mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-27 12:11:13 +00:00
6517282af7
It is faster than doing a query that propagates downstream and should be enough Elements: faac, gsmenc, opusenc, sbcenc, voamrwbenc, adpcmenc, sirenenc
371 lines
12 KiB
C
371 lines
12 KiB
C
/* GStreamer SBC audio encoder
|
|
* Copyright (C) 2004-2010 Marcel Holtmann <marcel@holtmann.org>
|
|
* Copyright (C) 2013 Tim-Philipp Müller <tim centricular net>
|
|
*
|
|
* This library is free software; you can redistribute it and/or
|
|
* modify it under the terms of the GNU Lesser General Public
|
|
* License as published by the Free Software Foundation; either
|
|
* version 2.1 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
|
|
* Lesser General Public License for more details.
|
|
*
|
|
* You should have received a copy of the GNU Lesser 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-sbenc
|
|
*
|
|
* This element encodes raw integer PCM audio into a Bluetooth SBC audio.
|
|
*
|
|
* Encoding paramets such as blocks, subbands, bitpool, channel-mode, and
|
|
* allocation-mode can be set by adding a capsfilter element with appropriate
|
|
* filtercaps after the sbcenc encoder element.
|
|
*
|
|
* <refsect2>
|
|
* <title>Example pipelines</title>
|
|
* |[
|
|
* gst-launch-1.0 -v audiotestsrc ! sbcenc ! rtpsbcpay ! udpsink
|
|
* ]| Encode a sine wave into SBC, RTP payload it and send over the network using UDP
|
|
* </refsect2>
|
|
*/
|
|
|
|
#ifdef HAVE_CONFIG_H
|
|
#include <config.h>
|
|
#endif
|
|
|
|
#include <string.h>
|
|
|
|
#include "gstsbcenc.h"
|
|
|
|
GST_DEBUG_CATEGORY_STATIC (sbc_enc_debug);
|
|
#define GST_CAT_DEFAULT sbc_enc_debug
|
|
|
|
G_DEFINE_TYPE (GstSbcEnc, gst_sbc_enc, GST_TYPE_AUDIO_ENCODER);
|
|
|
|
static GstStaticPadTemplate sbc_enc_sink_factory =
|
|
GST_STATIC_PAD_TEMPLATE ("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-raw, format=" GST_AUDIO_NE (S16) ", "
|
|
"rate = (int) { 16000, 32000, 44100, 48000 }, "
|
|
"channels = (int) [ 1, 2 ]"));
|
|
|
|
static GstStaticPadTemplate sbc_enc_src_factory =
|
|
GST_STATIC_PAD_TEMPLATE ("src", GST_PAD_SRC, GST_PAD_ALWAYS,
|
|
GST_STATIC_CAPS ("audio/x-sbc, "
|
|
"rate = (int) { 16000, 32000, 44100, 48000 }, "
|
|
"channels = (int) [ 1, 2 ], "
|
|
"channel-mode = (string) { mono, dual, stereo, joint }, "
|
|
"blocks = (int) { 4, 8, 12, 16 }, "
|
|
"subbands = (int) { 4, 8 }, "
|
|
"allocation-method = (string) { snr, loudness }, "
|
|
"bitpool = (int) [ 2, 64 ]"));
|
|
|
|
|
|
static gboolean gst_sbc_enc_start (GstAudioEncoder * enc);
|
|
static gboolean gst_sbc_enc_stop (GstAudioEncoder * enc);
|
|
static gboolean gst_sbc_enc_set_format (GstAudioEncoder * enc,
|
|
GstAudioInfo * info);
|
|
static GstFlowReturn gst_sbc_enc_handle_frame (GstAudioEncoder * enc,
|
|
GstBuffer * buffer);
|
|
|
|
static gboolean
|
|
gst_sbc_enc_set_format (GstAudioEncoder * audio_enc, GstAudioInfo * info)
|
|
{
|
|
const gchar *allocation_method, *channel_mode;
|
|
GstSbcEnc *enc = GST_SBC_ENC (audio_enc);
|
|
GstStructure *s;
|
|
GstCaps *caps, *filter_caps;
|
|
GstCaps *output_caps = NULL;
