/* copy of the same structure in the vorbis plugin */
const GstAudioChannelPosition gst_opus_channel_positions[][8] = {
{ /* Mono */
- GST_AUDIO_CHANNEL_POSITION_FRONT_MONO},
+ GST_AUDIO_CHANNEL_POSITION_MONO},
{ /* Stereo */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT},
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
- GST_AUDIO_CHANNEL_POSITION_LFE,
+ GST_AUDIO_CHANNEL_POSITION_LFE1,
},
{ /* 6.1 Surround, in Vorbis spec since 2010-01-13 */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
- GST_AUDIO_CHANNEL_POSITION_LFE},
+ GST_AUDIO_CHANNEL_POSITION_LFE1},
{ /* 7.1 Surround, in Vorbis spec since 2010-01-13 */
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
- GST_AUDIO_CHANNEL_POSITION_LFE},
+ GST_AUDIO_CHANNEL_POSITION_LFE1},
};
const char *gst_opus_channel_names[] = {
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) { " GST_AUDIO_NE (S16) " }, "
+ "layout = (string) interleaved, "
"rate = (int) { 48000, 24000, 16000, 12000, 8000 }, "
"channels = (int) [ 1, 8 ] ")
);
return DB_TO_LINEAR (gst_opus_dec_get_r128_gain (r128_gain));
}
-static GstCaps *
-gst_opus_dec_negotiate (GstOpusDec * dec)
+static void
+gst_opus_dec_negotiate (GstOpusDec * dec, const GstAudioChannelPosition * pos)
{
GstCaps *caps = gst_pad_get_allowed_caps (GST_AUDIO_DECODER_SRC_PAD (dec));
GstStructure *s;
+ GstAudioInfo info;
caps = gst_caps_make_writable (caps);
gst_caps_truncate (caps);
GST_INFO_OBJECT (dec, "Negotiated %d channels, %d Hz", dec->n_channels,
dec->sample_rate);
- return caps;
+ /* pass valid order to audio info */
+ if (pos) {
+ memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
+ gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
+ }
+
+ /* set up source format */
+ gst_audio_info_init (&info);
+ gst_audio_info_set_format (&info, GST_AUDIO_FORMAT_S16,
+ dec->sample_rate, dec->n_channels, pos ? dec->opus_pos : NULL);
+ gst_audio_decoder_set_output_format (GST_AUDIO_DECODER (dec), &info);
+
+ /* but we still need the opus order for later reordering */
+ if (pos) {
+ memcpy (dec->opus_pos, pos, sizeof (pos[0]) * dec->n_channels);
+ gst_audio_channel_positions_to_valid_order (dec->opus_pos, dec->n_channels);
+ } else {
+ dec->opus_pos[0] = GST_AUDIO_CHANNEL_POSITION_INVALID;
+ }
+
+ dec->info = info;
}
static GstFlowReturn
gst_opus_dec_parse_header (GstOpusDec * dec, GstBuffer * buf)
{
const guint8 *data;
- GstCaps *caps;
- const GstAudioChannelPosition *pos = NULL;
+ GstAudioChannelPosition pos[64];
+ const GstAudioChannelPosition *posn = NULL;
+ GstMapInfo map;
g_return_val_if_fail (gst_opus_header_is_id_header (buf), GST_FLOW_ERROR);
- data = gst_buffer_map (buf, NULL, NULL, GST_MAP_READ);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ data = map.data;
g_return_val_if_fail (dec->n_channels == 0
|| dec->n_channels == data[9], GST_FLOW_ERROR);
case 6:
case 7:
case 8:
- pos = gst_opus_channel_positions[dec->n_channels - 1];
+ posn = gst_opus_channel_positions[dec->n_channels - 1];
break;
default:{
gint i;
- GstAudioChannelPosition *posn =
- g_new (GstAudioChannelPosition, dec->n_channels);
GST_ELEMENT_WARNING (GST_ELEMENT (dec), STREAM, DECODE,
(NULL), ("Using NONE channel layout for more than 8 channels"));
for (i = 0; i < dec->n_channels; i++)
- posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
+ pos[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
- pos = posn;
+ posn = pos;
}
}
} else {
}
}
- caps = gst_opus_dec_negotiate (dec);
-
- if (pos) {
- GST_DEBUG_OBJECT (dec, "Setting channel positions on caps");
- gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
- }
-
- if (dec->n_channels > 8) {
- g_free ((GstAudioChannelPosition *) pos);
- }
-
- GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
- gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
- gst_caps_unref (caps);
+ gst_opus_dec_negotiate (dec, posn);
- gst_buffer_unmap (buf, (guint8 *) data, -1);
