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gstreamer/ext/faac/gstfaac.c
Mark Nauwelaerts bc6549a108 faac: handle trailing encoder output
2010-10-11 17:13:10 +02:00

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/* GStreamer FAAC (Free AAC Encoder) plugin
* Copyright (C) 2003 Ronald Bultje <rbultje@ronald.bitfreak.net>
* Copyright (C) 2009 Mark Nauwelaerts <mnauw@users.sourceforge.net>
*
* 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-faac
* @see_also: faad
*
* faac encodes raw audio to AAC (MPEG-4 part 3) streams.
*
* The #GstFaac:outputformat property determines whether or not the
* AAC data needs additional framing provided by a container
* (such as Matroska or Quicktime).
* This is required for raw data, whereas ADTS formatted AAC already provides
* framing and needs no container.
*
* The #GstFaac:profile property determines the AAC profile, where the default
* LC (Low Complexity) profile is most widely used, supported and suitable for
* general use. The other profiles are very rarely used and often not supported.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch audiotestsrc wave=sine num-buffers=100 ! audioconvert ! faac ! matroskamux ! filesink location=sine.mkv
* ]| Encode a sine beep as aac and write to matroska container.
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <stdlib.h>
#include <string.h>
#include <gst/audio/multichannel.h>
#include "gstfaac.h"
#define SINK_CAPS \
"audio/x-raw-int, " \
"endianness = (int) BYTE_ORDER, " \
"signed = (boolean) true, " \
"width = (int) 16, " \
"depth = (int) 16, " \
"rate = (int) [ 8000, 96000 ], " \
"channels = (int) [ 1, 6 ] "
/* these don't seem to work? */
#if 0
"audio/x-raw-int, "
"endianness = (int) BYTE_ORDER, "
"signed = (boolean) true, "
"width = (int) 32, "
"depth = (int) { 24, 32 }, "
"rate = (int) [ 8000, 96000], "
"channels = (int) [ 1, 6]; "
"audio/x-raw-float, "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32, "
"rate = (int) [ 8000, 96000], " "channels = (int) [ 1, 6]"
#endif
#define SRC_CAPS \
"audio/mpeg, " \
"mpegversion = (int) { 4, 2 }, " \
"channels = (int) [ 1, 6 ], " \
"rate = (int) [ 8000, 96000 ], " \
"stream-format = (string) { adts, raw } "
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SRC_CAPS));
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (SINK_CAPS));
enum
{
ARG_0,
ARG_OUTPUTFORMAT,
ARG_BITRATE,
ARG_PROFILE,
ARG_TNS,
ARG_MIDSIDE,
ARG_SHORTCTL
};
static void gst_faac_base_init (GstFaacClass * klass);
static void gst_faac_class_init (GstFaacClass * klass);
static void gst_faac_init (GstFaac * faac);
static void gst_faac_finalize (GObject * object);
static void gst_faac_reset (GstFaac * faac);
static void gst_faac_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_faac_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static gboolean gst_faac_sink_event (GstPad * pad, GstEvent * event);
static gboolean gst_faac_configure_source_pad (GstFaac * faac);
static gboolean gst_faac_sink_setcaps (GstPad * pad, GstCaps * caps);
static GstCaps *gst_faac_sink_getcaps (GstPad * pad);
static GstFlowReturn gst_faac_push_buffers (GstFaac * faac, gboolean force);
static GstFlowReturn gst_faac_chain (GstPad * pad, GstBuffer * data);
