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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-07-05 06:05:53 +00:00
gsmenc: port to audioencoder
This commit is contained in:
Mark Nauwelaerts
parent
6b4038fed3
commit
0acfa3cc1e
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@ -43,39 +43,12 @@ enum
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ARG_0
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};
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static void gst_gsmenc_base_init (gpointer g_class);
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static void gst_gsmenc_class_init (GstGSMEnc * klass);
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static void gst_gsmenc_init (GstGSMEnc * gsmenc);
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static void gst_gsmenc_finalize (GObject * object);
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static gboolean gst_gsmenc_setcaps (GstPad * pad, GstCaps * caps);
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static GstFlowReturn gst_gsmenc_chain (GstPad * pad, GstBuffer * buf);
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static GstElementClass *parent_class = NULL;
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GType
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gst_gsmenc_get_type (void)
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{
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static GType gsmenc_type = 0;
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if (!gsmenc_type) {
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static const GTypeInfo gsmenc_info = {
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sizeof (GstGSMEncClass),
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gst_gsmenc_base_init,
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NULL,
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(GClassInitFunc) gst_gsmenc_class_init,
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NULL,
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NULL,
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sizeof (GstGSMEnc),
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0,
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(GInstanceInitFunc) gst_gsmenc_init,
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};
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gsmenc_type =
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g_type_register_static (GST_TYPE_ELEMENT, "GstGSMEnc", &gsmenc_info, 0);
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}
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return gsmenc_type;
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}
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static gboolean gst_gsmenc_start (GstAudioEncoder * enc);
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static gboolean gst_gsmenc_stop (GstAudioEncoder * enc);
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static gboolean gst_gsmenc_set_format (GstAudioEncoder * enc,
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GstAudioInfo * info);
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static GstFlowReturn gst_gsmenc_handle_frame (GstAudioEncoder * enc,
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GstBuffer * in_buf);
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static GstStaticPadTemplate gsmenc_src_template =
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GST_STATIC_PAD_TEMPLATE ("src",
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@ -95,6 +68,9 @@ GST_STATIC_PAD_TEMPLATE ("sink",
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"depth = (int) 16, " "rate = (int) 8000, " "channels = (int) 1")
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);
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GST_BOILERPLATE (GstGSMEnc, gst_gsmenc, GstAudioEncoder,
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GST_TYPE_AUDIO_ENCODER);
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static void
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gst_gsmenc_base_init (gpointer g_class)
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{
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@ -110,34 +86,32 @@ gst_gsmenc_base_init (gpointer g_class)
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}
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static void
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gst_gsmenc_class_init (GstGSMEnc * klass)
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gst_gsmenc_class_init (GstGSMEncClass * klass)
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{
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GObjectClass *gobject_class;
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GstAudioEncoderClass *base_class;
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gobject_class = (GObjectClass *) klass;
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base_class = (GstAudioEncoderClass *) klass;
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parent_class = g_type_class_peek_parent (klass);
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gobject_class->finalize = gst_gsmenc_finalize;
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base_class->start = GST_DEBUG_FUNCPTR (gst_gsmenc_start);
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base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmenc_stop);
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base_class->set_format = GST_DEBUG_FUNCPTR (gst_gsmenc_set_format);
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base_class->handle_frame = GST_DEBUG_FUNCPTR (gst_gsmenc_handle_frame);
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GST_DEBUG_CATEGORY_INIT (gsmenc_debug, "gsmenc", 0, "GSM Encoder");
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}
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static void
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gst_gsmenc_init (GstGSMEnc * gsmenc)
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gst_gsmenc_init (GstGSMEnc * gsmenc, GstGSMEncClass * klass)
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{
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}
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static gboolean
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gst_gsmenc_start (GstAudioEncoder * enc)
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{
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GstGSMEnc *gsmenc = GST_GSMENC (enc);
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gint use_wav49;
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/* create the sink and src pads */
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gsmenc->sinkpad =
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gst_pad_new_from_static_template (&gsmenc_sink_template, "sink");
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gst_pad_set_chain_function (gsmenc->sinkpad, gst_gsmenc_chain);
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gst_pad_set_setcaps_function (gsmenc->sinkpad, gst_gsmenc_setcaps);
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gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->sinkpad);
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gsmenc->srcpad =
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gst_pad_new_from_static_template (&gsmenc_src_template, "src");
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gst_element_add_pad (GST_ELEMENT (gsmenc), gsmenc->srcpad);
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GST_DEBUG_OBJECT (enc, "start");
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gsmenc->state = gsm_create ();
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@ -145,78 +119,72 @@ gst_gsmenc_init (GstGSMEnc * gsmenc)
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use_wav49 = 0;
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gsm_option (gsmenc->state, GSM_OPT_WAV49, &use_wav49);
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gsmenc->adapter = gst_adapter_new ();
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gsmenc->next_ts = 0;
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}
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static void
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gst_gsmenc_finalize (GObject * object)
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{
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GstGSMEnc *gsmenc;
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gsmenc = GST_GSMENC (object);
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g_object_unref (gsmenc->adapter);
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gsm_destroy (gsmenc->state);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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return TRUE;