|
|
guint sampleframes_per_frame;
|
|
|
|
enc->rate = GST_AUDIO_INFO_RATE (info);
|
|
enc->channels = GST_AUDIO_INFO_CHANNELS (info);
|
|
|
|
/* negotiate output format based on downstream caps restrictions */
|
|
caps = gst_pad_get_allowed_caps (GST_AUDIO_ENCODER_SRC_PAD (enc));
|
|
if (caps == GST_CAPS_NONE || gst_caps_is_empty (caps))
|
|
goto failure;
|
|
|
|
if (caps == NULL)
|
|
caps = gst_static_pad_template_get_caps (&sbc_enc_src_factory);
|
|
|
|
/* fixate output caps */
|
|
filter_caps = gst_caps_new_simple ("audio/x-sbc", "rate", G_TYPE_INT,
|
|
enc->rate, "channels", G_TYPE_INT, enc->channels, NULL);
|
|
output_caps = gst_caps_intersect (caps, filter_caps);
|
|
gst_caps_unref (filter_caps);
|
|
|
|
if (output_caps == NULL || gst_caps_is_empty (output_caps)) {
|
|
GST_WARNING_OBJECT (enc, "Couldn't negotiate output caps with input rate "
|
|
"%d and input channels %d and allowed output caps %" GST_PTR_FORMAT,
|
|
enc->rate, enc->channels, caps);
|
|
goto failure;
|
|
}
|
|
|
|
gst_caps_unref (caps);
|
|
caps = NULL;
|
|
|
|
GST_DEBUG_OBJECT (enc, "fixating caps %" GST_PTR_FORMAT, output_caps);
|
|
output_caps = gst_caps_truncate (output_caps);
|
|
s = gst_caps_get_structure (output_caps, 0);
|
|
if (enc->channels == 1)
|
|
gst_structure_fixate_field_string (s, "channel-mode", "mono");
|
|
else
|
|
gst_structure_fixate_field_string (s, "channel-mode", "joint");
|
|
|
|
gst_structure_fixate_field_nearest_int (s, "bitpool", 64);
|
|
gst_structure_fixate_field_nearest_int (s, "blocks", 16);
|
|
gst_structure_fixate_field_nearest_int (s, "subbands", 8);
|
|
gst_structure_fixate_field_string (s, "allocation-method", "loudness");
|
|
s = NULL;
|
|
|
|
/* in case there's anything else left to fixate */
|
|
output_caps = gst_caps_fixate (output_caps);
|
|
gst_caps_set_simple (output_caps, "parsed", G_TYPE_BOOLEAN, TRUE, NULL);
|
|
|
|
GST_INFO_OBJECT (enc, "output caps %" GST_PTR_FORMAT, output_caps);
|
|
|
|
/* let's see what we fixated to */
|
|
s = gst_caps_get_structure (output_caps, 0);
|
|
gst_structure_get_int (s, "blocks", &enc->blocks);
|
|
gst_structure_get_int (s, "subbands", &enc->subbands);
|
|
gst_structure_get_int (s, "bitpool", &enc->bitpool);
|
|
allocation_method = gst_structure_get_string (s, "allocation-method");
|
|
channel_mode = gst_structure_get_string (s, "channel-mode");
|
|
|
|
/* We want channel-mode and channels coherent */
|
|
if (enc->channels == 1) {
|
|
if (g_strcmp0 (channel_mode, "mono") != 0) {
|
|
GST_ERROR_OBJECT (enc, "Can't have channel-mode '%s' for 1 channel",
|
|
channel_mode);
|
|
goto failure;
|
|
}
|
|
} else {
|
|
if (g_strcmp0 (channel_mode, "joint") != 0 &&
|
|
g_strcmp0 (channel_mode, "stereo") != 0 &&
|
|
g_strcmp0 (channel_mode, "dual") != 0) {
|
|
GST_ERROR_OBJECT (enc, "Can't have channel-mode '%s' for 2 channels",
|
|
channel_mode);
|
|
goto failure;
|
|
}
|
|
}
|
|
|
|
/* we want to be handed all available samples in handle_frame, but always
|
|
* enough to encode a frame */
|
|
sampleframes_per_frame = enc->blocks * enc->subbands;
|
|