static GstStateChangeReturn gst_faac_change_state (GstElement * element,
GstStateChange transition);
static GstElementClass *parent_class = NULL;
GST_DEBUG_CATEGORY_STATIC (faac_debug);
#define GST_CAT_DEFAULT faac_debug
#define FAAC_DEFAULT_MPEGVERSION 4
#define FAAC_DEFAULT_OUTPUTFORMAT 0 /* RAW */
#define FAAC_DEFAULT_BITRATE 128 * 1000
#define FAAC_DEFAULT_PROFILE LOW
#define FAAC_DEFAULT_TNS FALSE
#define FAAC_DEFAULT_MIDSIDE TRUE
#define FAAC_DEFAULT_SHORTCTL SHORTCTL_NORMAL
GType
gst_faac_get_type (void)
{
static GType gst_faac_type = 0;
if (!gst_faac_type) {
static const GTypeInfo gst_faac_info = {
sizeof (GstFaacClass),
(GBaseInitFunc) gst_faac_base_init,
NULL,
(GClassInitFunc) gst_faac_class_init,
NULL,
NULL,
sizeof (GstFaac),
0,
(GInstanceInitFunc) gst_faac_init,
};
const GInterfaceInfo preset_interface_info = {
NULL, /* interface_init */
NULL, /* interface_finalize */
NULL /* interface_data */
};
gst_faac_type = g_type_register_static (GST_TYPE_ELEMENT,
"GstFaac", &gst_faac_info, 0);
g_type_add_interface_static (gst_faac_type, GST_TYPE_PRESET,
&preset_interface_info);
}
return gst_faac_type;
}
static void
gst_faac_base_init (GstFaacClass * klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details_simple (element_class, "AAC audio encoder",
"Codec/Encoder/Audio",
"Free MPEG-2/4 AAC encoder",
"Ronald Bultje <rbultje@ronald.bitfreak.net>");
GST_DEBUG_CATEGORY_INIT (faac_debug, "faac", 0, "AAC encoding");
}
#define GST_TYPE_FAAC_PROFILE (gst_faac_profile_get_type ())
static GType
gst_faac_profile_get_type (void)
{
static GType gst_faac_profile_type = 0;
if (!gst_faac_profile_type) {
static GEnumValue gst_faac_profile[] = {
{MAIN, "MAIN", "Main profile"},
{LOW, "LC", "Low complexity profile"},
{SSR, "SSR", "Scalable sampling rate profile"},
{LTP, "LTP", "Long term prediction profile"},
{0, NULL, NULL},
};
gst_faac_profile_type = g_enum_register_static ("GstFaacProfile",
gst_faac_profile);
}
return gst_faac_profile_type;
}
#define GST_TYPE_FAAC_SHORTCTL (gst_faac_shortctl_get_type ())
static GType
gst_faac_shortctl_get_type (void)
{
static GType gst_faac_shortctl_type = 0;
if (!gst_faac_shortctl_type) {
static GEnumValue gst_faac_shortctl[] = {
{SHORTCTL_NORMAL, "SHORTCTL_NORMAL", "Normal block type"},
{SHORTCTL_NOSHORT, "SHORTCTL_NOSHORT", "No short blocks"},
{SHORTCTL_NOLONG, "SHORTCTL_NOLONG", "No long blocks"},
{0, NULL, NULL},
};
gst_faac_shortctl_type = g_enum_register_static ("GstFaacShortCtl",
gst_faac_shortctl);
}
return gst_faac_shortctl_type;
}
#define GST_TYPE_FAAC_OUTPUTFORMAT (gst_faac_outputformat_get_type ())
static GType
gst_faac_outputformat_get_type (void)
{
static GType gst_faac_outputformat_type = 0;
if (!gst_faac_outputformat_type) {
static GEnumValue gst_faac_outputformat[] = {
{0, "OUTPUTFORMAT_RAW", "Raw AAC"},
{1, "OUTPUTFORMAT_ADTS", "ADTS headers"},
{0, NULL, NULL},
};
gst_faac_outputformat_type = g_enum_register_static ("GstFaacOutputFormat",
gst_faac_outputformat);
}
return gst_faac_outputformat_type;
}
static void
gst_faac_class_init (GstFaacClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
parent_class = g_type_class_peek_parent (klass);
gobject_class->set_property = gst_faac_set_property;
gobject_class->get_property = gst_faac_get_property;
gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_faac_finalize);
/* properties */
g_object_class_install_property (gobject_class, ARG_BITRATE,
g_param_spec_int ("bitrate", "Bitrate (bps)", "Bitrate in bits/sec",
8 * 1000, 320 * 1000, FAAC_DEFAULT_BITRATE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_PROFILE,
g_param_spec_enum ("profile", "Profile", "MPEG/AAC encoding profile",
GST_TYPE_FAAC_PROFILE, FAAC_DEFAULT_PROFILE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_TNS,
g_param_spec_boolean ("tns", "TNS", "Use temporal noise shaping",
FAAC_DEFAULT_TNS, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_MIDSIDE,
g_param_spec_boolean ("midside", "Midside", "Allow mid/side encoding",
FAAC_DEFAULT_MIDSIDE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_SHORTCTL,
g_param_spec_enum ("shortctl", "Block type",
"Block type encorcing",
GST_TYPE_FAAC_SHORTCTL, FAAC_DEFAULT_SHORTCTL, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, ARG_OUTPUTFORMAT,
g_param_spec_enum ("outputformat", "Output format",
"Format of output frames",
GST_TYPE_FAAC_OUTPUTFORMAT, FAAC_DEFAULT_OUTPUTFORMAT,
G_PARAM_READWRITE));
/* virtual functions */
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_faac_change_state);
}
static void
gst_faac_init (GstFaac * faac)
{
faac->sinkpad = gst_pad_new_from_static_template (&sink_template, "sink");
gst_pad_set_chain_function (faac->sinkpad,
GST_DEBUG_FUNCPTR (gst_faac_chain));
gst_pad_set_setcaps_function (faac->sinkpad,
GST_DEBUG_FUNCPTR (gst_faac_sink_setcaps));
gst_pad_set_getcaps_function (faac->sinkpad,
GST_DEBUG_FUNCPTR (gst_faac_sink_getcaps));
gst_pad_set_event_function (faac->sinkpad,
GST_DEBUG_FUNCPTR (gst_faac_sink_event));
gst_element_add_pad (GST_ELEMENT (faac), faac->sinkpad);
faac->srcpad = gst_pad_new_from_static_template (&src_template, "src");
gst_pad_use_fixed_caps (faac->srcpad);
gst_element_add_pad (GST_ELEMENT (faac), faac->srcpad);
faac->adapter = gst_adapter_new ();
/* default properties */
faac->bitrate = FAAC_DEFAULT_BITRATE;
faac->profile = FAAC_DEFAULT_PROFILE;
faac->shortctl = FAAC_DEFAULT_SHORTCTL;
faac->outputformat = FAAC_DEFAULT_OUTPUTFORMAT;
faac->tns = FAAC_DEFAULT_TNS;
faac->midside = FAAC_DEFAULT_MIDSIDE;
gst_faac_reset (faac);
}
static void
gst_faac_reset (GstFaac * faac)
{
faac->handle = NULL;
faac->samplerate = -1;
faac->channels = -1;
faac->offset = 0;
gst_adapter_clear (faac->adapter);
}
static void
gst_faac_finalize (GObject * object)
{
GstFaac *faac = (GstFaac *) object;
g_object_unref (faac->adapter);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_faac_close_encoder (GstFaac * faac)
{
if (faac->handle)
faacEncClose (faac->handle);
faac->handle = NULL;
gst_adapter_clear (faac->adapter);
faac->offset = 0;
}
static const GstAudioChannelPosition aac_channel_positions[][8] = {
{GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
{GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT},
{
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE}
};
static GstCaps *
gst_faac_sink_getcaps (GstPad * pad)
{
static volatile gsize sinkcaps = 0;
if (g_once_init_enter (&sinkcaps)) {
GstCaps *tmp = gst_caps_new_empty ();
GstStructure *s, *t;
gint i, c;
s = gst_structure_new ("audio/x-raw-int",