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}
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static gboolean
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gst_gsmenc_setcaps (GstPad * pad, GstCaps * caps)
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gst_gsmenc_stop (GstAudioEncoder * enc)
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{
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GstGSMEnc *gsmenc;
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GstCaps *srccaps;
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GstGSMEnc *gsmenc = GST_GSMENC (enc);
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gsmenc = GST_GSMENC (gst_pad_get_parent (pad));
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srccaps = gst_static_pad_template_get_caps (&gsmenc_src_template);
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gst_pad_set_caps (gsmenc->srcpad, srccaps);
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gst_object_unref (gsmenc);
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GST_DEBUG_OBJECT (enc, "stop");
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gsm_destroy (gsmenc->state);
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return TRUE;
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}
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static gboolean
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gst_gsmenc_set_format (GstAudioEncoder * benc, GstAudioInfo * info)
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{
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GstGSMEnc *gsmenc;
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GstCaps *srccaps;
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gsmenc = GST_GSMENC (benc);
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srccaps = gst_static_pad_template_get_caps (&gsmenc_src_template);
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gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (benc), srccaps);
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/* report needs to base class */
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gst_audio_encoder_set_frame_samples_min (benc, 160);
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gst_audio_encoder_set_frame_samples_max (benc, 160);
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gst_audio_encoder_set_frame_max (benc, 1);
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return TRUE;
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}
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static GstFlowReturn
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gst_gsmenc_chain (GstPad * pad, GstBuffer * buf)
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gst_gsmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
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{
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GstGSMEnc *gsmenc;
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gsm_signal *data;
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GstFlowReturn ret = GST_FLOW_OK;
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GstBuffer *outbuf;
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gsmenc = GST_GSMENC (gst_pad_get_parent (pad));
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gsmenc = GST_GSMENC (benc);
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if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DISCONT)) {
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gst_adapter_clear (gsmenc->adapter);
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}
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gst_adapter_push (gsmenc->adapter, buf);
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while (gst_adapter_available (gsmenc->adapter) >= 320) {
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GstBuffer *outbuf;
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outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
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GST_BUFFER_TIMESTAMP (outbuf) = gsmenc->next_ts;
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GST_BUFFER_DURATION (outbuf) = 20 * GST_MSECOND;
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gsmenc->next_ts += 20 * GST_MSECOND;
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/* encode 160 16-bit samples into 33 bytes */
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data = (gsm_signal *) gst_adapter_peek (gsmenc->adapter, 320);
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gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf));
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gst_adapter_flush (gsmenc->adapter, 320);
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gst_buffer_set_caps (outbuf, GST_PAD_CAPS (gsmenc->srcpad));
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GST_DEBUG_OBJECT (gsmenc, "Pushing buffer of size %d",
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GST_BUFFER_SIZE (outbuf));
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ret = gst_pad_push (gsmenc->srcpad, outbuf);
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/* we don't deal with squeezing remnants, so simply discard those */
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if (G_UNLIKELY (buffer == NULL)) {
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GST_DEBUG_OBJECT (gsmenc, "no data");
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goto done;
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}
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gst_object_unref (gsmenc);
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if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < 320)) {
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GST_DEBUG_OBJECT (gsmenc, "discarding trailing data %d",
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GST_BUFFER_SIZE (buffer));
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ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
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goto done;
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}
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outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
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/* encode 160 16-bit samples into 33 bytes */
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data = (gsm_signal *) GST_BUFFER_DATA (buffer);
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gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf));
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GST_LOG_OBJECT (gsmenc, "encoded to %d bytes", GST_BUFFER_SIZE (outbuf));
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ret = gst_audio_encoder_finish_frame (benc, outbuf, 160);
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done:
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return ret;
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}
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@ -21,7 +21,7 @@
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#define __GST_GSMENC_H__
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#include <gst/gst.h>
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#include <gst/base/gstadapter.h>
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#include <gst/audio/gstaudioencoder.h>
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#ifdef GSM_HEADER_IN_SUBDIR
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#include <gsm/gsm.h>
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@ -47,20 +47,14 @@ typedef struct _GstGSMEncClass GstGSMEncClass;
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struct _GstGSMEnc
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{
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GstElement element;
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/* pads */
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GstPad *sinkpad, *srcpad;
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GstAdapter *adapter;
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GstAudioEncoder element;
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gsm state;
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GstClockTime next_ts;
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gboolean firstBuf;
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};
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struct _GstGSMEncClass
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{
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GstElementClass parent_class;
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GstAudioEncoderClass parent_class;
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};
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GType gst_gsmenc_get_type (void);
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