gst_audio_encoder_set_frame_samples_min (audio_enc, sampleframes_per_frame);
|
|
gst_audio_encoder_set_frame_samples_max (audio_enc, sampleframes_per_frame);
|
|
gst_audio_encoder_set_frame_max (audio_enc, 0);
|
|
|
|
/* FIXME: what to do with left-over samples at the end? can we encode them? */
|
|
gst_audio_encoder_set_hard_min (audio_enc, TRUE);
|
|
|
|
/* and configure encoder based on the output caps we negotiated */
|
|
if (enc->rate == 16000)
|
|
enc->sbc.frequency = SBC_FREQ_16000;
|
|
else if (enc->rate == 32000)
|
|
enc->sbc.frequency = SBC_FREQ_32000;
|
|
else if (enc->rate == 44100)
|
|
enc->sbc.frequency = SBC_FREQ_44100;
|
|
else if (enc->rate == 48000)
|
|
enc->sbc.frequency = SBC_FREQ_48000;
|
|
else
|
|
goto failure;
|
|
|
|
if (enc->blocks == 4)
|
|
enc->sbc.blocks = SBC_BLK_4;
|
|
else if (enc->blocks == 8)
|
|
enc->sbc.blocks = SBC_BLK_8;
|
|
else if (enc->blocks == 12)
|
|
enc->sbc.blocks = SBC_BLK_12;
|
|
else if (enc->blocks == 16)
|
|
enc->sbc.blocks = SBC_BLK_16;
|
|
else
|
|
goto failure;
|
|
|
|
enc->sbc.subbands = (enc->subbands == 4) ? SBC_SB_4 : SBC_SB_8;
|
|
enc->sbc.bitpool = enc->bitpool;
|
|
|
|
if (channel_mode == NULL || allocation_method == NULL)
|
|
goto failure;
|
|
|
|
if (strcmp (channel_mode, "joint") == 0)
|
|
enc->sbc.mode = SBC_MODE_JOINT_STEREO;
|
|
else if (strcmp (channel_mode, "stereo") == 0)
|
|
enc->sbc.mode = SBC_MODE_STEREO;
|
|
else if (strcmp (channel_mode, "dual") == 0)
|
|
enc->sbc.mode = SBC_MODE_DUAL_CHANNEL;
|
|
else if (strcmp (channel_mode, "mono") == 0)
|
|
enc->sbc.mode = SBC_MODE_MONO;
|
|
else if (strcmp (channel_mode, "auto") == 0)
|
|
enc->sbc.mode = SBC_MODE_JOINT_STEREO;
|
|
else
|
|
goto failure;
|
|
|
|
if (strcmp (allocation_method, "loudness") == 0)
|
|
enc->sbc.allocation = SBC_AM_LOUDNESS;
|
|
else if (strcmp (allocation_method, "snr") == 0)
|
|
enc->sbc.allocation = SBC_AM_SNR;
|
|
else
|
|
goto failure;
|
|
|
|
if (!gst_audio_encoder_set_output_format (audio_enc, output_caps))
|
|
goto failure;
|
|
|
|
return gst_audio_encoder_negotiate (audio_enc);
|
|
|
|
failure:
|
|
if (output_caps)
|
|
gst_caps_unref (output_caps);
|
|
if (caps)
|
|
gst_caps_unref (caps);
|
|
return FALSE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_sbc_enc_handle_frame (GstAudioEncoder * audio_enc, GstBuffer * buffer)
|
|
{
|
|
GstSbcEnc *enc = GST_SBC_ENC (audio_enc);
|
|
GstMapInfo in_map, out_map;
|
|
GstBuffer *outbuf = NULL;
|
|
guint samples_per_frame, frames, i = 0;
|
|
|
|
/* no fancy draining */
|
|
if (buffer == NULL)
|
|
return GST_FLOW_OK;
|
|
|
|
if (G_UNLIKELY (enc->channels == 0 || enc->blocks == 0 || enc->subbands == 0))
|
|
return GST_FLOW_NOT_NEGOTIATED;
|
|
|
|
samples_per_frame = enc->channels * enc->blocks * enc->subbands;
|
|
|
|
if (!gst_buffer_map (buffer, &in_map, GST_MAP_READ))
|
|
goto map_failed;
|
|
|
|
frames = in_map.size / (samples_per_frame * sizeof (gint16));
|
|
|
|
GST_LOG_OBJECT (enc,
|
|
"encoding %" G_GSIZE_FORMAT " samples into %u SBC frames",
|
|
in_map.size / (enc->channels * sizeof (gint16)), frames);
|
|
|
|
if (frames > 0) {
|
|
gsize frame_len;
