn ret = GST_FLOW_OK;
GstSegment *segment;
GstClockTime duration;
if (G_LIKELY (buf)) {
gst_buffer_map (buf, &map, GST_MAP_READ);
bdata = map.data;
bsize = map.size;
if (G_UNLIKELY (bsize % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
/* If encoding part of a frame, and we have no set stop time on
* the output segment, we update the segment stop time to reflect
* the last sample. This will let oggmux set the last page's
* granpos to tell a decoder the dummy samples should be clipped.
*/
segment = &GST_AUDIO_ENCODER_OUTPUT_SEGMENT (enc);
GST_DEBUG_OBJECT (enc, "existing output segment %" GST_SEGMENT_FORMAT,
segment);
if (!GST_CLOCK_TIME_IS_VALID (segment->stop)) {
int input_samples = bsize / (enc->channels * 2);
GST_DEBUG_OBJECT (enc,
"No stop time and partial frame, updating segment");
duration =
gst_util_uint64_scale (enc->encoded_samples + input_samples,
GST_SECOND, enc->rate);
segment->stop = segment->start + duration;
GST_DEBUG_OBJECT (enc, "new output segment %" GST_SEGMENT_FORMAT,
segment);
gst_pad_push_event (GST_AUDIO_ENCODER_SRC_PAD (enc),
gst_event_new_segment (segment));
}
size = ((bsize / bytes) + 1) * bytes;
data0 = data = g_malloc0 (size);
memcpy (data, bdata, bsize);
gst_buffer_unmap (buf, &map);
bdata = NULL;
} else {
data = bdata;
size = bsize;
}
} else {
GST_DEBUG_OBJECT (enc, "nothing to drain");
goto done;
}
samples = size / (2 * enc->channels);
speex_bits_reset (&enc->bits);
/* FIXME what about dropped samples if DTS enabled ?? */
while (size) {
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)", frame_size, bytes);
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;
}
speex_bits_insert_terminator (&enc->bits);
outsize = speex_bits_nbytes (&enc->bits);
if (bdata)
gst_buffer_unmap (buf, &map);
#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);
if ((GST_FLOW_OK != ret))
goto done;
#endif
outbuf = gst_buffer_new_allocate (NULL, outsize, NULL);
gst_buffer_map (outbuf, &map, GST_MAP_WRITE);
written = speex_bits_write (&enc->bits, (gchar *) map.data, outsize);
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, &map);
gst_buffer_resize (outbuf, 0, written);
if (!dtx_ret)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
ret = gst_audio_encoder_finish_frame (GST_AUDIO_ENCODER (enc),
outbuf, samples);
enc->encoded_samples += frame_size;
done:
g_free (data0);
return ret;
}
/*
* (really really) FIXME: move into core (dixit tpm)
*/
/*
* _gst_caps_set_buffer_array:
* @caps: (transfer full): 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: (transfer full): input caps with a streamheader field added, or NULL
* if some error occurred
*/
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);
g_value_init (&array, GST_TYPE_ARRAY);
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_HEADER);
g_value_init (&value, GST_TYPE_BUFFER);
buf = gst_buffer_copy (buf);
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
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 *);
}
va_end (va);
gst_structure_set_value (structure, field, &array);
g_value_unset (&array);
return caps;
}
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;
GList *headers;
/* create header buffer */
data = (guint8 *) speex_header_to_packet (&enc->header, &data_len);
buf1 = gst_buffer_new_wrapped_full (0,
data, data_len, 0, data_len, data, (GDestroyNotify) speex_header_free);
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_audio_encoder_set_output_format (GST_AUDIO_ENCODER (enc), caps);
gst_caps_unref (caps);
/* push out buffers */
/* store buffers for later pre_push sending */
headers = NULL;
GST_DEBUG_OBJECT (enc, "storing header buffers");
headers = g_list_prepend (headers, buf2);
headers = g_list_prepend (headers, buf1);
gst_audio_encoder_set_headers (benc, headers);
enc->header_sent = TRUE;
}
GST_DEBUG_OBJECT (enc, "received buffer %p of %" G_GSIZE_FORMAT " bytes", buf,
buf ? gst_buffer_get_size (buf) : 0);
ret = gst_speex_enc_encode (enc, buf);
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;
}
}
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;
}
}
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