ext/amrnb/amrnbenc.c - gst-plugins-ugly1.0 - Git at Google

 /* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
  * Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.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., 51 Franklin St, Fifth Floor,
  * Boston, MA 02110-1301, USA.
  */

 /**
  * SECTION:element-amrnbenc
  * @see_also: #GstAmrnbDec, #GstAmrnbParse
  *
  * AMR narrowband encoder based on the
  * <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>.
  *
  * <refsect2>
  * <title>Example launch line</title>
  * |[
  * gst-launch-1.0 filesrc location=abc.wav ! wavparse ! audioconvert ! audioresample ! amrnbenc ! filesink location=abc.amr
  * ]|
  * Please note that the above stream misses the header, that is needed to play
  * the stream.
  * </refsect2>
  */

 #ifdef HAVE_CONFIG_H
 #include "config.h"
 #endif

 #include "amrnbenc.h"

 static GType
 gst_amrnbenc_bandmode_get_type (void)
 {
   static GType gst_amrnbenc_bandmode_type = 0;
   static const GEnumValue gst_amrnbenc_bandmode[] = {
     {MR475, "MR475", "MR475"},
     {MR515, "MR515", "MR515"},
     {MR59, "MR59", "MR59"},
     {MR67, "MR67", "MR67"},
     {MR74, "MR74", "MR74"},
     {MR795, "MR795", "MR795"},
     {MR102, "MR102", "MR102"},
     {MR122, "MR122", "MR122"},
     {MRDTX, "MRDTX", "MRDTX"},
     {0, NULL, NULL},
   };
   if (!gst_amrnbenc_bandmode_type) {
     gst_amrnbenc_bandmode_type =
         g_enum_register_static ("GstAmrnbEncBandMode", gst_amrnbenc_bandmode);
   }
   return gst_amrnbenc_bandmode_type;
 }

 #define GST_AMRNBENC_BANDMODE_TYPE (gst_amrnbenc_bandmode_get_type())

 #define BANDMODE_DEFAULT MR122
 enum
 {
   PROP_0,
   PROP_BANDMODE
 };

 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) 8000," "channels = (int) 1")
     );

 static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
     GST_PAD_SRC,
     GST_PAD_ALWAYS,
     GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1")
     );

 GST_DEBUG_CATEGORY_STATIC (gst_amrnbenc_debug);
 #define GST_CAT_DEFAULT gst_amrnbenc_debug

 static gboolean gst_amrnbenc_start (GstAudioEncoder * enc);
 static gboolean gst_amrnbenc_stop (GstAudioEncoder * enc);
 static gboolean gst_amrnbenc_set_format (GstAudioEncoder * enc,
     GstAudioInfo * info);
 static GstFlowReturn gst_amrnbenc_handle_frame (GstAudioEncoder * enc,
     GstBuffer * in_buf);

 #define gst_amrnbenc_parent_class parent_class
 G_DEFINE_TYPE (GstAmrnbEnc, gst_amrnbenc, GST_TYPE_AUDIO_ENCODER);

 static void
 gst_amrnbenc_set_property (GObject * object, guint prop_id,
     const GValue * value, GParamSpec * pspec)
 {
   GstAmrnbEnc *self = GST_AMRNBENC (object);

   switch (prop_id) {
     case PROP_BANDMODE:
       self->bandmode = g_value_get_enum (value);
       break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
   }
   return;
 }

 static void
 gst_amrnbenc_get_property (GObject * object, guint prop_id,
     GValue * value, GParamSpec * pspec)
 {
   GstAmrnbEnc *self = GST_AMRNBENC (object);

   switch (prop_id) {
     case PROP_BANDMODE:
       g_value_set_enum (value, self->bandmode);
       break;
     default:
       G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
       break;
   }
   return;
 }

 static void
 gst_amrnbenc_class_init (GstAmrnbEncClass * 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_amrnbenc_set_property;
   object_class->get_property = gst_amrnbenc_get_property;

   base_class->start = GST_DEBUG_FUNCPTR (gst_amrnbenc_start);
   base_class->stop = GST_DEBUG_FUNCPTR (gst_amrnbenc_stop);
   base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrnbenc_set_format);
   base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrnbenc_handle_frame);

   g_object_class_install_property (object_class, PROP_BANDMODE,
       g_param_spec_enum ("band-mode", "Band Mode",
           "Encoding Band Mode (Kbps)", GST_AMRNBENC_BANDMODE_TYPE,
           BANDMODE_DEFAULT,
           G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));

   gst_element_class_add_static_pad_template (element_class, &sink_template);
   gst_element_class_add_static_pad_template (element_class, &src_template);

   gst_element_class_set_static_metadata (element_class, "AMR-NB audio encoder",
       "Codec/Encoder/Audio",
       "Adaptive Multi-Rate Narrow-Band audio encoder",
       "Wim Taymans <wim.taymans@gmail.com>");

   GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0,
       "AMR-NB audio encoder");
 }

 static void
 gst_amrnbenc_init (GstAmrnbEnc * amrnbenc)
 {
   GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (amrnbenc));
 }

 static gboolean
 gst_amrnbenc_start (GstAudioEncoder * enc)
 {
   GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);

   GST_DEBUG_OBJECT (amrnbenc, "start");

   if (!(amrnbenc->handle = Encoder_Interface_init (0)))
     return FALSE;

   return TRUE;
 }

 static gboolean
 gst_amrnbenc_stop (GstAudioEncoder * enc)
 {
   GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);

   GST_DEBUG_OBJECT (amrnbenc, "stop");

   Encoder_Interface_exit (amrnbenc->handle);

   return TRUE;
 }

 static gboolean
 gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
 {
   GstAmrnbEnc *amrnbenc;
   GstCaps *copy;

   amrnbenc = GST_AMRNBENC (enc);

   /* parameters already parsed for us */
   amrnbenc->rate = GST_AUDIO_INFO_RATE (info);
   amrnbenc->channels = GST_AUDIO_INFO_CHANNELS (info);

   /* we do not really accept other input, but anyway ... */
   /* this is not wrong but will sound bad */
   if (amrnbenc->channels != 1) {
     g_warning ("amrnbdec is only optimized for mono channels");
   }
   if (amrnbenc->rate != 8000) {
     g_warning ("amrnbdec is only optimized for 8000 Hz samplerate");
   }

   /* create reverse caps */
   copy = gst_caps_new_simple ("audio/AMR",
       "channels", G_TYPE_INT, amrnbenc->channels,
       "rate", G_TYPE_INT, amrnbenc->rate, NULL);

   gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (amrnbenc), copy);
   gst_caps_unref (copy);

   /* report needs to base class: hand one frame at a time */
   gst_audio_encoder_set_frame_samples_min (enc, 160);
   gst_audio_encoder_set_frame_samples_max (enc, 160);
   gst_audio_encoder_set_frame_max (enc, 1);

   return TRUE;
 }

 static GstFlowReturn
 gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
 {
   GstAmrnbEnc *amrnbenc;
   GstFlowReturn ret;
   GstBuffer *out;
   GstMapInfo in_map, out_map;
   gsize out_size;

   amrnbenc = GST_AMRNBENC (enc);

   g_return_val_if_fail (amrnbenc->handle, GST_FLOW_FLUSHING);

   /* we don't deal with squeezing remnants, so simply discard those */
   if (G_UNLIKELY (buffer == NULL)) {
     GST_DEBUG_OBJECT (amrnbenc, "no data");
     return GST_FLOW_OK;
   }

   gst_buffer_map (buffer, &in_map, GST_MAP_READ);

   if (G_UNLIKELY (in_map.size < 320)) {
     gst_buffer_unmap (buffer, &in_map);
     GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data of %" G_GSIZE_FORMAT
         " bytes", in_map.size);
     return gst_audio_encoder_finish_frame (enc, NULL, -1);
   }

   /* get output, max size is 32 */
   out = gst_buffer_new_and_alloc (32);
   /* AMR encoder actually writes into the source data buffers it gets */
   /* should be able to handle that with what we are given */

   gst_buffer_map (out, &out_map, GST_MAP_WRITE);
   /* encode */
   out_size =
       Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
       (short *) in_map.data, out_map.data, 0);
   gst_buffer_unmap (out, &out_map);
   gst_buffer_resize (out, 0, out_size);
   gst_buffer_unmap (buffer, &in_map);

   GST_LOG_OBJECT (amrnbenc, "output data size %" G_GSIZE_FORMAT, out_size);

   if (out_size) {
     ret = gst_audio_encoder_finish_frame (enc, out, 160);
   } else {
     /* should not happen (without dtx or so at least) */
     GST_WARNING_OBJECT (amrnbenc, "no encoded data; discarding input");
     gst_buffer_unref (out);
     ret = gst_audio_encoder_finish_frame (enc, NULL, -1);
   }

   return ret;
 }
	/* GStreamer Adaptive Multi-Rate Narrow-Band (AMR-NB) plugin
	* Copyright (C) 2004 Ronald Bultje <rbultje@ronald.bitfreak.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., 51 Franklin St, Fifth Floor,
	* Boston, MA 02110-1301, USA.
	*/

