mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-07-05 22:25:55 +00:00
Code Issues 1 Projects Releases Wiki Activity

rtp: Update for the new audio caps

Browse source
This commit is contained in:
Sebastian Dröge 2012-01-04 13:48:36 +01:00
parent 19788be6b1
commit 4885f34458
9 changed files with 122 additions and 80 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

42
gst/rtp/gstrtpL16depay.c
View file

@ -25,7 +25,6 @@
#include <stdlib.h>
#include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
#include "gstrtpL16depay.h"
#include "gstrtpchannels.h"
@ -39,6 +38,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) S16_BE, "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
@ -131,6 +131,7 @@ gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
gboolean res;
const gchar *channel_order;
const GstRTPChannelOrder *order;
GstAudioInfo *info;
rtpL16depay = GST_RTP_L16_DEPAY (depayload);
@ -171,32 +172,32 @@ gst_rtp_L16_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
}
depayload->clock_rate = clock_rate;
rtpL16depay->rate = clock_rate;
rtpL16depay->channels = channels;
srccaps = gst_caps_new_simple ("audio/x-raw",
"format", G_TYPE_STRING, "S16_BE",
"rate", G_TYPE_INT, clock_rate, "channels", G_TYPE_INT, channels, NULL);
info = &rtpL16depay->info;
gst_audio_info_init (info);
info->finfo = gst_audio_format_get_info (GST_AUDIO_FORMAT_S16BE);
info->rate = clock_rate;
info->channels = channels;
info->bpf = (info->finfo->width / 8) * channels;
/* add channel positions */
channel_order = gst_structure_get_string (structure, "channel-order");
order = gst_rtp_channels_get_by_order (channels, channel_order);
rtpL16depay->order = order;
if (order) {
gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0),
order->pos);
memcpy (info->position, order->pos,
sizeof (GstAudioChannelPosition) * channels);
gst_audio_channel_positions_to_valid_order (info->position, info->channels);
} else {
GstAudioChannelPosition *pos;
GST_ELEMENT_WARNING (rtpL16depay, STREAM, DECODE,
(NULL), ("Unknown channel order '%s' for %d channels",
GST_STR_NULL (channel_order), channels));
/* create default NONE layout */
pos = gst_rtp_channels_create_default (channels);
gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
g_free (pos);
gst_rtp_channels_create_default (channels, info->position);
}
srccaps = gst_audio_info_to_caps (info);
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);
@ -237,6 +238,14 @@ gst_rtp_L16_depay_process (GstRTPBaseDepayload * depayload, GstBuffer * buf)
GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT);
}
outbuf = gst_buffer_make_writable (outbuf);
if (rtpL16depay->order &&
!gst_audio_buffer_reorder_channels (outbuf,
rtpL16depay->info.finfo->format, rtpL16depay->info.channels,
rtpL16depay->info.position, rtpL16depay->order->pos)) {
goto reorder_failed;
}
gst_rtp_buffer_unmap (&rtp);
return outbuf;
@ -249,6 +258,13 @@ empty_packet:
gst_rtp_buffer_unmap (&rtp);
return NULL;
}
reorder_failed:
{
GST_ELEMENT_ERROR (rtpL16depay, STREAM, DECODE,
("Channel reordering failed."), (NULL));
gst_rtp_buffer_unmap (&rtp);
return NULL;
}
}
gboolean

7
gst/rtp/gstrtpL16depay.h
View file

@ -22,6 +22,9 @@
#include <gst/gst.h>
#include <gst/rtp/gstrtpbasedepayload.h>
#include <gst/audio/audio.h>
#include "gstrtpchannels.h"
G_BEGIN_DECLS
@ -45,8 +48,8 @@ struct _GstRtpL16Depay
{
GstRTPBaseDepayload depayload;
guint rate;
guint channels;
GstAudioInfo info;
const GstRTPChannelOrder *order;
};
/* Standard definition defining a class for this element. */