+ gst_buffer_unmap (buf, &map);
return GST_FLOW_OK;
}
opus_dec_chain_parse_data (GstOpusDec * dec, GstBuffer * buffer)
{
GstFlowReturn res = GST_FLOW_OK;
- gsize size, out_size;
+ gsize size;
guint8 *data;
GstBuffer *outbuf;
gint16 *out_data;
int samples;
unsigned int packet_size;
GstBuffer *buf;
+ GstMapInfo map, omap;
if (dec->state == NULL) {
/* If we did not get any headers, default to 2 channels */
if (dec->n_channels == 0) {
- GstCaps *caps;
GST_INFO_OBJECT (dec, "No header, assuming single stream");
dec->n_channels = 2;
dec->sample_rate = 48000;
- caps = gst_opus_dec_negotiate (dec);
- GST_INFO_OBJECT (dec, "Setting src caps to %" GST_PTR_FORMAT, caps);
- gst_pad_set_caps (GST_AUDIO_DECODER_SRC_PAD (dec), caps);
- gst_caps_unref (caps);
/* default stereo mapping */
dec->channel_mapping_family = 0;
dec->channel_mapping[0] = 0;
dec->channel_mapping[1] = 1;
dec->n_streams = 1;
dec->n_stereo_streams = 1;
+
+ gst_opus_dec_negotiate (dec, NULL);
}
GST_DEBUG_OBJECT (dec, "Creating decoder with %d channels, %d Hz",
buf = dec->use_inband_fec && dec->last_buffer ? dec->last_buffer : buffer;
if (buf) {
- data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ data = map.data;
+ size = map.size;
GST_DEBUG_OBJECT (dec, "Using buffer of size %u", size);
} else {
goto buffer_failed;
}
- out_data = (gint16 *) gst_buffer_map (outbuf, &out_size, NULL, GST_MAP_WRITE);
+ gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
+ out_data = (gint16 *) omap.data;
if (dec->use_inband_fec) {
if (dec->last_buffer) {
/* normal decode */
n = opus_multistream_decode (dec->state, data, size, out_data, samples, 0);
}
- gst_buffer_unmap (buf, data, size);
- gst_buffer_unmap (outbuf, out_data, out_size);
+ gst_buffer_unmap (outbuf, &omap);
+ if (buf)
+ gst_buffer_unmap (buf, &map);
if (n < 0) {
GST_ELEMENT_ERROR (dec, STREAM, DECODE, ("Decoding error: %d", n), (NULL));
if (gst_buffer_get_size (outbuf) == 0) {
gst_buffer_unref (outbuf);
outbuf = NULL;
+ } else if (dec->opus_pos[0] != GST_AUDIO_CHANNEL_POSITION_INVALID) {
+ gst_audio_buffer_reorder_channels (outbuf, GST_AUDIO_FORMAT_S16,
+ dec->n_channels, dec->opus_pos, dec->info.position);
}
/* Apply gain */
gsize rsize;
unsigned int i, nsamples;
double volume = dec->r128_gain_volume;
- gint16 *samples =
- (gint16 *) gst_buffer_map (outbuf, &rsize, NULL, GST_MAP_READWRITE);
+ gint16 *samples;
+ gst_buffer_map (outbuf, &omap, GST_MAP_READWRITE);
+ samples = (gint16 *) omap.data;
+ rsize = omap.size;
GST_DEBUG_OBJECT (dec, "Applying gain: volume %f", volume);
nsamples = rsize / 2;
for (i = 0; i < nsamples; ++i) {
int sample = (int) (samples[i] * volume + 0.5);
samples[i] = sample < -32768 ? -32768 : sample > 32767 ? 32767 : sample;
}
- gst_buffer_unmap (outbuf, samples, rsize);
+ gst_buffer_unmap (outbuf, &omap);
}
res = gst_audio_decoder_finish_frame (GST_AUDIO_DECODER (dec), outbuf, 1);
memcmp_buffers (GstBuffer * buf1, GstBuffer * buf2)
{
gsize size1, size2;
- gpointer data1;
gboolean res;
+ GstMapInfo map;
size1 = gst_buffer_get_size (buf1);
size2 = gst_buffer_get_size (buf2);
if (size1 != size2)
return FALSE;
- data1 = gst_buffer_map (buf1, NULL, NULL, GST_MAP_READ);
- res = gst_buffer_memcmp (buf2, 0, data1, size1) == 0;
- gst_buffer_unmap (buf1, data1, size1);
+ gst_buffer_map (buf1, &map, GST_MAP_READ);
+ res = gst_buffer_memcmp (buf2, 0, map.data, map.size) == 0;
+ gst_buffer_unmap (buf1, &map);
return res;
}
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) " FORMAT_STR ", "
+ "layout = (string) interleaved, "
"rate = (int) { 8000, 12000, 16000, 24000, 48000 }, "
"channels = (int) [ 1, 2 ] ")
);
GST_DEBUG_OBJECT (enc, "sink getcaps");
- peercaps = gst_pad_peer_query_caps (GST_AUDIO_ENCODER_SRC_PAD (benc), filter);
+ peercaps = gst_pad_peer_query_caps (GST_AUDIO_ENCODER_SRC_PAD (benc), NULL);
if (!peercaps) {
GST_DEBUG_OBJECT (benc, "No peercaps, returning template sink caps");