	/**
	* SECTION:element-amrnbenc
	* @see_also: #GstAmrnbDec, #GstAmrnbParse
	*
	* AMR narrowband encoder based on the
	* <ulink url="http://sourceforge.net/projects/opencore-amr">opencore codec implementation</ulink>.
	*
	* <refsect2>
	* <title>Example launch line</title>
	* \|[
	* gst-launch-1.0 filesrc location=abc.wav ! wavparse ! audioconvert ! audioresample ! amrnbenc ! filesink location=abc.amr
	* ]\|
	* Please note that the above stream misses the header, that is needed to play
	* the stream.
	* </refsect2>
	*/

	#ifdef HAVE_CONFIG_H
	#include "config.h"
	#endif

	#include "amrnbenc.h"

	static GType
	gst_amrnbenc_bandmode_get_type (void)
	{
	static GType gst_amrnbenc_bandmode_type = 0;
	static const GEnumValue gst_amrnbenc_bandmode[] = {
	{MR475, "MR475", "MR475"},
	{MR515, "MR515", "MR515"},
	{MR59, "MR59", "MR59"},
	{MR67, "MR67", "MR67"},
	{MR74, "MR74", "MR74"},
	{MR795, "MR795", "MR795"},
	{MR102, "MR102", "MR102"},
	{MR122, "MR122", "MR122"},
	{MRDTX, "MRDTX", "MRDTX"},
	{0, NULL, NULL},
	};
	if (!gst_amrnbenc_bandmode_type) {
	gst_amrnbenc_bandmode_type =
	g_enum_register_static ("GstAmrnbEncBandMode", gst_amrnbenc_bandmode);
	}
	return gst_amrnbenc_bandmode_type;
	}

	#define GST_AMRNBENC_BANDMODE_TYPE (gst_amrnbenc_bandmode_get_type())

	#define BANDMODE_DEFAULT MR122
	enum
	{
	PROP_0,
	PROP_BANDMODE
	};

	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) 8000," "channels = (int) 1")
	);

	static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
	GST_PAD_SRC,
	GST_PAD_ALWAYS,
	GST_STATIC_CAPS ("audio/AMR, " "rate = (int) 8000, " "channels = (int) 1")
	);

	GST_DEBUG_CATEGORY_STATIC (gst_amrnbenc_debug);
	#define GST_CAT_DEFAULT gst_amrnbenc_debug

	static gboolean gst_amrnbenc_start (GstAudioEncoder * enc);
	static gboolean gst_amrnbenc_stop (GstAudioEncoder * enc);
	static gboolean gst_amrnbenc_set_format (GstAudioEncoder * enc,
	GstAudioInfo * info);
	static GstFlowReturn gst_amrnbenc_handle_frame (GstAudioEncoder * enc,
	GstBuffer * in_buf);

	#define gst_amrnbenc_parent_class parent_class
	G_DEFINE_TYPE (GstAmrnbEnc, gst_amrnbenc, GST_TYPE_AUDIO_ENCODER);

	static void
	gst_amrnbenc_set_property (GObject * object, guint prop_id,
	const GValue * value, GParamSpec * pspec)
	{
	GstAmrnbEnc *self = GST_AMRNBENC (object);

	switch (prop_id) {
	case PROP_BANDMODE:
	self->bandmode = g_value_get_enum (value);
	break;
	default:
	G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
	break;
	}
	return;
	}

	static void
	gst_amrnbenc_get_property (GObject * object, guint prop_id,
	GValue * value, GParamSpec * pspec)
	{
	GstAmrnbEnc *self = GST_AMRNBENC (object);

	switch (prop_id) {
	case PROP_BANDMODE:
	g_value_set_enum (value, self->bandmode);
	break;
	default:
	G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
	break;
	}
	return;
	}

	static void
	gst_amrnbenc_class_init (GstAmrnbEncClass * 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_amrnbenc_set_property;
	object_class->get_property = gst_amrnbenc_get_property;

	base_class->start = GST_DEBUG_FUNCPTR (gst_amrnbenc_start);
	base_class->stop = GST_DEBUG_FUNCPTR (gst_amrnbenc_stop);
	base_class->set_format = GST_DEBUG_FUNCPTR (gst_amrnbenc_set_format);
	base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_amrnbenc_handle_frame);

	g_object_class_install_property (object_class, PROP_BANDMODE,
	g_param_spec_enum ("band-mode", "Band Mode",
	"Encoding Band Mode (Kbps)", GST_AMRNBENC_BANDMODE_TYPE,
	BANDMODE_DEFAULT,
	G_PARAM_READWRITE \| G_PARAM_CONSTRUCT \| G_PARAM_STATIC_STRINGS));

	gst_element_class_add_static_pad_template (element_class, &sink_template);
	gst_element_class_add_static_pad_template (element_class, &src_template);

	gst_element_class_set_static_metadata (element_class, "AMR-NB audio encoder",
	"Codec/Encoder/Audio",
	"Adaptive Multi-Rate Narrow-Band audio encoder",
	"Wim Taymans <wim.taymans@gmail.com>");