"endianness", G_TYPE_INT, G_BYTE_ORDER,
"signed", G_TYPE_BOOLEAN, TRUE,
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16, "rate", GST_TYPE_INT_RANGE, 8000, 96000, NULL);
for (i = 1; i <= 6; i++) {
GValue chanpos = { 0 };
GValue pos = { 0 };
t = gst_structure_copy (s);
gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
g_value_init (&chanpos, GST_TYPE_ARRAY);
g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
for (c = 0; c < i; c++) {
g_value_set_enum (&pos, aac_channel_positions[i - 1][c]);
gst_value_array_append_value (&chanpos, &pos);
}
g_value_unset (&pos);
gst_structure_set_value (t, "channel-positions", &chanpos);
g_value_unset (&chanpos);
gst_caps_append_structure (tmp, t);
}
gst_structure_free (s);
GST_DEBUG_OBJECT (pad, "Generated sinkcaps: %" GST_PTR_FORMAT, tmp);
g_once_init_leave (&sinkcaps, (gsize) tmp);
}
return gst_caps_ref ((GstCaps *) sinkcaps);
}
static gboolean
gst_faac_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstFaac *faac = GST_FAAC (gst_pad_get_parent (pad));
GstStructure *structure = gst_caps_get_structure (caps, 0);
faacEncHandle *handle;
gint channels, samplerate, width;
gulong samples, bytes, fmt = 0, bps = 0;
gboolean result = FALSE;
if (!gst_caps_is_fixed (caps))
goto refuse_caps;
if (!gst_structure_get_int (structure, "channels", &channels) ||
!gst_structure_get_int (structure, "rate", &samplerate)) {
goto refuse_caps;
}
if (gst_structure_has_name (structure, "audio/x-raw-int")) {
gst_structure_get_int (structure, "width", &width);
switch (width) {
case 16:
fmt = FAAC_INPUT_16BIT;
bps = 2;
break;
case 24:
case 32:
fmt = FAAC_INPUT_32BIT;
bps = 4;
break;
default:
g_return_val_if_reached (FALSE);
}
} else if (gst_structure_has_name (structure, "audio/x-raw-float")) {
fmt = FAAC_INPUT_FLOAT;
bps = 4;
}
if (!fmt)
goto refuse_caps;
/* If the encoder is initialized, do not
reinitialize it again if not necessary */
if (faac->handle) {
if (samplerate == faac->samplerate && channels == faac->channels &&
fmt == faac->format)
return TRUE;
/* clear out pending frames */
gst_faac_push_buffers (faac, TRUE);
gst_faac_close_encoder (faac);
}
if (!(handle = faacEncOpen (samplerate, channels, &samples, &bytes)))
goto setup_failed;
/* ok, record and set up */
faac->format = fmt;
faac->bps = bps;
faac->handle = handle;
faac->bytes = bytes;
faac->samples = samples;
faac->channels = channels;
faac->samplerate = samplerate;
/* finish up */
result = gst_faac_configure_source_pad (faac);
done:
gst_object_unref (faac);
return result;
/* ERRORS */
setup_failed:
{
GST_ELEMENT_ERROR (faac, LIBRARY, SETTINGS, (NULL), (NULL));
goto done;
}
refuse_caps:
{
GST_WARNING_OBJECT (faac, "refused caps %" GST_PTR_FORMAT, caps);
goto done;
}
}
static gboolean
gst_faac_configure_source_pad (GstFaac * faac)
{
GstCaps *allowed_caps;
GstCaps *srccaps;
gboolean ret = FALSE;
gint n, ver, mpegversion = 2;
faacEncConfiguration *conf;
guint maxbitrate;
mpegversion = FAAC_DEFAULT_MPEGVERSION;
allowed_caps = gst_pad_get_allowed_caps (faac->srcpad);
GST_DEBUG_OBJECT (faac, "allowed caps: %" GST_PTR_FORMAT, allowed_caps);
if (allowed_caps) {
if (gst_caps_is_empty (allowed_caps))
goto empty_caps;
if (!gst_caps_is_any (allowed_caps)) {
for (n = 0; n < gst_caps_get_size (allowed_caps); n++) {
