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gstreamer/ext/voaacenc/gstvoaacenc.c
Tim-Philipp Müller 9e1b75fda3 Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
2012-11-04 00:09:59 +00:00

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/* GStreamer AAC encoder plugin
* Copyright (C) 2011 Kan Hu <kan.hu@linaro.org>
*
* 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-voaacenc
*
* AAC audio encoder based on vo-aacenc library
* <ulink url="http://sourceforge.net/projects/opencore-amr/files/vo-aacenc/">vo-aacenc library source file</ulink>.
*
* <refsect2>
* <title>Example launch line</title>
* |[
* gst-launch filesrc location=abc.wav ! wavparse ! audioresample ! audioconvert ! voaacenc ! filesink location=abc.aac
* ]|
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <string.h>
#include <gst/pbutils/codec-utils.h>
#include "gstvoaacenc.h"
#define VOAAC_ENC_DEFAULT_BITRATE (128000)
#define VOAAC_ENC_DEFAULT_OUTPUTFORMAT (0) /* RAW */
#define VOAAC_ENC_MPEGVERSION (4)
#define VOAAC_ENC_CODECDATA_LEN (2)
#define VOAAC_ENC_BITS_PER_SAMPLE (16)
enum
{
PROP_0,
PROP_BITRATE
};
#define SAMPLE_RATES " 8000, " \
"11025, " \
"12000, " \
"16000, " \
"22050, " \
"24000, " \
"32000, " \
"44100, " \
"48000, " \
"64000, " \
"88200, " \
"96000"
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " GST_AUDIO_NE (S16) ", "
"layout = (string) interleaved, "
"rate = (int) { " SAMPLE_RATES " }, " "channels = (int) [1, 2]")
);
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/mpeg, "
"mpegversion = (int) 4, "
"rate = (int) { " SAMPLE_RATES " }, "
"channels = (int) [1, 2], "
"stream-format = (string) { adts, raw }, " "base-profile = (string) lc")
);
GST_DEBUG_CATEGORY_STATIC (gst_voaacenc_debug);
#define GST_CAT_DEFAULT gst_voaacenc_debug
static gboolean voaacenc_core_init (GstVoAacEnc * voaacenc);
static gboolean voaacenc_core_set_parameter (GstVoAacEnc * voaacenc);
static void voaacenc_core_uninit (GstVoAacEnc * voaacenc);
static gboolean gst_voaacenc_start (GstAudioEncoder * enc);
static gboolean gst_voaacenc_stop (GstAudioEncoder * enc);
static gboolean gst_voaacenc_set_format (GstAudioEncoder * enc,
GstAudioInfo * info);
static GstFlowReturn gst_voaacenc_handle_frame (GstAudioEncoder * enc,
GstBuffer * in_buf);
static GstCaps *gst_voaacenc_getcaps (GstAudioEncoder * enc, GstCaps * filter);
G_DEFINE_TYPE (GstVoAacEnc, gst_voaacenc, GST_TYPE_AUDIO_ENCODER);
static void
gst_voaacenc_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstVoAacEnc *self = GST_VOAACENC (object);
switch (prop_id) {
case PROP_BITRATE:
self->bitrate = g_value_get_int (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
return;
}
static void
gst_voaacenc_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstVoAacEnc *self = GST_VOAACENC (object);
switch (prop_id) {
case PROP_BITRATE:
g_value_set_int (value, self->bitrate);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
return;
}
static void
gst_voaacenc_class_init (GstVoAacEncClass * klass)
{
GObjectClass *object_class = G_OBJECT_CLASS (klass);
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
object_class->set_property = GST_DEBUG_FUNCPTR (gst_voaacenc_set_property);
object_class->get_property = GST_DEBUG_FUNCPTR (gst_voaacenc_get_property);
base_class->start = GST_DEBUG_FUNCPTR (gst_voaacenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_voaacenc_stop);
base_class->set_format = GST_DEBUG_FUNCPTR (gst_voaacenc_set_format);
base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_voaacenc_handle_frame);
base_class->getcaps = GST_DEBUG_FUNCPTR (gst_voaacenc_getcaps);
