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authorMark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>2012-01-26 23:28:07 +0100
committerMark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>2012-01-26 23:28:07 +0100
commitde606f64eb6075a28e2510d6b9497914907fbe93 (patch)
tree82522522f9b4c3714f0d6d233b2b8a98e9ea6079 /ext/gsm
parent2b5c6d67eef52628cfaf8783a21dd36ef48e90f1 (diff)
downloadgstreamer-plugins-bad-de606f64eb6075a28e2510d6b9497914907fbe93.tar.gz
gsm: port to 0.11
Diffstat (limited to 'ext/gsm')
-rw-r--r--ext/gsm/gstgsmdec.c58
-rw-r--r--ext/gsm/gstgsmenc.c47
2 files changed, 50 insertions, 55 deletions
diff --git a/ext/gsm/gstgsmdec.c b/ext/gsm/gstgsmdec.c
index 2bf475f26..502eb1742 100644
--- a/ext/gsm/gstgsmdec.c
+++ b/ext/gsm/gstgsmdec.c
@@ -67,20 +67,22 @@ static GstStaticPadTemplate gsmdec_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "endianness = (int) BYTE_ORDER, "
- "signed = (boolean) true, "
- "width = (int) 16, "
- "depth = (int) 16, " "rate = (int) [1, MAX], " "channels = (int) 1")
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_AUDIO_NE (S16) ", "
+ "layout = (string) interleaved, "
+ "rate = (int) [1, MAX], channels = (int) 1")
);
-GST_BOILERPLATE (GstGSMDec, gst_gsmdec, GstAudioDecoder,
- GST_TYPE_AUDIO_DECODER);
+G_DEFINE_TYPE (GstGSMDec, gst_gsmdec, GST_TYPE_AUDIO_DECODER);
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_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",
"Codec/Decoder/Audio",
"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->stop = GST_DEBUG_FUNCPTR (gst_gsmdec_stop);
@@ -108,7 +102,7 @@ gst_gsmdec_class_init (GstGSMDecClass * klass)
}
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);
/* Setting up src caps based on the input sample rate. */
- srccaps = gst_caps_new_simple ("audio/x-raw-int",
- "endianness", G_TYPE_INT, G_BYTE_ORDER,
- "signed", G_TYPE_BOOLEAN, TRUE,
- "width", G_TYPE_INT, 16,
- "depth", G_TYPE_INT, 16,
+ srccaps = gst_caps_new_simple ("audio/x-raw",
+ "format", G_TYPE_STRING, GST_AUDIO_NE (S16),
+ "layout", G_TYPE_STRING, "interleaved",
"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);
return ret;
@@ -208,7 +200,7 @@ gst_gsmdec_parse (GstAudioDecoder * dec, GstAdapter * adapter,
}
if (size < gsmdec->needed)
- return GST_FLOW_UNEXPECTED;
+ return GST_FLOW_EOS;
*offset = 0;
*length = gsmdec->needed;
@@ -223,6 +215,7 @@ gst_gsmdec_handle_frame (GstAudioDecoder * dec, GstBuffer * buffer)
gsm_byte *data;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
+ GstMapInfo map, omap;
/* no fancy draining */
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));
/* now encode frame into the output buffer */
- data = (gsm_byte *) GST_BUFFER_DATA (buffer);
- if (gsm_decode (gsmdec->state, data,
- (gsm_signal *) GST_BUFFER_DATA (outbuf)) < 0) {
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
+ gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
+ data = (gsm_byte *) map.data;
+ if (gsm_decode (gsmdec->state, data, (gsm_signal *) omap.data) < 0) {
/* invalid frame */
GST_AUDIO_DECODER_ERROR (gsmdec, 1, STREAM, DECODE, (NULL),
("tried to decode an invalid frame"), ret);
- if (ret != GST_FLOW_OK)
- goto exit;
+ gst_buffer_unmap (outbuf, &omap);
gst_buffer_unref (outbuf);
outbuf = NULL;
+ } else {
+ gst_buffer_unmap (outbuf, &omap);
}
+ gst_buffer_unmap (buffer, &map);
+
gst_audio_decoder_finish_frame (dec, outbuf, 1);
-exit:
return ret;
}
diff --git a/ext/gsm/gstgsmenc.c b/ext/gsm/gstgsmenc.c
index e8c97c1f0..3df26dc11 100644
--- a/ext/gsm/gstgsmenc.c
+++ b/ext/gsm/gstgsmenc.c
@@ -61,20 +61,22 @@ static GstStaticPadTemplate gsmenc_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "endianness = (int) BYTE_ORDER, "
- "signed = (boolean) true, "
- "width = (int) 16, "
- "depth = (int) 16, " "rate = (int) 8000, " "channels = (int) 1")
+ GST_STATIC_CAPS ("audio/x-raw, "
+ "format = (string) " GST_AUDIO_NE (S16) ", "
+ "layout = (string) interleaved, "
+ "rate = (int) 8000, channels = (int) 1")
);
-GST_BOILERPLATE (GstGSMEnc, gst_gsmenc, GstAudioEncoder,
- GST_TYPE_AUDIO_ENCODER);
+G_DEFINE_TYPE (GstGSMEnc, gst_gsmenc, GST_TYPE_AUDIO_ENCODER);
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_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",
"Codec/Encoder/Audio",
"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->stop = GST_DEBUG_FUNCPTR (gst_gsmenc_stop);
@@ -101,7 +95,7 @@ gst_gsmenc_class_init (GstGSMEncClass * klass)
}
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;
GstFlowReturn ret = GST_FLOW_OK;
GstBuffer *outbuf;
+ GstMapInfo map, omap;
gsmenc = GST_GSMENC (benc);
@@ -165,20 +160,24 @@ gst_gsmenc_handle_frame (GstAudioEncoder * benc, GstBuffer * buffer)
goto done;
}
- if (G_UNLIKELY (GST_BUFFER_SIZE (buffer) < 320)) {
- GST_DEBUG_OBJECT (gsmenc, "discarding trailing data %d",
- GST_BUFFER_SIZE (buffer));
+ gst_buffer_map (buffer, &map, GST_MAP_READ);
+ if (G_UNLIKELY (map.size < 320)) {
+ 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);
goto done;
}
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 */
- data = (gsm_signal *) GST_BUFFER_DATA (buffer);
- gsm_encode (gsmenc->state, data, (gsm_byte *) GST_BUFFER_DATA (outbuf));
+ data = (gsm_signal *) map.data;
+ 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);