GstStructure *s = gst_caps_get_structure (allowed_caps, n);
if (gst_structure_get_int (s, "mpegversion", &ver) &&
(ver == 4 || ver == 2)) {
mpegversion = ver;
break;
}
}
}
gst_caps_unref (allowed_caps);
}
/* we negotiated caps update current configuration */
conf = faacEncGetCurrentConfiguration (faac->handle);
conf->mpegVersion = (mpegversion == 4) ? MPEG4 : MPEG2;
conf->aacObjectType = faac->profile;
conf->allowMidside = faac->midside;
conf->useLfe = 0;
conf->useTns = faac->tns;
conf->bitRate = faac->bitrate / faac->channels;
conf->inputFormat = faac->format;
conf->outputFormat = faac->outputformat;
conf->shortctl = faac->shortctl;
/* check, warn and correct if the max bitrate for the given samplerate is
* exceeded. Maximum of 6144 bit for a channel */
maxbitrate =
(unsigned int) (6144.0 * (double) faac->samplerate / (double) 1024.0 +
.5);
if (conf->bitRate > maxbitrate) {
GST_ELEMENT_WARNING (faac, RESOURCE, SETTINGS, (NULL),
("bitrate %lu exceeds maximum allowed bitrate of %u for samplerate %d. "
"Setting bitrate to %u", conf->bitRate, maxbitrate,
faac->samplerate, maxbitrate));
conf->bitRate = maxbitrate;
}
/* default 0 to start with, libfaac chooses based on bitrate */
conf->bandWidth = 0;
if (!faacEncSetConfiguration (faac->handle, conf))
goto set_failed;
/* let's see what really happened,
* note that this may not really match desired rate */
GST_DEBUG_OBJECT (faac, "average bitrate: %lu kbps",
(conf->bitRate + 500) / 1000 * faac->channels);
GST_DEBUG_OBJECT (faac, "quantization quality: %ld", conf->quantqual);
GST_DEBUG_OBJECT (faac, "bandwidth: %d Hz", conf->bandWidth);
/* now create a caps for it all */
srccaps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, mpegversion,
"channels", G_TYPE_INT, faac->channels,
"rate", G_TYPE_INT, faac->samplerate,
"stream-format", G_TYPE_STRING, (faac->outputformat ? "adts" : "raw"),
NULL);
if (!faac->outputformat) {
GstBuffer *codec_data;
guint8 *config = NULL;
gulong config_len = 0;
/* get the config string */
GST_DEBUG_OBJECT (faac, "retrieving decoder info");
faacEncGetDecoderSpecificInfo (faac->handle, &config, &config_len);
/* copy it into a buffer */
codec_data = gst_buffer_new_and_alloc (config_len);
memcpy (GST_BUFFER_DATA (codec_data), config, config_len);
free (config);
/* add to caps */
gst_caps_set_simple (srccaps,
"codec_data", GST_TYPE_BUFFER, codec_data, NULL);
gst_buffer_unref (codec_data);
}
GST_DEBUG_OBJECT (faac, "src pad caps: %" GST_PTR_FORMAT, srccaps);
ret = gst_pad_set_caps (faac->srcpad, srccaps);
gst_caps_unref (srccaps);
return ret;
/* ERROR */
empty_caps:
{
gst_caps_unref (allowed_caps);
return FALSE;
}
set_failed:
{
GST_WARNING_OBJECT (faac, "Faac doesn't support the current configuration");
return FALSE;
}
}
static GstFlowReturn
gst_faac_push_buffers (GstFaac * faac, gboolean force)
{
GstFlowReturn ret = GST_FLOW_OK;
gint av, frame_size, size, ret_size;
GstBuffer *outbuf;
guint64 timestamp, distance;
const guint8 *data;
/* samples already considers channel count */
frame_size = faac->samples * faac->bps;
while (G_LIKELY (ret == GST_FLOW_OK)) {
av = gst_adapter_available (faac->adapter);
GST_LOG_OBJECT (faac, "pushing: force: %d, frame_size: %d, av: %d, "
