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gsm: port to 0.11

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This commit is contained in:
Mark Nauwelaerts 2012-01-26 23:28:07 +01:00
parent 2b5c6d67ee
commit de606f64eb
3 changed files with 51 additions and 56 deletions
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2
configure.ac
View file

@ -325,7 +325,7 @@ GST_PLUGINS_NONPORTED=" adpcmdec adpcmenc aiff asfmux \
videomeasure videosignal vmnc \ videomeasure videosignal vmnc \
decklink fbdev linsys shm vcd \ decklink fbdev linsys shm vcd \
apexsink bz2 cdaudio celt cog curl dc1394 dirac directfb resindvd \ apexsink bz2 cdaudio celt cog curl dc1394 dirac directfb resindvd \
gsettings gsm jp2k ladspa modplug mimic \ gsettings jp2k ladspa modplug mimic \
musepack musicbrainz nas neon ofa openal opencv rsvg schro sdl smooth sndfile soundtouch spandsp timidity \ musepack musicbrainz nas neon ofa openal opencv rsvg schro sdl smooth sndfile soundtouch spandsp timidity \
wildmidi xvid apple_media lv2 teletextdec opus dvb" wildmidi xvid apple_media lv2 teletextdec opus dvb"
AC_SUBST(GST_PLUGINS_NONPORTED) AC_SUBST(GST_PLUGINS_NONPORTED)

58
ext/gsm/gstgsmdec.c
View file

@ -67,20 +67,22 @@ static GstStaticPadTemplate gsmdec_src_template =
GST_STATIC_PAD_TEMPLATE ("src", GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_SRC,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, " GST_STATIC_CAPS ("audio/x-raw, "
"endianness = (int) BYTE_ORDER, " "format = (string) " GST_AUDIO_NE (S16) ", "
"signed = (boolean) true, " "layout = (string) interleaved, "
"width = (int) 16, " "rate = (int) [1, MAX], channels = (int) 1")
"depth = (int) 16, " "rate = (int) [1, MAX], " "channels = (int) 1")
); );
GST_BOILERPLATE (GstGSMDec, gst_gsmdec, GstAudioDecoder, G_DEFINE_TYPE (GstGSMDec, gst_gsmdec, GST_TYPE_AUDIO_DECODER);
GST_TYPE_AUDIO_DECODER);
static void static void
gst_gsmdec_base_init (gpointer g_class) gst_gsmdec_class_init (GstGSMDecClass * klass)
{ {
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); GstElementClass *element_class;
GstAudioDecoderClass *base_class;
element_class = (GstElementClass *) klass;
base_class = (GstAudioDecoderClass *) klass;
gst_element_class_add_pad_template (element_class, gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gsmdec_sink_template)); gst_static_pad_template_get (&gsmdec_sink_template));
@ -89,14 +91,6 @@ gst_gsmdec_base_init (gpointer g_class)
gst_element_class_set_details_simple (element_class, "GSM audio decoder", gst_element_class_set_details_simple (element_class, "GSM audio decoder",
"Codec/Decoder/Audio", "Codec/Decoder/Audio",
"Decodes GSM encoded audio", "Philippe Khalaf <burger@speedy.org>"); "Decodes GSM encoded audio", "Philippe Khalaf <burger@speedy.org>");
}
static void
gst_gsmdec_class_init (GstGSMDecClass * klass)
{
GstAudioDecoderClass *base_class;
base_class = (GstAudioDecoderClass *) klass;
base_class->start = GST_DEBUG_FUNCPTR (gst_gsmdec_start); base_class->start = GST_DEBUG_FUNCPTR (gst_gsmdec_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmdec_stop); base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmdec_stop);
@ -108,7 +102,7 @@ gst_gsmdec_class_init (GstGSMDecClass * klass)
} }
static void static void
gst_gsmdec_init (GstGSMDec * gsmdec, GstGSMDecClass * klass) gst_gsmdec_init (GstGSMDec * gsmdec)
{ {
} }
@ -170,14 +164,12 @@ gst_gsmdec_set_format (GstAudioDecoder * dec, GstCaps * caps)
gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49); gsm_option (gsmdec->state, GSM_OPT_WAV49, &gsmdec->use_wav49);
/* Setting up src caps based on the input sample rate. */ /* Setting up src caps based on the input sample rate. */
srccaps = gst_caps_new_simple ("audio/x-raw-int", srccaps = gst_caps_new_simple ("audio/x-raw",
"endianness", G_TYPE_INT, G_BYTE_ORDER, "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
"signed", G_TYPE_BOOLEAN, TRUE, "layout", G_TYPE_STRING, "interleaved",
"width", G_TYPE_INT, 16,
"depth", G_TYPE_INT, 16,
"rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, 1, NULL); "rate", G_TYPE_INT, rate, "channels", G_TYPE_INT, 1, NULL);
ret = gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), srccaps); ret = gst_audio_decoder_set_outcaps (dec, srccaps);
gst_caps_unref (srccaps); gst_caps_unref (srccaps);
return ret; return ret;
@ -208,7 +200,7 @@ gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
} }
if (size < gsmdec->needed) if (size < gsmdec->needed)
return GST_FLOW_UNEXPECTED; return GST_FLOW_EOS;
*offset = 0; *offset = 0;
*length = gsmdec->needed; *length = gsmdec->needed;
@ -223,6 +215,7 @@ gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
gsm_byte *data; gsm_byte *data;
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf; GstBuffer *outbuf;
GstMapInfo map, omap;
/* no fancy draining */ /* no fancy draining */
if (G_UNLIKELY (!buffer)) if (G_UNLIKELY (!buffer))
@ -234,20 +227,23 @@ gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal)); outbuf = gst_buffer_new_and_alloc (ENCODED_SAMPLES * sizeof (gsm_signal));
/* now encode frame into the output buffer */ /* now encode frame into the output buffer */
data = (gsm_byte *) GST_BUFFER_DATA (buffer); gst_buffer_map (buffer, &map, GST_MAP_READ);
if (gsm_decode (gsmdec->state, data, gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
(gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) { data = (gsm_byte *) map.data;
if (gsm_decode (gsmdec->state, data, (gsm_signal *) omap.data) < 0) {
/* invalid frame */ /* invalid frame */
GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL), GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL),
("tried to decode an invalid frame"), ret); ("tried to decode an invalid frame"), ret);
if (ret != GST_FLOW_OK) gst_buffer_unmap (outbuf, &omap);
goto exit;
gst_buffer_unref (outbuf); gst_buffer_unref (outbuf);
outbuf = NULL; outbuf = NULL;
} else {
gst_buffer_unmap (outbuf, &omap);
} }
gst_buffer_unmap (buffer, &map);
gst_audio_decoder_finish_frame (dec, outbuf, 1); gst_audio_decoder_finish_frame (dec, outbuf, 1);
exit:
return ret; return ret;
} }

