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gst/rtp/gstrtpmp4gpay.*: Make more things work.
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:
parent
e2a22e497c
commit
28906ed743
2 changed files with 217 additions and 11 deletions
10
ChangeLog
10
ChangeLog
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@ -1,3 +1,13 @@
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2006-02-12 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init),
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(gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_parse_audio_config),
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(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
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(gst_rtp_mp4g_pay_setcaps), (gst_rtp_mp4g_pay_flush):
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* gst/rtp/gstrtpmp4gpay.h:
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Make more things work.
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Handle ACC config strings.
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2006-02-12 Thomas Vander Stichele <thomas at apestaart dot org>
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2006-02-12 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
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* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
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@ -50,7 +50,7 @@ GST_STATIC_PAD_TEMPLATE ("src",
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"media = (string) \"video\", "
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"media = (string) \"video\", "
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"payload = (int) [ 96, 127 ], "
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"payload = (int) [ 96, 127 ], "
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"clock-rate = (int) [1, MAX ], "
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"clock-rate = (int) [1, MAX ], "
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"encoding-name = (string) \"MP4G-ES\", "
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"encoding-name = (string) \"mpeg4-generic\", "
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/* required string params */
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/* required string params */
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"streamtype = (string) { \"4\", \"5\" }, " /* 4 = video, 5 = audio */
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"streamtype = (string) { \"4\", \"5\" }, " /* 4 = video, 5 = audio */
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"profile-level-id = (int) [1,MAX], "
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"profile-level-id = (int) [1,MAX], "
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@ -165,6 +165,7 @@ gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay)
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rtpmp4gpay->adapter = gst_adapter_new ();
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rtpmp4gpay->adapter = gst_adapter_new ();
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rtpmp4gpay->rate = 90000;
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rtpmp4gpay->rate = 90000;
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rtpmp4gpay->profile = 1;
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rtpmp4gpay->profile = 1;
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rtpmp4gpay->mode = "";
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}
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}
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static void
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static void
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@ -180,40 +181,234 @@ gst_rtp_mp4g_pay_finalize (GObject * object)
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G_OBJECT_CLASS (parent_class)->finalize (object);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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}
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#if 0
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static unsigned sampling_table[16] = {
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96000, 88200, 64000, 48000, 44100, 32000, 24000, 22050,
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16000, 12000, 11025, 8000, 7350, 0, 0, 0
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};
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static gboolean
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gst_rtp_mp4g_pay_parse_audio_config (GstRtpMP4GPay * rtpmp4gpay,
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GstBuffer * buffer)
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{
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guint8 *data;
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guint size;
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guint8 objectType;
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guint8 samplingIdx;
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guint8 channelCfg;
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data = GST_BUFFER_DATA (buffer);
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size = GST_BUFFER_SIZE (buffer);
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if (size < 2)
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goto too_short;
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/* only AAC LC for now */
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objectType = (data[0] & 0xf8) >> 3;
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if (objectType != 2)
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goto unsupported_type;
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samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
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/* only fixed values for now */
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if (samplingIdx > 12 && samplingIdx != 15)
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goto wrong_freq;
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channelCfg = ((data[1] & 0x78) >> 3);
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if (channelCfg > 7)
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goto wrong_channels;
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/* rtp rate depends on sampling rate of the audio */
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if (samplingIdx == 15) {
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if (size < 5)
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goto too_short;
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/* index of 15 means we get the rate in the next 24 bits */
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rtpmp4gpay->rate = ((data[1] & 0x7f) << 17) |
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((data[2]) << 9) | ((data[3]) << 1) | ((data[4] & 0x80) >> 7);
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} else {
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/* else use the rate from the table */
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rtpmp4gpay->rate = sampling_table[samplingIdx];
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}
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/* extra rtp params contain the number of channels */
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rtpmp4gpay->params = channelCfg;
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/* audio stream type */
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rtpmp4gpay->streamtype = 5;
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/* mode */
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rtpmp4gpay->mode = "ACC-hbr";
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/* profile (should be 1) */
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rtpmp4gpay->profile = objectType - 1;
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GST_DEBUG_OBJECT (rtpmp4gpay,
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"objectType: %d, samplingIdx: %d (%d), channelCfg: %d", objectType,
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samplingIdx, rtpmp4gpay->rate, channelCfg);
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return TRUE;
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/* ERROR */
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too_short:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
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(NULL), ("config string too short"));
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return FALSE;
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}
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unsupported_type:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
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(NULL), ("unsupported object type %d", objectType));
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return FALSE;
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}
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wrong_freq:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
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(NULL), ("unsupported frequency index %d", samplingIdx));
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return FALSE;
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}
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wrong_channels:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, NOT_IMPLEMENTED,
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(NULL), ("unsupported number of channels %d", channelCfg));
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return FALSE;
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}
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}
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#define VOS_STARTCODE 0x000001B0
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static gboolean
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gst_rtp_mp4g_pay_parse_video_config (GstRtpMP4GPay * rtpmp4gpay,
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GstBuffer * buffer)
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{
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guint8 *data;
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guint size;
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guint32 code;
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data = GST_BUFFER_DATA (buffer);
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size = GST_BUFFER_SIZE (buffer);
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if (size < 5)
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goto too_short;
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code = GST_READ_UINT32_BE (data);
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if (code == VOS_STARTCODE) {