"offset: %d", force, frame_size, av, faac->offset);
/* idea:
* - start of adapter corresponds with what has already been encoded
* (i.e. really returned by faac)
* - start + offset is what needs to be fed to faac next
* That way we can timestamp the output based
* on adapter provided timestamp (and duration is a fixed frame duration) */
/* not enough data for one frame and no flush forcing */
if (!force && (av < frame_size + faac->offset))
break;
if (G_LIKELY (av - faac->offset >= frame_size)) {
GST_LOG_OBJECT (faac, "encoding a frame");
data = gst_adapter_peek (faac->adapter, faac->offset + frame_size);
data += faac->offset;
size = frame_size;
} else if (av - faac->offset > 0) {
GST_LOG_OBJECT (faac, "encoding leftover");
data = gst_adapter_peek (faac->adapter, av);
data += faac->offset;
size = av - faac->offset;
} else {
GST_LOG_OBJECT (faac, "emptying encoder");
data = NULL;
size = 0;
}
outbuf = gst_buffer_new_and_alloc (faac->bytes);
if (G_UNLIKELY ((ret_size = faacEncEncode (faac->handle, (gint32 *) data,
size / faac->bps, GST_BUFFER_DATA (outbuf),
faac->bytes)) < 0)) {
gst_buffer_unref (outbuf);
goto encode_failed;
}
GST_LOG_OBJECT (faac, "encoder return: %d", ret_size);
/* consumed, advanced view */
faac->offset += size;
g_assert (faac->offset <= av);
if (G_UNLIKELY (!ret_size)) {
gst_buffer_unref (outbuf);
if (size)
continue;
else
break;
}
/* deal with encoder lead-out */
if (G_UNLIKELY (av == 0 && faac->offset == 0)) {
GST_DEBUG_OBJECT (faac, "encoder returned additional data");
/* continuous with previous output, ok to have 0 duration */
timestamp = faac->next_ts;
} else {
/* after some caching, finally some data */
/* adapter gives time */
timestamp = gst_adapter_prev_timestamp (faac->adapter, &distance);
}
if (G_LIKELY ((av = gst_adapter_available (faac->adapter)) >= frame_size)) {
/* must have then come from a complete frame */
gst_adapter_flush (faac->adapter, frame_size);
faac->offset -= frame_size;
size = frame_size;
} else {
/* otherwise leftover */
gst_adapter_clear (faac->adapter);
faac->offset = 0;
size = av;
}
GST_BUFFER_SIZE (outbuf) = ret_size;
if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (timestamp)))
GST_BUFFER_TIMESTAMP (outbuf) = timestamp +
GST_FRAMES_TO_CLOCK_TIME (distance / faac->channels / faac->bps,
faac->samplerate);
GST_BUFFER_DURATION (outbuf) =
GST_FRAMES_TO_CLOCK_TIME (size / faac->channels / faac->bps,
faac->samplerate);
faac->next_ts =
GST_BUFFER_TIMESTAMP (outbuf) + GST_BUFFER_DURATION (outbuf);
/* perhaps check/set DISCONT based on timestamps ? */
GST_LOG_OBJECT (faac, "Pushing out buffer time: %" GST_TIME_FORMAT
" duration: %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)));
gst_buffer_set_caps (outbuf, GST_PAD_CAPS (faac->srcpad));
ret = gst_pad_push (faac->srcpad, outbuf);
}
/* in case encoder returns less than expected, clear our view as well */
if (G_UNLIKELY (force)) {
#ifndef GST_DISABLE_GST_DEBUG
if ((av = gst_adapter_available (faac->adapter)))
GST_WARNING_OBJECT (faac, "encoder left %d bytes; discarding", av);
#endif
gst_adapter_clear (faac->adapter);
faac->offset = 0;
}
return ret;
/* ERRORS */
encode_failed:
{
GST_ELEMENT_ERROR (faac, LIBRARY, ENCODE, (NULL), (NULL));
gst_buffer_unref (outbuf);
return GST_FLOW_ERROR;