78
gst/rtp/gstrtpL16pay.c
View file

@ -24,7 +24,6 @@
#include <string.h>
#include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpL16pay.h"
@ -39,6 +38,7 @@ GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("audio/x-raw, "
"format = (string) S16_BE, "
"layout = (string) interleaved, "
"rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
);
@ -68,6 +68,9 @@ static gboolean gst_rtp_L16_pay_setcaps (GstRTPBasePayload * basepayload,
GstCaps * caps);
static GstCaps *gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload,
GstPad * pad, GstCaps * filter);
static GstFlowReturn
gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer);
#define gst_rtp_L16_pay_parent_class parent_class
G_DEFINE_TYPE (GstRtpL16Pay, gst_rtp_L16_pay, GST_TYPE_RTP_BASE_AUDIO_PAYLOAD);
@ -83,6 +86,7 @@ gst_rtp_L16_pay_class_init (GstRtpL16PayClass * klass)
gstrtpbasepayload_class->set_caps = gst_rtp_L16_pay_setcaps;
gstrtpbasepayload_class->get_caps = gst_rtp_L16_pay_getcaps;
gstrtpbasepayload_class->handle_buffer = gst_rtp_L16_pay_handle_buffer;
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_rtp_L16_pay_src_template));
@ -113,72 +117,54 @@ static gboolean
gst_rtp_L16_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
{
GstRtpL16Pay *rtpL16pay;
GstStructure *structure;
gint channels, rate;
gboolean res;
gchar *params;
GstAudioChannelPosition *pos;
GstAudioInfo *info;
const GstRTPChannelOrder *order;
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
rtpL16pay = GST_RTP_L16_PAY (basepayload);
structure = gst_caps_get_structure (caps, 0);
info = &rtpL16pay->info;
gst_audio_info_init (info);
if (!gst_audio_info_from_caps (info, caps))
goto invalid_caps;
/* first parse input caps */
if (!gst_structure_get_int (structure, "rate", &rate))
goto no_rate;
order = gst_rtp_channels_get_by_pos (info->channels, info->position);
rtpL16pay->order = order;
if (!gst_structure_get_int (structure, "channels", &channels))
goto no_channels;
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L16",
info->rate);
params = g_strdup_printf ("%d", info->channels);
/* get the channel order */
pos = gst_audio_get_channel_positions (structure);
if (pos)
order = gst_rtp_channels_get_by_pos (channels, pos);
else
order = NULL;
gst_rtp_base_payload_set_options (basepayload, "audio", TRUE, "L16", rate);
params = g_strdup_printf ("%d", channels);
if (!order && channels > 2) {
if (!order && info->channels > 2) {
GST_ELEMENT_WARNING (rtpL16pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", channels));
(NULL), ("Unknown channel order for %d channels", info->channels));
}
if (order && order->name) {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, "channel-order", G_TYPE_STRING, order->name, NULL);
info->channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, NULL);
info->channels, NULL);
}
g_free (params);
g_free (pos);
rtpL16pay->rate = rate;
rtpL16pay->channels = channels;
/* octet-per-sample is 2 * channels for L16 */
gst_rtp_base_audio_payload_set_sample_options (rtpbaseaudiopayload,
2 * rtpL16pay->channels);
2 * info->channels);
return res;
/* ERRORS */
no_rate:
invalid_caps:
{
GST_DEBUG_OBJECT (rtpL16pay, "no rate given");
return FALSE;
}
no_channels:
{
GST_DEBUG_OBJECT (rtpL16pay, "no channels given");
GST_DEBUG_OBJECT (rtpL16pay, "invalid caps");
return FALSE;
}
}
@ -232,6 +218,26 @@ gst_rtp_L16_pay_getcaps (GstRTPBasePayload * rtppayload, GstPad * pad,
return caps;
}
static GstFlowReturn
gst_rtp_L16_pay_handle_buffer (GstRTPBasePayload * basepayload,
GstBuffer * buffer)
{
GstRtpL16Pay *rtpL16pay;
rtpL16pay = GST_RTP_L16_PAY (basepayload);
buffer = gst_buffer_make_writable (buffer);
if (rtpL16pay->order &&
!gst_audio_buffer_reorder_channels (buffer, rtpL16pay->info.finfo->format,
rtpL16pay->info.channels, rtpL16pay->info.position,
rtpL16pay->order->pos)) {
return GST_FLOW_ERROR;
}
return GST_RTP_BASE_PAYLOAD_CLASS (parent_class)->handle_buffer (basepayload,
buffer);
}
gboolean
gst_rtp_L16_pay_plugin_init (GstPlugin * plugin)
{

