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gst/rtp/gstrtpmp4gpay.*: Make more things work.

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Original commit message from CVS:
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init),
(gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_parse_audio_config),
(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
(gst_rtp_mp4g_pay_setcaps), (gst_rtp_mp4g_pay_flush):
* gst/rtp/gstrtpmp4gpay.h:
Make more things work.
Handle ACC config strings.
This commit is contained in:
Wim Taymans 2006-02-12 18:59:36 +00:00
parent e2a22e497c
commit 28906ed743
2 changed files with 217 additions and 11 deletions
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10
ChangeLog
View file

@ -1,3 +1,13 @@
2006-02-12 Wim Taymans <wim@fluendo.com>
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init),
(gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_parse_audio_config),
(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
(gst_rtp_mp4g_pay_setcaps), (gst_rtp_mp4g_pay_flush):
* gst/rtp/gstrtpmp4gpay.h:
Make more things work.
Handle ACC config strings.
2006-02-12 Thomas Vander Stichele <thomas at apestaart dot org>
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):

218
gst/rtp/gstrtpmp4gpay.c
View file

@ -50,7 +50,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
"media = (string) \"video\", "
"payload = (int) [ 96, 127 ], "
"clock-rate = (int) [1, MAX ], "
"encoding-name = (string) \"MP4G-ES\", "
"encoding-name = (string) \"mpeg4-generic\", "
/* required string params */
"streamtype = (string) { \"4\", \"5\" }, " /* 4 = video, 5 = audio */
"profile-level-id = (int) [1,MAX], "
@ -165,6 +165,7 @@ gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay)
rtpmp4gpay->adapter = gst_adapter_new ();
rtpmp4gpay->rate = 90000;
rtpmp4gpay->profile = 1;
rtpmp4gpay->mode = "";
}
static void
@ -180,40 +181,234 @@ gst_rtp_mp4g_pay_finalize (GObject * object)
G_OBJECT_CLASS (parent_class)->finalize (object);
}
#if 0
static unsigned sampling_table[16] = {
96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
16000, 12000, 11025, 8000, 7350, 0, 0, 0
};
static gboolean
gst_rtp_mp4g_pay_parse_audio_config (GstRtpMP4GPay * rtpmp4gpay,
GstBuffer * buffer)
{
guint8 *data;
guint size;
guint8 objectType;
guint8 samplingIdx;
guint8 channelCfg;
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
if (size < 2)
goto too_short;
/* only AAC LC for now */
objectType = (data[0] & 0xf8) >> 3;
if (objectType != 2)
goto unsupported_type;
samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
/* only fixed values for now */
if (samplingIdx > 12 && samplingIdx != 15)
goto wrong_freq;
channelCfg = ((data[1] & 0x78) >> 3);
if (channelCfg > 7)
goto wrong_channels;
/* rtp rate depends on sampling rate of the audio */
if (samplingIdx == 15) {
if (size < 5)
goto too_short;
/* index of 15 means we get the rate in the next 24 bits */
rtpmp4gpay->rate = ((data[1] & 0x7f) << 17) |
((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7);
} else {
/* else use the rate from the table */
rtpmp4gpay->rate = sampling_table[samplingIdx];
}
/* extra rtp params contain the number of channels */
rtpmp4gpay->params = channelCfg;
/* audio stream type */
rtpmp4gpay->streamtype = 5;
/* mode */
rtpmp4gpay->mode = "ACC-hbr";
/* profile (should be 1) */
rtpmp4gpay->profile = objectType - 1;
GST_DEBUG_OBJECT (rtpmp4gpay,
"objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType,
samplingIdx, rtpmp4gpay->rate, channelCfg);
return TRUE;
/* ERROR */
too_short:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("config string too short"));
return FALSE;
}
unsupported_type:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported object type %d", objectType));
return FALSE;
}
wrong_freq:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported frequency index %d", samplingIdx));