}
}
static gboolean
gst_faac_sink_event (GstPad * pad, GstEvent * event)
{
GstFaac *faac;
gboolean ret;
faac = GST_FAAC (gst_pad_get_parent (pad));
GST_LOG_OBJECT (faac, "received %s", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
{
if (faac->handle) {
/* flush first */
GST_DEBUG_OBJECT (faac, "Pushing out remaining buffers because of EOS");
gst_faac_push_buffers (faac, TRUE);
}
ret = gst_pad_event_default (pad, event);
break;
}
default:
ret = gst_pad_event_default (pad, event);
break;
}
gst_object_unref (faac);
return ret;
}
static GstFlowReturn
gst_faac_chain (GstPad * pad, GstBuffer * inbuf)
{
GstFlowReturn result = GST_FLOW_OK;
GstFaac *faac;
faac = GST_FAAC (gst_pad_get_parent (pad));
if (!faac->handle)
goto no_handle;
GST_LOG_OBJECT (faac, "Got buffer time: %" GST_TIME_FORMAT " duration: %"
GST_TIME_FORMAT, GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (inbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (inbuf)));
gst_adapter_push (faac->adapter, inbuf);
result = gst_faac_push_buffers (faac, FALSE);
done:
gst_object_unref (faac);
return result;
/* ERRORS */
no_handle:
{
GST_ELEMENT_ERROR (faac, CORE, NEGOTIATION, (NULL),
("format wasn't negotiated before chain function"));
gst_buffer_unref (inbuf);
result = GST_FLOW_ERROR;
goto done;
}
}
static void
gst_faac_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstFaac *faac = GST_FAAC (object);
GST_OBJECT_LOCK (faac);
switch (prop_id) {
case ARG_BITRATE:
faac->bitrate = g_value_get_int (value);
break;
case ARG_PROFILE:
faac->profile = g_value_get_enum (value);
break;
case ARG_TNS:
faac->tns = g_value_get_boolean (value);
break;
case ARG_MIDSIDE:
faac->midside = g_value_get_boolean (value);
break;
case ARG_SHORTCTL:
faac->shortctl = g_value_get_enum (value);
break;
case ARG_OUTPUTFORMAT:
faac->outputformat = g_value_get_enum (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (faac);
}
static void
gst_faac_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstFaac *faac = GST_FAAC (object);
GST_OBJECT_LOCK (faac);
switch (prop_id) {
case ARG_BITRATE:
g_value_set_int (value, faac->bitrate);
break;
case ARG_PROFILE:
g_value_set_enum (value, faac->profile);
break;
case ARG_TNS:
g_value_set_boolean (value, faac->tns);
break;
case ARG_MIDSIDE:
g_value_set_boolean (value, faac->midside);
break;
case ARG_SHORTCTL:
g_value_set_enum (value, faac->shortctl);
break;
case ARG_OUTPUTFORMAT:
g_value_set_enum (value, faac->outputformat);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
GST_OBJECT_UNLOCK (faac);
}
static GstStateChangeReturn
gst_faac_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
GstFaac *faac = GST_FAAC (element);
/* upwards state changes */
switch (transition) {
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
/* downwards state changes */
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_READY:
{
gst_faac_close_encoder (faac);
gst_faac_reset (faac);
break;
}
default:
break;
}
return ret;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "faac", GST_RANK_SECONDARY,
GST_TYPE_FAAC);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"faac",
"Free AAC Encoder (FAAC)",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
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