g_object_class_install_property (object_class, PROP_BITRATE,
g_param_spec_int ("bitrate",
"Bitrate",
"Target Audio Bitrate",
0, G_MAXINT, VOAAC_ENC_DEFAULT_BITRATE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_set_static_metadata (element_class, "AAC audio encoder",
"Codec/Encoder/Audio", "AAC audio encoder", "Kan Hu <kan.hu@linaro.org>");
GST_DEBUG_CATEGORY_INIT (gst_voaacenc_debug, "voaacenc", 0, "voaac encoder");
}
static void
gst_voaacenc_init (GstVoAacEnc * voaacenc)
{
voaacenc->bitrate = VOAAC_ENC_DEFAULT_BITRATE;
voaacenc->output_format = VOAAC_ENC_DEFAULT_OUTPUTFORMAT;
/* init rest */
voaacenc->handle = NULL;
}
static gboolean
gst_voaacenc_start (GstAudioEncoder * enc)
{
GstVoAacEnc *voaacenc = GST_VOAACENC (enc);
GST_DEBUG_OBJECT (enc, "start");
if (voaacenc_core_init (voaacenc) == FALSE)
return FALSE;
voaacenc->rate = 0;
voaacenc->channels = 0;
return TRUE;
}
static gboolean
gst_voaacenc_stop (GstAudioEncoder * enc)
{
GstVoAacEnc *voaacenc = GST_VOAACENC (enc);
GST_DEBUG_OBJECT (enc, "stop");
voaacenc_core_uninit (voaacenc);
return TRUE;
}
#define VOAAC_ENC_MAX_CHANNELS 6
/* describe the channels position */
static const GstAudioChannelPosition
aac_channel_positions[][VOAAC_ENC_MAX_CHANNELS] = {
{ /* 1 ch: Mono */
GST_AUDIO_CHANNEL_POSITION_MONO},
{ /* 2 ch: front left + front right (front stereo) */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
{ /* 3 ch: front center + front stereo */
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
{ /* 4 ch: front center + front stereo + back center */
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER},
{ /* 5 ch: front center + front stereo + back stereo */
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},
{ /* 6ch: front center + front stereo + back stereo + LFE */
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_LFE1}
};
static gpointer
gst_voaacenc_generate_sink_caps (gpointer data)
{
GstCaps *caps;
gint i, c;
static const int rates[] = {
8000, 11025, 12000, 16000, 22050, 24000,
32000, 44100, 48000, 64000, 88200, 96000
};
GValue rates_arr = { 0, };
GValue tmp = { 0, };
GstStructure *s, *t;
g_value_init (&rates_arr, GST_TYPE_LIST);
g_value_init (&tmp, G_TYPE_INT);
for (i = 0; i < G_N_ELEMENTS (rates); i++) {
g_value_set_int (&tmp, rates[i]);
gst_value_list_append_value (&rates_arr, &tmp);
}
g_value_unset (&tmp);
s = gst_structure_new ("audio/x-raw",
"format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"layout", G_TYPE_STRING, "interleaved", NULL);
gst_structure_set_value (s, "rate", &rates_arr);
caps = gst_caps_new_empty ();
for (i = 1; i <= 2 /* VOAAC_ENC_MAX_CHANNELS */ ; i++) {
guint64 channel_mask = 0;
t = gst_structure_copy (s);
gst_structure_set (t, "channels", G_TYPE_INT, i, NULL);
if (i > 1) {
for (c = 0; c < i; c++)
channel_mask |=
G_GUINT64_CONSTANT (1) << aac_channel_positions[i - 1][c];
gst_structure_set (t, "channel-mask", GST_TYPE_BITMASK, channel_mask,
NULL);
}
gst_caps_append_structure (caps, t);
}
gst_structure_free (s);
g_value_unset (&rates_arr);
GST_DEBUG ("generated sink caps: %" GST_PTR_FORMAT, caps);
return caps;
}
static GstCaps *
gst_voaacenc_get_sink_caps (void)
{
static GOnce g_once = G_ONCE_INIT;
GstCaps *caps;
g_once (&g_once, gst_voaacenc_generate_sink_caps, NULL);
caps = g_once.retval;
return caps;
}
static GstCaps *
gst_voaacenc_getcaps (GstAudioEncoder * benc, GstCaps * filter)
{
return gst_audio_encoder_proxy_getcaps (benc, gst_voaacenc_get_sink_caps (),
filter);
}
/* check downstream caps to configure format */
static void