6
gst/rtp/gstrtpL16pay.h
View file

@ -23,6 +23,8 @@
#include <gst/gst.h>
#include <gst/rtp/gstrtpbaseaudiopayload.h>
#include "gstrtpchannels.h"
G_BEGIN_DECLS
#define GST_TYPE_RTP_L16_PAY \
@ -43,8 +45,8 @@ struct _GstRtpL16Pay
{
GstRTPBaseAudioPayload payload;
gint rate;
gint channels;
GstAudioInfo info;
const GstRTPChannelOrder *order;
};
struct _GstRtpL16PayClass

21
gst/rtp/gstrtpchannels.c
View file

@ -33,14 +33,16 @@ static gboolean
check_channels (const GstRTPChannelOrder * order,
const GstAudioChannelPosition * pos)
{
gint i;
gint i, j;
gboolean res = TRUE;
for (i = 0; i < order->channels; i++) {
if (order->pos[i] != pos[i]) {
res = FALSE;
break;
for (j = 0; j < order->channels; j++) {
if (order->pos[j] == pos[i])
break;
}
if (j == order->channels)
return FALSE;
}
return res;
}
@ -150,18 +152,13 @@ gst_rtp_channels_get_by_index (gint channels, guint idx)
* Returns: a #GstAudioChannelPosition with all the channel position info set to
* #GST_AUDIO_CHANNEL_POSITION_NONE.
*/
GstAudioChannelPosition *
gst_rtp_channels_create_default (gint channels)
void
gst_rtp_channels_create_default (gint channels, GstAudioChannelPosition * posn)
{
gint i;
GstAudioChannelPosition *posn;
g_return_val_if_fail (channels > 0, NULL);
posn = g_new (GstAudioChannelPosition, channels);
g_return_if_fail (channels > 0);
for (i = 0; i < channels; i++)
posn[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
return posn;
}

28
gst/rtp/gstrtpchannels.h
View file

@ -21,7 +21,9 @@
#include <stdlib.h>
#include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
#ifndef __GST_RTP_CHANNELS_H__
#define __GST_RTP_CHANNELS_H__
typedef struct
{
@ -41,14 +43,14 @@ static const GstAudioChannelPosition pos_4_2[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE
GST_AUDIO_CHANNEL_POSITION_LFE1
};
static const GstAudioChannelPosition pos_4_3[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE
GST_AUDIO_CHANNEL_POSITION_LFE1
};
static const GstAudioChannelPosition pos_5_1[] = {
@ -65,14 +67,14 @@ static const GstAudioChannelPosition pos_6_1[] = {
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE
GST_AUDIO_CHANNEL_POSITION_LFE1
};
static const GstAudioChannelPosition pos_6_2[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT
};
@ -81,7 +83,7 @@ static const GstAudioChannelPosition pos_8_1[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
@ -92,7 +94,7 @@ static const GstAudioChannelPosition pos_8_2[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
@ -103,7 +105,7 @@ static const GstAudioChannelPosition pos_8_3[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_LFE,
GST_AUDIO_CHANNEL_POSITION_LFE1,
GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
@ -111,7 +113,7 @@ static const GstAudioChannelPosition pos_8_3[] = {
};
static const GstAudioChannelPosition pos_def_1[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_MONO
GST_AUDIO_CHANNEL_POSITION_MONO
};
static const GstAudioChannelPosition pos_def_2[] = {
@ -129,7 +131,7 @@ static const GstAudioChannelPosition pos_def_4[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE
GST_AUDIO_CHANNEL_POSITION_LFE1
};
static const GstAudioChannelPosition pos_def_5[] = {
@ -146,7 +148,7 @@ static const GstAudioChannelPosition pos_def_6[] = {
GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
GST_AUDIO_CHANNEL_POSITION_LFE
GST_AUDIO_CHANNEL_POSITION_LFE1
};
static const GstRTPChannelOrder channel_orders[] =
@ -183,4 +185,6 @@ const GstRTPChannelOrder * gst_rtp_channels_get_by_order (gint channels,
const gchar *order);
const GstRTPChannelOrder * gst_rtp_channels_get_by_index (gint channels, guint idx);
GstAudioChannelPosition * gst_rtp_channels_create_default (gint channels);
void gst_rtp_channels_create_default (gint channels, GstAudioChannelPosition *pos);
#endif /* __GST_RTP_CHANNELS_H__ */

