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Where applicable, remove methods that don't do anything different than the default implementation.
237 lines
6.5 KiB
C
237 lines
6.5 KiB
C
/*
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* Siren Encoder Gst Element
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*
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* @author: Youness Alaoui <kakaroto@kakaroto.homelinux.net>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*
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*/
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/**
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* SECTION:element-sirenenc
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*
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* This encodes audio buffers into the Siren 16 codec (a 16khz extension of
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* G.722.1) that is meant to be compatible with the Microsoft Windows Live
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* Messenger(tm) implementation.
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*
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* Ref: http://www.polycom.com/company/about_us/technology/siren_g7221/index.html
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include "gstsirenenc.h"
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#include <string.h>
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GST_DEBUG_CATEGORY (sirenenc_debug);
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#define GST_CAT_DEFAULT (sirenenc_debug)
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#define FRAME_DURATION (20 * GST_MSECOND)
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static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-siren, " "dct-length = (int) 320"));
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static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("audio/x-raw, format = (string) \"S16LE\", "
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"rate = (int) 16000, " "channels = (int) 1"));
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static gboolean gst_siren_enc_start (GstAudioEncoder * enc);
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static gboolean gst_siren_enc_stop (GstAudioEncoder * enc);
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static gboolean gst_siren_enc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_siren_enc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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G_DEFINE_TYPE (GstSirenEnc, gst_siren_enc, GST_TYPE_AUDIO_ENCODER);
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static void
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gst_siren_enc_class_init (GstSirenEncClass * klass)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
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GstAudioEncoderClass *base_class = GST_AUDIO_ENCODER_CLASS (klass);
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GST_DEBUG_CATEGORY_INIT (sirenenc_debug, "sirenenc", 0, "sirenenc");
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&srctemplate));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sinktemplate));
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gst_element_class_set_static_metadata (element_class, "Siren Encoder element",
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"Codec/Encoder/Audio ",
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"Encode 16bit PCM streams into the Siren7 codec",
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"Youness Alaoui <kakaroto@kakaroto.homelinux.net>");
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base_class->start = GST_DEBUG_FUNCPTR (gst_siren_enc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_siren_enc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_siren_enc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_siren_enc_handle_frame);
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GST_DEBUG ("Class Init done");
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}
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static void
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gst_siren_enc_init (GstSirenEnc * enc)
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{
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}
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static gboolean
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gst_siren_enc_start (GstAudioEncoder * enc)
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{
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GstSirenEnc *senc = GST_SIREN_ENC (enc);
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GST_DEBUG_OBJECT (enc, "start");
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senc->encoder = Siren7_NewEncoder (16000);
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return TRUE;
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}
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static gboolean
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gst_siren_enc_stop (GstAudioEncoder * enc)
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{
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GstSirenEnc *senc = GST_SIREN_ENC (enc);
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GST_DEBUG_OBJECT (senc, "stop");
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Siren7_CloseEncoder (senc->encoder);
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return TRUE;
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}
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static gboolean
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gst_siren_enc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
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{
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gboolean res;
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GstCaps *outcaps;
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outcaps = gst_static_pad_template_get_caps (&srctemplate);
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res = gst_audio_encoder_set_output_format (benc, outcaps);
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gst_caps_unref (outcaps);
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/* report needs to base class */
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gst_audio_encoder_set_frame_samples_min (benc, 320);
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gst_audio_encoder_set_frame_samples_max (benc, 320);
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/* no remainder or flushing please */
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gst_audio_encoder_set_hard_min (benc, TRUE);
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gst_audio_encoder_set_drainable (benc, FALSE);
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return res;
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}
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static GstFlowReturn
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gst_siren_enc_handle_frame (GstAudioEncoder * benc, GstBuffer * buf)
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{
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GstSirenEnc *enc;
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GstFlowReturn ret = GST_FLOW_OK;
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GstBuffer *out_buf;
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guint8 *in_data, *out_data;
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guint i, size, num_frames;
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gint out_size;
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#ifndef GST_DISABLE_GST_DEBUG
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gint in_size;
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#endif
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gint encode_ret;
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GstMapInfo inmap, outmap;
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g_return_val_if_fail (buf != NULL, GST_FLOW_ERROR);
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enc = GST_SIREN_ENC (benc);
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size = gst_buffer_get_size (buf);
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GST_LOG_OBJECT (enc, "Received buffer of size %d", size);
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g_return_val_if_fail (size > 0, GST_FLOW_ERROR);
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g_return_val_if_fail (size % 640 == 0, GST_FLOW_ERROR);
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/* we need to process 640 input bytes to produce 40 output bytes */
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/* calculate the amount of frames we will handle */
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num_frames = size / 640;
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/* this is the input/output size */
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#ifndef GST_DISABLE_GST_DEBUG
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in_size = num_frames * 640;
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#endif
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out_size = num_frames * 40;
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GST_LOG_OBJECT (enc, "we have %u frames, %u in, %u out", num_frames, in_size,
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out_size);
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/* get a buffer */
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out_buf = gst_audio_encoder_allocate_output_buffer (benc, out_size);
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if (out_buf == NULL)
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goto alloc_failed;
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/* get the input data for all the frames */
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gst_buffer_map (buf, &inmap, GST_MAP_READ);
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gst_buffer_map (out_buf, &outmap, GST_MAP_READ);
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in_data = inmap.data;
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out_data = outmap.data;
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for (i = 0; i < num_frames; i++) {
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GST_LOG_OBJECT (enc, "Encoding frame %u/%u", i, num_frames);
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/* encode 640 input bytes to 40 output bytes */
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encode_ret = Siren7_EncodeFrame (enc->encoder, in_data, out_data);
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if (encode_ret != 0)
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goto encode_error;
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/* move to next frame */
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out_data += 40;
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in_data += 640;
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}
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gst_buffer_unmap (buf, &inmap);
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gst_buffer_unmap (out_buf, &outmap);
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GST_LOG_OBJECT (enc, "Finished encoding");
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/* we encode all we get, pass it along */
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ret = gst_audio_encoder_finish_frame (benc, out_buf, -1);
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done:
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return ret;
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/* ERRORS */
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alloc_failed:
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{
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GST_DEBUG_OBJECT (enc, "failed to pad_alloc buffer: %d (%s)", ret,
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gst_flow_get_name (ret));
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goto done;
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}
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encode_error:
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{
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GST_ELEMENT_ERROR (enc, STREAM, ENCODE, (NULL),
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("Error encoding frame: %d", encode_ret));
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ret = GST_FLOW_ERROR;
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gst_buffer_unref (out_buf);
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goto done;
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}
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}
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gboolean
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gst_siren_enc_plugin_init (GstPlugin * plugin)
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{
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return gst_element_register (plugin, "sirenenc",
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GST_RANK_MARGINAL, GST_TYPE_SIREN_ENC);
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}
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