gst_voaacenc_negotiate (GstVoAacEnc * voaacenc)
{
GstCaps *caps;
caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (voaacenc));
GST_DEBUG_OBJECT (voaacenc, "allowed caps: %" GST_PTR_FORMAT, caps);
if (caps && gst_caps_get_size (caps) > 0) {
GstStructure *s = gst_caps_get_structure (caps, 0);
const gchar *str = NULL;
if ((str = gst_structure_get_string (s, "stream-format"))) {
if (strcmp (str, "adts") == 0) {
GST_DEBUG_OBJECT (voaacenc, "use ADTS format for output");
voaacenc->output_format = 1;
} else if (strcmp (str, "raw") == 0) {
GST_DEBUG_OBJECT (voaacenc, "use RAW format for output");
voaacenc->output_format = 0;
} else {
GST_DEBUG_OBJECT (voaacenc, "unknown stream-format: %s", str);
voaacenc->output_format = VOAAC_ENC_DEFAULT_OUTPUTFORMAT;
}
}
}
if (caps)
gst_caps_unref (caps);
}
static gint
gst_voaacenc_get_rate_index (gint rate)
{
static const gint rate_table[] = {
96000, 88200, 64000, 48000, 44100, 32000,
24000, 22050, 16000, 12000, 11025, 8000
};
gint i;
for (i = 0; i < G_N_ELEMENTS (rate_table); ++i) {
if (rate == rate_table[i]) {
return i;
}
}
return -1;
}
static GstCaps *
gst_voaacenc_create_source_pad_caps (GstVoAacEnc * voaacenc)
{
GstCaps *caps = NULL;
gint index;
GstBuffer *codec_data;
GstMapInfo map;
if ((index = gst_voaacenc_get_rate_index (voaacenc->rate)) >= 0) {
codec_data = gst_buffer_new_and_alloc (VOAAC_ENC_CODECDATA_LEN);
gst_buffer_map (codec_data, &map, GST_MAP_WRITE);
/* LC profile only */
map.data[0] = ((0x02 << 3) | (index >> 1));
map.data[1] = ((index & 0x01) << 7) | (voaacenc->channels << 3);
caps = gst_caps_new_simple ("audio/mpeg",
"mpegversion", G_TYPE_INT, VOAAC_ENC_MPEGVERSION,
"channels", G_TYPE_INT, voaacenc->channels,
"rate", G_TYPE_INT, voaacenc->rate,
"stream-format", G_TYPE_STRING,
(voaacenc->output_format ? "adts" : "raw")
, NULL);
gst_codec_utils_aac_caps_set_level_and_profile (caps, map.data,
VOAAC_ENC_CODECDATA_LEN);
gst_buffer_unmap (codec_data, &map);
if (!voaacenc->output_format) {
gst_caps_set_simple (caps, "codec_data", GST_TYPE_BUFFER, codec_data,
NULL);
}
gst_buffer_unref (codec_data);
}
return caps;
}
static gboolean
gst_voaacenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
{
gboolean ret = FALSE;
GstVoAacEnc *voaacenc;
GstCaps *src_caps;
voaacenc = GST_VOAACENC (benc);
/* get channel count */
voaacenc->channels = GST_AUDIO_INFO_CHANNELS (info);
voaacenc->rate = GST_AUDIO_INFO_RATE (info);
/* precalc buffer size as it's constant now */
voaacenc->inbuf_size = voaacenc->channels * 2 * 1024;
gst_voaacenc_negotiate (voaacenc);
/* create reverse caps */
src_caps = gst_voaacenc_create_source_pad_caps (voaacenc);
if (src_caps) {
gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (voaacenc),
src_caps);
gst_caps_unref (src_caps);
ret = voaacenc_core_set_parameter (voaacenc);
}
/* report needs to base class */
gst_audio_encoder_set_frame_samples_min (benc, 1024);
gst_audio_encoder_set_frame_samples_max (benc, 1024);
gst_audio_encoder_set_frame_max (benc, 1);
return ret;
}
static GstFlowReturn
gst_voaacenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
{
GstVoAacEnc *voaacenc;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *out;
VO_AUDIO_OUTPUTINFO output_info = { {0} };
VO_CODECBUFFER input = { 0 };
VO_CODECBUFFER output = { 0 };
GstMapInfo map, omap;
GstAudioInfo *info = gst_audio_encoder_get_audio_info (benc);
voaacenc = GST_VOAACENC (benc);
g_return_val_if_fail (voaacenc->handle, GST_FLOW_NOT_NEGOTIATED);
/* we don't deal with squeezing remnants, so simply discard those */
if (G_UNLIKELY (buf == NULL)) {
GST_DEBUG_OBJECT (benc, "no data");