|
|
|
|
frame_len = sbc_get_frame_length (&enc->sbc);
|
|
outbuf = gst_audio_encoder_allocate_output_buffer (audio_enc,
|
|
frames * frame_len);
|
|
|
|
if (outbuf == NULL)
|
|
goto no_buffer;
|
|
|
|
gst_buffer_map (outbuf, &out_map, GST_MAP_WRITE);
|
|
|
|
for (i = 0; i < frames; ++i) {
|
|
gssize ret, written = 0;
|
|
|
|
ret = sbc_encode (&enc->sbc, in_map.data + (i * samples_per_frame * 2),
|
|
samples_per_frame * 2, out_map.data + (i * frame_len), frame_len,
|
|
&written);
|
|
|
|
if (ret < 0 || written != frame_len) {
|
|
GST_WARNING_OBJECT (enc, "encoding error, ret = %" G_GSSIZE_FORMAT ", "
|
|
"written = %" G_GSSIZE_FORMAT, ret, written);
|
|
break;
|
|
}
|
|
}
|
|
|
|
gst_buffer_unmap (outbuf, &out_map);
|
|
|
|
if (i > 0)
|
|
gst_buffer_set_size (outbuf, i * frame_len);
|
|
else
|
|
gst_buffer_replace (&outbuf, NULL);
|
|
}
|
|
|
|
done:
|
|
|
|
gst_buffer_unmap (buffer, &in_map);
|
|
|
|
return gst_audio_encoder_finish_frame (audio_enc, outbuf,
|
|
i * (samples_per_frame / enc->channels));
|
|
|
|
/* ERRORS */
|
|
no_buffer:
|
|
{
|
|
GST_ERROR_OBJECT (enc, "could not allocate output buffer");
|
|
goto done;
|
|
}
|
|
map_failed:
|
|
{
|
|
GST_ERROR_OBJECT (enc, "could not map input buffer");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
gst_sbc_enc_start (GstAudioEncoder * audio_enc)
|
|
{
|
|
GstSbcEnc *enc = GST_SBC_ENC (audio_enc);
|
|
|
|
GST_INFO_OBJECT (enc, "Setup subband codec");
|
|
sbc_init (&enc->sbc, 0);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_sbc_enc_stop (GstAudioEncoder * audio_enc)
|
|
{
|
|
GstSbcEnc *enc = GST_SBC_ENC (audio_enc);
|
|
|
|
GST_INFO_OBJECT (enc, "Finish subband codec");
|
|
sbc_finish (&enc->sbc);
|
|
|
|
enc->subbands = 0;
|
|
enc->blocks = 0;
|
|
enc->rate = 0;
|
|
enc->channels = 0;
|
|
enc->bitpool = 0;
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_sbc_enc_class_init (GstSbcEncClass * klass)
|
|
{
|
|
GstAudioEncoderClass *encoder_class = GST_AUDIO_ENCODER_CLASS (klass);
|
|
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
|
|
|
|
encoder_class->start = GST_DEBUG_FUNCPTR (gst_sbc_enc_start);
|
|
encoder_class->stop = GST_DEBUG_FUNCPTR (gst_sbc_enc_stop);
|
|
encoder_class->set_format = GST_DEBUG_FUNCPTR (gst_sbc_enc_set_format);
|
|
encoder_class->handle_frame = GST_DEBUG_FUNCPTR (gst_sbc_enc_handle_frame);
|
|
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&sbc_enc_sink_factory));
|
|
|
|
gst_element_class_add_pad_template (element_class,
|
|
gst_static_pad_template_get (&sbc_enc_src_factory));
|
|
|
|
gst_element_class_set_static_metadata (element_class,
|
|
"Bluetooth SBC audio encoder", "Codec/Encoder/Audio",
|
|
"Encode an SBC audio stream", "Marcel Holtmann <marcel@holtmann.org>");
|
|
|
|
GST_DEBUG_CATEGORY_INIT (sbc_enc_debug, "sbcenc", 0, "SBC encoding element");
|
|
}
|
|
|
|
static void
|
|
gst_sbc_enc_init (GstSbcEnc * self)
|
|
{
|
|
GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (self));
|
|
self->subbands = 0;
|
|
self->blocks = 0;
|
|
self->rate = 0;
|
|
self->channels = 0;
|
|
self->bitpool = 0;
|
|
}
|