47
ext/gsm/gstgsmenc.c
View file

@ -61,20 +61,22 @@ static GstStaticPadTemplate gsmenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink", GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_SINK,
GST_PAD_ALWAYS, GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw-int, " GST_STATIC_CAPS ("audio/x-raw, "
"endianness = (int) BYTE_ORDER, " "format = (string) " GST_AUDIO_NE (S16) ", "
"signed = (boolean) true, " "layout = (string) interleaved, "
"width = (int) 16, " "rate = (int) 8000, channels = (int) 1")
"depth = (int) 16, " "rate = (int) 8000, " "channels = (int) 1")
); );
GST_BOILERPLATE (GstGSMEnc, gst_gsmenc, GstAudioEncoder, G_DEFINE_TYPE (GstGSMEnc, gst_gsmenc, GST_TYPE_AUDIO_ENCODER);
GST_TYPE_AUDIO_ENCODER);
static void static void
gst_gsmenc_base_init (gpointer g_class) gst_gsmenc_class_init (GstGSMEncClass * klass)
{ {
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); GstElementClass *element_class;
GstAudioEncoderClass *base_class;
element_class = (GstElementClass *) klass;
base_class = (GstAudioEncoderClass *) klass;
gst_element_class_add_pad_template (element_class, gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&gsmenc_sink_template)); gst_static_pad_template_get (&gsmenc_sink_template));
@ -83,14 +85,6 @@ gst_gsmenc_base_init (gpointer g_class)
gst_element_class_set_details_simple (element_class, "GSM audio encoder", gst_element_class_set_details_simple (element_class, "GSM audio encoder",
"Codec/Encoder/Audio", "Codec/Encoder/Audio",
"Encodes GSM audio", "Philippe Khalaf <burger@speedy.org>"); "Encodes GSM audio", "Philippe Khalaf <burger@speedy.org>");
}
static void
gst_gsmenc_class_init (GstGSMEncClass * klass)
{
GstAudioEncoderClass *base_class;
base_class = (GstAudioEncoderClass *) klass;
base_class->start = GST_DEBUG_FUNCPTR (gst_gsmenc_start); base_class->start = GST_DEBUG_FUNCPTR (gst_gsmenc_start);
base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmenc_stop); base_class->stop = GST_DEBUG_FUNCPTR (gst_gsmenc_stop);
@ -101,7 +95,7 @@ gst_gsmenc_class_init (GstGSMEncClass * klass)
} }
static void static void
gst_gsmenc_init (GstGSMEnc * gsmenc, GstGSMEncClass * klass) gst_gsmenc_init (GstGSMEnc * gsmenc)
{ {
} }
@ -156,6 +150,7 @@ gst_gsmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
gsm_signal *data; gsm_signal *data;
GstFlowReturn ret = GST_FLOW_OK; GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf; GstBuffer *outbuf;
GstMapInfo map, omap;
gsmenc = GST_GSMENC (benc); gsmenc = GST_GSMENC (benc);
@ -165,20 +160,24 @@ gst_gsmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
goto done; goto done;
} }
if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < 320)) { gst_buffer_map (buffer, &map, GST_MAP_READ);
GST_DEBUG_OBJECT (gsmenc, "discarding trailing data %d", if (G_UNLIKELY (map.size < 320)) {
GST_BUFFER_SIZE (buffer)); GST_DEBUG_OBJECT (gsmenc, "discarding trailing data %d", (gint) map.size);
gst_buffer_unmap (buffer, &map);
ret = gst_audio_encoder_finish_frame (benc, NULL, -1); ret = gst_audio_encoder_finish_frame (benc, NULL, -1);
goto done; goto done;
} }
outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte)); outbuf = gst_buffer_new_and_alloc (33 * sizeof (gsm_byte));
gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
/* encode 160 16-bit samples into 33 bytes */ /* encode 160 16-bit samples into 33 bytes */
data = (gsm_signal *) GST_BUFFER_DATA (buffer); data = (gsm_signal *) map.data;
gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf)); gsm_encode (gsmenc->state, data, (gsm_byte *) omap.data);
GST_LOG_OBJECT (gsmenc, "encoded to %d bytes", GST_BUFFER_SIZE (outbuf)); GST_LOG_OBJECT (gsmenc, "encoded to %d bytes", (gint) omap.size);
gst_buffer_unmap (buffer, &map);
gst_buffer_unmap (buffer, &omap);
ret = gst_audio_encoder_finish_frame (benc, outbuf, 160); ret = gst_audio_encoder_finish_frame (benc, outbuf, 160);