return
gsize bsize, size;
gsize bytes = enc->frame_samples * enc->n_channels * 2;
gint ret = GST_FLOW_OK;
+ GstMapInfo map;
g_mutex_lock (enc->property_lock);
if (G_LIKELY (buf)) {
- bdata = gst_buffer_map (buf, &bsize, NULL, GST_MAP_READ);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ bdata = map.data;
+ bsize = map.size;
if (G_UNLIKELY (bsize % bytes)) {
GST_DEBUG_OBJECT (enc, "draining; adding silence samples");
size = ((bsize / bytes) + 1) * bytes;
mdata = g_malloc0 (size);
memcpy (mdata, bdata, bsize);
- gst_buffer_unmap (buf, bdata, bsize);
- bdata = NULL;
data = mdata;
} else {
data = bdata;
while (size) {
gint encoded_size;
- unsigned char *out_data;
- gsize out_size;
GstBuffer *outbuf;
+ GstMapInfo omap;
outbuf = gst_buffer_new_and_alloc (enc->max_payload_size * enc->n_channels);
if (!outbuf)
GST_DEBUG_OBJECT (enc, "encoding %d samples (%d bytes)",
enc->frame_samples, (int) bytes);
- out_data = gst_buffer_map (outbuf, &out_size, NULL, GST_MAP_WRITE);
+ gst_buffer_map (outbuf, &omap, GST_MAP_WRITE);
encoded_size =
opus_multistream_encode (enc->state, (const gint16 *) data,
- enc->frame_samples, out_data, enc->max_payload_size * enc->n_channels);
- gst_buffer_unmap (outbuf, out_data, out_size);
+ enc->frame_samples, omap.data, enc->max_payload_size * enc->n_channels);
+ gst_buffer_unmap (outbuf, &omap);
if (encoded_size < 0) {
GST_ERROR_OBJECT (enc, "Encoding failed: %d", encoded_size);
} else if (encoded_size > enc->max_payload_size) {
GST_WARNING_OBJECT (enc,
"Encoded size %d is higher than max payload size (%d bytes)",
- out_size, enc->max_payload_size);
+ encoded_size, enc->max_payload_size);
ret = GST_FLOW_ERROR;
goto done;
}
done:
if (bdata)
- gst_buffer_unmap (buf, bdata, bsize);
+ gst_buffer_unmap (buf, &map);
g_mutex_unlock (enc->property_lock);
if (mdata)
/* negotiate with these caps */
GST_DEBUG_OBJECT (enc, "here are the caps: %" GST_PTR_FORMAT, caps);
- gst_pad_set_caps (GST_AUDIO_ENCODER_SRC_PAD (enc), caps);
+ gst_audio_encoder_set_output_format (benc, caps);
gst_caps_unref (caps);
enc->header_sent = TRUE;
g_assert (gst_buffer_is_writable (buf));
/* mark buffer */
- GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
g_value_init (&value, GST_TYPE_BUFFER);
buf = gst_buffer_copy (buf);
- GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
+ GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_HEADER);
gst_value_set_buffer (&value, buf);
gst_buffer_unref (buf);
gst_value_array_append_value (&array, &value);
{
int n_streams, family;
gboolean multistream;
+ GstMapInfo map;
guint8 *data;
- gsize size;
g_return_if_fail (caps);
g_return_if_fail (headers && !*headers);
g_return_if_fail (gst_buffer_get_size (buf1) >= 19);
- data = gst_buffer_map (buf1, &size, NULL, GST_MAP_READ);
+ gst_buffer_map (buf1, &map, GST_MAP_READ);
+ data = map.data;
/* work out the number of streams */
family = data[18];
n_streams = 1;
} else {
/* only included in the header for family > 0 */
- if (size >= 20)
+ if (map.size >= 20)
n_streams = data[19];
else {
g_warning ("family > 0 but header buffer size < 20");
- gst_buffer_unmap (buf1, data, size);
+ gst_buffer_unmap (buf1, &map);
return;
}
}
- gst_buffer_unmap (buf1, data, size);
+ gst_buffer_unmap (buf1, &map);
/* mark and put on caps */
multistream = n_streams > 1;
guint8 *data = NULL;
guint8 channels, channel_mapping_family, n_streams, n_stereo_streams;
gboolean ret = FALSE;
+ GstMapInfo map;
if (size < 19)
goto beach;
if (!gst_opus_header_is_header (buf, "OpusHead", 8))
goto beach;
- data = gst_buffer_map (buf, &size, NULL, GST_MAP_READ);
+ gst_buffer_map (buf, &map, GST_MAP_READ);
+ data = map.data;
+ size = map.size;
channels = data[9];
beach:
if (data)
- gst_buffer_unmap (buf, data, size);
+ gst_buffer_unmap (buf, &map);
return ret;
}
projects / platform / upstream / gst-plugins-base.git / commitdiff