goto exit;
}
if (memcmp (info->position, aac_channel_positions[info->channels - 1],
sizeof (GstAudioChannelPosition) * info->channels) != 0) {
buf = gst_buffer_make_writable (buf);
gst_audio_buffer_reorder_channels (buf, info->finfo->format,
info->channels, info->position,
aac_channel_positions[info->channels - 1]);
}
gst_buffer_map (buf, &map, GST_MAP_READ);
if (G_UNLIKELY (map.size < voaacenc->inbuf_size)) {
gst_buffer_unmap (buf, &map);
GST_DEBUG_OBJECT (voaacenc, "discarding trailing data %d", (gint) map.size);
ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
goto exit;
}
/* max size */
out = gst_buffer_new_and_alloc (voaacenc->inbuf_size);
gst_buffer_map (out, &omap, GST_MAP_WRITE);
output.Buffer = omap.data;
output.Length = voaacenc->inbuf_size;
g_assert (map.size == voaacenc->inbuf_size);
input.Buffer = map.data;
input.Length = voaacenc->inbuf_size;
voaacenc->codec_api.SetInputData (voaacenc->handle, &input);
/* encode */
if (voaacenc->codec_api.GetOutputData (voaacenc->handle, &output,
&output_info) != VO_ERR_NONE) {
gst_buffer_unmap (buf, &map);
gst_buffer_unmap (out, &omap);
gst_buffer_unref (out);
goto encode_failed;
}
GST_LOG_OBJECT (voaacenc, "encoded to %lu bytes", output.Length);
gst_buffer_unmap (buf, &map);
gst_buffer_unmap (out, &omap);
gst_buffer_resize (out, 0, output.Length);
ret = gst_audio_encoder_finish_frame (benc, out, 1024);
exit:
return ret;
/* ERRORS */
encode_failed:
{
GST_ELEMENT_ERROR (voaacenc, STREAM, ENCODE, (NULL), ("encode failed"));
ret = GST_FLOW_ERROR;
goto exit;
}
}
static VO_U32
voaacenc_core_mem_alloc (VO_S32 uID, VO_MEM_INFO * pMemInfo)
{
if (!pMemInfo)
return VO_ERR_INVALID_ARG;
pMemInfo->VBuffer = g_malloc (pMemInfo->Size);
return 0;
}
static VO_U32
voaacenc_core_mem_free (VO_S32 uID, VO_PTR pMem)
{
g_free (pMem);
return 0;
}
static VO_U32
voaacenc_core_mem_set (VO_S32 uID, VO_PTR pBuff, VO_U8 uValue, VO_U32 uSize)
{
memset (pBuff, uValue, uSize);
return 0;
}
static VO_U32
voaacenc_core_mem_copy (VO_S32 uID, VO_PTR pDest, VO_PTR pSource, VO_U32 uSize)
{
memcpy (pDest, pSource, uSize);
return 0;
}
static VO_U32
voaacenc_core_mem_check (VO_S32 uID, VO_PTR pBuffer, VO_U32 uSize)
{
return 0;
}
static gboolean
voaacenc_core_init (GstVoAacEnc * voaacenc)
{
VO_CODEC_INIT_USERDATA user_data = { 0 };
voGetAACEncAPI (&voaacenc->codec_api);
voaacenc->mem_operator.Alloc = voaacenc_core_mem_alloc;
voaacenc->mem_operator.Copy = voaacenc_core_mem_copy;
voaacenc->mem_operator.Free = voaacenc_core_mem_free;
voaacenc->mem_operator.Set = voaacenc_core_mem_set;
voaacenc->mem_operator.Check = voaacenc_core_mem_check;
user_data.memflag = VO_IMF_USERMEMOPERATOR;
user_data.memData = &voaacenc->mem_operator;
voaacenc->codec_api.Init (&voaacenc->handle, VO_AUDIO_CodingAAC, &user_data);
if (voaacenc->handle == NULL) {
return FALSE;
}
return TRUE;
}
static gboolean
voaacenc_core_set_parameter (GstVoAacEnc * voaacenc)
{
AACENC_PARAM params = { 0 };
guint32 ret;
params.sampleRate = voaacenc->rate;
params.bitRate = voaacenc->bitrate;
params.nChannels = voaacenc->channels;
if (voaacenc->output_format) {
params.adtsUsed = 1;
} else {
params.adtsUsed = 0;
}
ret =
voaacenc->codec_api.SetParam (voaacenc->handle, VO_PID_AAC_ENCPARAM,
&params);
if (ret != VO_ERR_NONE) {
GST_ERROR_OBJECT (voaacenc, "Failed to set encoder parameters");
return FALSE;
}
return TRUE;
}
static void
voaacenc_core_uninit (GstVoAacEnc * voaacenc)
{
if (voaacenc->handle) {
voaacenc->codec_api.Uninit (voaacenc->handle);
voaacenc->handle = NULL;
}
}
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