6
gst/rtp/gstrtpg722depay.c
View file

@ -25,7 +25,6 @@
#include <stdlib.h>
#include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
#include "gstrtpg722depay.h"
#include "gstrtpchannels.h"
@ -127,8 +126,10 @@ gst_rtp_g722_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
gint channels;
GstCaps *srccaps;
gboolean res;
#if 0
const gchar *channel_order;
const GstRTPChannelOrder *order;
#endif
rtpg722depay = GST_RTP_G722_DEPAY (depayload);
@ -179,6 +180,8 @@ gst_rtp_g722_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
srccaps = gst_caps_new_simple ("audio/G722",
"rate", G_TYPE_INT, samplerate, "channels", G_TYPE_INT, channels, NULL);
/* FIXME: Do something with the channel order */
#if 0
/* add channel positions */
channel_order = gst_structure_get_string (structure, "channel-order");
@ -197,6 +200,7 @@ gst_rtp_g722_depay_setcaps (GstRTPBaseDepayload * depayload, GstCaps * caps)
gst_audio_set_channel_positions (gst_caps_get_structure (srccaps, 0), pos);
g_free (pos);
}
#endif
res = gst_pad_set_caps (depayload->srcpad, srccaps);
gst_caps_unref (srccaps);

12
gst/rtp/gstrtpg722pay.c
View file

@ -24,7 +24,6 @@
#include <string.h>
#include <gst/audio/audio.h>
#include <gst/audio/multichannel.h>
#include <gst/rtp/gstrtpbuffer.h>
#include "gstrtpg722pay.h"
@ -105,8 +104,10 @@ gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
gint rate, channels, clock_rate;
gboolean res;
gchar *params;
#if 0
GstAudioChannelPosition *pos;
const GstRTPChannelOrder *order;
#endif
GstRTPBaseAudioPayload *rtpbaseaudiopayload;
rtpbaseaudiopayload = GST_RTP_BASE_AUDIO_PAYLOAD (basepayload);
@ -121,12 +122,15 @@ gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
if (!gst_structure_get_int (structure, "channels", &channels))
goto no_channels;
/* FIXME: Do something with the channel positions */
#if 0
/* get the channel order */
pos = gst_audio_get_channel_positions (structure);
if (pos)
order = gst_rtp_channels_get_by_pos (channels, pos);
else
order = NULL;
#endif
/* Clock rate is always 8000 Hz for G722 according to
* RFC 3551 although the sampling rate is 16000 Hz */
@ -136,6 +140,7 @@ gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
clock_rate);
params = g_strdup_printf ("%d", channels);
#if 0
if (!order && channels > 2) {
GST_ELEMENT_WARNING (rtpg722pay, STREAM, DECODE,
(NULL), ("Unknown channel order for %d channels", channels));
@ -146,13 +151,18 @@ gst_rtp_g722_pay_setcaps (GstRTPBasePayload * basepayload, GstCaps * caps)
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, "channel-order", G_TYPE_STRING, order->name, NULL);
} else {
#endif
res = gst_rtp_base_payload_set_outcaps (basepayload,
"encoding-params", G_TYPE_STRING, params, "channels", G_TYPE_INT,
channels, NULL);
#if 0
}
#endif
g_free (params);
#if 0
g_free (pos);
#endif
rtpg722pay->rate = rate;
rtpg722pay->channels = channels;

2
gst/rtp/gstrtpvrawpay.c
View file

@ -271,7 +271,7 @@ gst_rtp_vraw_pay_handle_buffer (GstRTPBasePayload * payload, GstBuffer * buffer)
width = GST_VIDEO_INFO_WIDTH (&rtpvrawpay->vinfo);
height = GST_VIDEO_INFO_HEIGHT (&rtpvrawpay->vinfo);
interlaced = !!(rtpvrawpay->vinfo.flags & GST_VIDEO_FLAG_INTERLACED);
interlaced = ! !(rtpvrawpay->vinfo.flags & GST_VIDEO_FLAG_INTERLACED);
/* start with line 0, offset 0 */
for (field = 0; field < 1 + interlaced; field++) {