return FALSE;
}
wrong_channels:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
(NULL), ("unsupported number of channels %d", channelCfg));
return FALSE;
}
}
#define VOS_STARTCODE 0x000001B0
static gboolean
gst_rtp_mp4g_pay_parse_video_config (GstRtpMP4GPay * rtpmp4gpay,
GstBuffer * buffer)
{
guint8 *data;
guint size;
guint32 code;
data = GST_BUFFER_DATA (buffer);
size = GST_BUFFER_SIZE (buffer);
if (size < 5)
goto too_short;
code = GST_READ_UINT32_BE (data);
if (code == VOS_STARTCODE) {
/* get profile */
rtpmp4gpay->profile = data[4];
} else {
GST_ELEMENT_WARNING (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("profile not found in config string"));
rtpmp4gpay->profile = 1;
}
/* fixed rate */
rtpmp4gpay->rate = 90000;
/* video stream type */
rtpmp4gpay->streamtype = 4;
/* no params for video */
rtpmp4gpay->params = 0;
/* mode */
rtpmp4gpay->mode = "generic";
GST_LOG_OBJECT (rtpmp4gpay, "profile %d", rtpmp4gpay->profile);
return TRUE;
/* ERROR */
too_short:
{
GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
(NULL), ("config string too short"));
return FALSE;
}
}
static void
gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay)
{
gchar *profile, *config;
gchar *config;
GValue v = { 0 };
profile = g_strdup_printf ("%d", rtpmp4gpay->profile);
#define MP4GCAPS \
"streamtype", G_TYPE_INT, rtpmp4gpay->streamtype, \
"profile-level-id", G_TYPE_INT, rtpmp4gpay->profile, \
"mode", G_TYPE_STRING, rtpmp4gpay->mode, \
"config", G_TYPE_STRING, config, \
"sizelength", G_TYPE_INT, 13, \
"indexlength", G_TYPE_INT, 3, \
"indexdeltalength", G_TYPE_INT, 3, \
NULL
g_value_init (&v, GST_TYPE_BUFFER);
gst_value_set_buffer (&v, rtpmp4gpay->config);
config = gst_value_serialize (&v);
gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
"profile-level-id", G_TYPE_STRING, profile,
"config", G_TYPE_STRING, config, NULL);
/* hmm, silly */
if (rtpmp4gpay->params) {
gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
"encoding-params", G_TYPE_INT, rtpmp4gpay->params, MP4GCAPS);
} else {
gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
MP4GCAPS);
}
g_value_unset (&v);
g_free (profile);
g_free (config);
#undef MP4GCAPS
}
#endif
static gboolean
gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
{
GstRtpMP4GPay *rtpmp4gpay;
GstStructure *structure;
const GValue *codec_info;
gboolean res = TRUE;
rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
structure = gst_caps_get_structure (caps, 0);
codec_info = gst_structure_get_value (structure, "codec_info");
if (codec_info) {
GST_LOG_OBJECT (rtpmp4gpay, "got codec_info");
if (G_VALUE_TYPE (codec_info) == GST_TYPE_BUFFER) {
GstBuffer *buffer;
const gchar *name;
buffer = gst_value_get_buffer (codec_info);
GST_LOG_OBJECT (rtpmp4gpay, "configuring codec_info");
name = gst_structure_get_name (structure);
/* parse buffer */
if (!strcmp (name, "audio/mpeg")) {
res = gst_rtp_mp4g_pay_parse_audio_config (rtpmp4gpay, buffer);
} else if (!strcmp (name, "video/mpeg")) {
res = gst_rtp_mp4g_pay_parse_video_config (rtpmp4gpay, buffer);
} else {
res = FALSE;
}
if (!res)
goto config_failed;
/* now we can configure the buffer */
if (rtpmp4gpay->config)
gst_buffer_unref (rtpmp4gpay->config);
rtpmp4gpay->config = gst_buffer_copy (buffer);
}
}
gst_basertppayload_set_options (payload, "video", TRUE, "mpeg4-generic",
rtpmp4gpay->rate);
return TRUE;
gst_rtp_mp4g_pay_new_caps (rtpmp4gpay);
return res;
/* ERRORS */
config_failed:
{
GST_DEBUG_OBJECT (rtpmp4gpay, "failed to parse config");
return FALSE;
}
}
static GstFlowReturn
@ -303,6 +498,7 @@ gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay)
gst_adapter_flush (rtpmp4gpay->adapter, payload_len);
/* marker only if the packet is complete */
gst_rtp_buffer_set_marker (outbuf, avail > payload_len);
GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_ts;