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/* get profile */
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rtpmp4gpay->profile = data[4];
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} else {
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GST_ELEMENT_WARNING (rtpmp4gpay, STREAM, FORMAT,
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(NULL), ("profile not found in config string"));
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rtpmp4gpay->profile = 1;
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}
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/* fixed rate */
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rtpmp4gpay->rate = 90000;
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/* video stream type */
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rtpmp4gpay->streamtype = 4;
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/* no params for video */
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rtpmp4gpay->params = 0;
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/* mode */
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rtpmp4gpay->mode = "generic";
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GST_LOG_OBJECT (rtpmp4gpay, "profile %d", rtpmp4gpay->profile);
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return TRUE;
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/* ERROR */
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too_short:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
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(NULL), ("config string too short"));
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return FALSE;
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}
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}
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static void
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static void
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gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay)
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gst_rtp_mp4g_pay_new_caps (GstRtpMP4GPay * rtpmp4gpay)
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{
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{
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gchar *profile, *config;
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gchar *config;
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GValue v = { 0 };
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GValue v = { 0 };
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profile = g_strdup_printf ("%d", rtpmp4gpay->profile);
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#define MP4GCAPS \
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"streamtype", G_TYPE_INT, rtpmp4gpay->streamtype, \
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"profile-level-id", G_TYPE_INT, rtpmp4gpay->profile, \
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"mode", G_TYPE_STRING, rtpmp4gpay->mode, \
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"config", G_TYPE_STRING, config, \
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"sizelength", G_TYPE_INT, 13, \
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"indexlength", G_TYPE_INT, 3, \
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"indexdeltalength", G_TYPE_INT, 3, \
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NULL
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g_value_init (&v, GST_TYPE_BUFFER);
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g_value_init (&v, GST_TYPE_BUFFER);
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gst_value_set_buffer (&v, rtpmp4gpay->config);
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gst_value_set_buffer (&v, rtpmp4gpay->config);
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config = gst_value_serialize (&v);
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config = gst_value_serialize (&v);
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gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
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/* hmm, silly */
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"profile-level-id", G_TYPE_STRING, profile,
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if (rtpmp4gpay->params) {
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"config", G_TYPE_STRING, config, NULL);
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gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
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"encoding-params", G_TYPE_INT, rtpmp4gpay->params, MP4GCAPS);
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} else {
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gst_basertppayload_set_outcaps (GST_BASE_RTP_PAYLOAD (rtpmp4gpay),
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MP4GCAPS);
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}
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g_value_unset (&v);
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g_value_unset (&v);
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g_free (profile);
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g_free (config);
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g_free (config);
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#undef MP4GCAPS
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}
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}
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#endif
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static gboolean
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static gboolean
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gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload, GstCaps * caps)
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{
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{
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GstRtpMP4GPay *rtpmp4gpay;
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GstRtpMP4GPay *rtpmp4gpay;
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GstStructure *structure;
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const GValue *codec_info;
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gboolean res = TRUE;
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rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
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rtpmp4gpay = GST_RTP_MP4G_PAY (payload);
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structure = gst_caps_get_structure (caps, 0);
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codec_info = gst_structure_get_value (structure, "codec_info");
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if (codec_info) {
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GST_LOG_OBJECT (rtpmp4gpay, "got codec_info");
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if (G_VALUE_TYPE (codec_info) == GST_TYPE_BUFFER) {
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GstBuffer *buffer;
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const gchar *name;
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buffer = gst_value_get_buffer (codec_info);
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GST_LOG_OBJECT (rtpmp4gpay, "configuring codec_info");
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name = gst_structure_get_name (structure);
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/* parse buffer */
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if (!strcmp (name, "audio/mpeg")) {
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res = gst_rtp_mp4g_pay_parse_audio_config (rtpmp4gpay, buffer);
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} else if (!strcmp (name, "video/mpeg")) {
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res = gst_rtp_mp4g_pay_parse_video_config (rtpmp4gpay, buffer);
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} else {
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res = FALSE;
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}
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if (!res)
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goto config_failed;
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/* now we can configure the buffer */
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if (rtpmp4gpay->config)
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gst_buffer_unref (rtpmp4gpay->config);
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rtpmp4gpay->config = gst_buffer_copy (buffer);
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}
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}
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gst_basertppayload_set_options (payload, "video", TRUE, "mpeg4-generic",
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gst_basertppayload_set_options (payload, "video", TRUE, "mpeg4-generic",
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rtpmp4gpay->rate);
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rtpmp4gpay->rate);
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return TRUE;
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gst_rtp_mp4g_pay_new_caps (rtpmp4gpay);
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return res;
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/* ERRORS */
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config_failed:
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{
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GST_DEBUG_OBJECT (rtpmp4gpay, "failed to parse config");
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return FALSE;
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}
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}
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}
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static GstFlowReturn
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static GstFlowReturn
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@ -303,6 +498,7 @@ gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay)
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gst_adapter_flush (rtpmp4gpay->adapter, payload_len);
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gst_adapter_flush (rtpmp4gpay->adapter, payload_len);
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/* marker only if the packet is complete */
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gst_rtp_buffer_set_marker (outbuf, avail > payload_len);
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gst_rtp_buffer_set_marker (outbuf, avail > payload_len);
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GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_ts;
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GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_ts;
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