	GST_DEBUG_CATEGORY_INIT (gst_amrnbenc_debug, "amrnbenc", 0,
	"AMR-NB audio encoder");
	}

	static void
	gst_amrnbenc_init (GstAmrnbEnc * amrnbenc)
	{
	GST_PAD_SET_ACCEPT_TEMPLATE (GST_AUDIO_ENCODER_SINK_PAD (amrnbenc));
	}

	static gboolean
	gst_amrnbenc_start (GstAudioEncoder * enc)
	{
	GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);

	GST_DEBUG_OBJECT (amrnbenc, "start");

	if (!(amrnbenc->handle = Encoder_Interface_init (0)))
	return FALSE;

	return TRUE;
	}

	static gboolean
	gst_amrnbenc_stop (GstAudioEncoder * enc)
	{
	GstAmrnbEnc *amrnbenc = GST_AMRNBENC (enc);

	GST_DEBUG_OBJECT (amrnbenc, "stop");

	Encoder_Interface_exit (amrnbenc->handle);

	return TRUE;
	}

	static gboolean
	gst_amrnbenc_set_format (GstAudioEncoder * enc, GstAudioInfo * info)
	{
	GstAmrnbEnc *amrnbenc;
	GstCaps *copy;

	amrnbenc = GST_AMRNBENC (enc);

	/* parameters already parsed for us */
	amrnbenc->rate = GST_AUDIO_INFO_RATE (info);
	amrnbenc->channels = GST_AUDIO_INFO_CHANNELS (info);

	/* we do not really accept other input, but anyway ... */
	/* this is not wrong but will sound bad */
	if (amrnbenc->channels != 1) {
	g_warning ("amrnbdec is only optimized for mono channels");
	}
	if (amrnbenc->rate != 8000) {
	g_warning ("amrnbdec is only optimized for 8000 Hz samplerate");
	}

	/* create reverse caps */
	copy = gst_caps_new_simple ("audio/AMR",
	"channels", G_TYPE_INT, amrnbenc->channels,
	"rate", G_TYPE_INT, amrnbenc->rate, NULL);

	gst_audio_encoder_set_output_format (GST_AUDIO_ENCODER (amrnbenc), copy);
	gst_caps_unref (copy);

	/* report needs to base class: hand one frame at a time */
	gst_audio_encoder_set_frame_samples_min (enc, 160);
	gst_audio_encoder_set_frame_samples_max (enc, 160);
	gst_audio_encoder_set_frame_max (enc, 1);

	return TRUE;
	}

	static GstFlowReturn
	gst_amrnbenc_handle_frame (GstAudioEncoder * enc, GstBuffer * buffer)
	{
	GstAmrnbEnc *amrnbenc;
	GstFlowReturn ret;
	GstBuffer *out;
	GstMapInfo in_map, out_map;
	gsize out_size;

	amrnbenc = GST_AMRNBENC (enc);

	g_return_val_if_fail (amrnbenc->handle, GST_FLOW_FLUSHING);

	/* we don't deal with squeezing remnants, so simply discard those */
	if (G_UNLIKELY (buffer == NULL)) {
	GST_DEBUG_OBJECT (amrnbenc, "no data");
	return GST_FLOW_OK;
	}

	gst_buffer_map (buffer, &in_map, GST_MAP_READ);

	if (G_UNLIKELY (in_map.size < 320)) {
	gst_buffer_unmap (buffer, &in_map);
	GST_DEBUG_OBJECT (amrnbenc, "discarding trailing data of %" G_GSIZE_FORMAT
	" bytes", in_map.size);
	return gst_audio_encoder_finish_frame (enc, NULL, -1);
	}

	/* get output, max size is 32 */
	out = gst_buffer_new_and_alloc (32);
	/* AMR encoder actually writes into the source data buffers it gets */
	/* should be able to handle that with what we are given */

	gst_buffer_map (out, &out_map, GST_MAP_WRITE);
	/* encode */
	out_size =
	Encoder_Interface_Encode (amrnbenc->handle, amrnbenc->bandmode,
	(short *) in_map.data, out_map.data, 0);
	gst_buffer_unmap (out, &out_map);
	gst_buffer_resize (out, 0, out_size);
	gst_buffer_unmap (buffer, &in_map);

	GST_LOG_OBJECT (amrnbenc, "output data size %" G_GSIZE_FORMAT, out_size);

	if (out_size) {
	ret = gst_audio_encoder_finish_frame (enc, out, 160);
	} else {
	/* should not happen (without dtx or so at least) */
	GST_WARNING_OBJECT (amrnbenc, "no encoded data; discarding input");
	gst_buffer_unref (out);
	ret = gst_audio_encoder_finish_frame (enc, NULL, -1);
	}

	return ret;
	}