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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-30 13:41:48 +00:00
rtpmp4gpay: implement perfect timestamps
Use bitreader for parsing the config string Reset state variables when going to READY Parse frame length and use it to keep track of the rtptimestamps
This commit is contained in:
Wim Taymans
parent
29b32853d5
commit
af70b300cc
2 changed files with 100 additions and 22 deletions
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@ -23,6 +23,7 @@
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#include <string.h>
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#include <gst/base/gstbitreader.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpmp4gpay.h"
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@ -74,6 +75,9 @@ GST_STATIC_PAD_TEMPLATE ("src",
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static void gst_rtp_mp4g_pay_finalize (GObject * object);
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static GstStateChangeReturn gst_rtp_mp4g_pay_change_state (GstElement * element,
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GstStateChange transition);
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static gboolean gst_rtp_mp4g_pay_setcaps (GstBaseRTPPayload * payload,
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GstCaps * caps);
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static GstFlowReturn gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload *
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@ -101,13 +105,17 @@ static void
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gst_rtp_mp4g_pay_class_init (GstRtpMP4GPayClass * klass)
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{
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GObjectClass *gobject_class;
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GstElementClass *gstelement_class;
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GstBaseRTPPayloadClass *gstbasertppayload_class;
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gobject_class = (GObjectClass *) klass;
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gstelement_class = (GstElementClass *) klass;
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gstbasertppayload_class = (GstBaseRTPPayloadClass *) klass;
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gobject_class->finalize = gst_rtp_mp4g_pay_finalize;
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gstelement_class->change_state = gst_rtp_mp4g_pay_change_state;
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gstbasertppayload_class->set_caps = gst_rtp_mp4g_pay_setcaps;
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gstbasertppayload_class->handle_buffer = gst_rtp_mp4g_pay_handle_buffer;
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@ -119,20 +127,14 @@ static void
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gst_rtp_mp4g_pay_init (GstRtpMP4GPay * rtpmp4gpay, GstRtpMP4GPayClass * klass)
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{
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rtpmp4gpay->adapter = gst_adapter_new ();
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rtpmp4gpay->rate = 90000;
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rtpmp4gpay->profile = g_strdup ("1");
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rtpmp4gpay->mode = "";
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}
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static void
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gst_rtp_mp4g_pay_finalize (GObject * object)
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gst_rtp_mp4g_pay_reset (GstRtpMP4GPay * rtpmp4gpay)
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{
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GstRtpMP4GPay *rtpmp4gpay;
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GST_DEBUG_OBJECT (rtpmp4gpay, "reset");
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rtpmp4gpay = GST_RTP_MP4G_PAY (object);
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g_object_unref (rtpmp4gpay->adapter);
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rtpmp4gpay->adapter = NULL;
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gst_adapter_clear (rtpmp4gpay->adapter);
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g_free (rtpmp4gpay->params);
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rtpmp4gpay->params = NULL;
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@ -144,6 +146,25 @@ gst_rtp_mp4g_pay_finalize (GObject * object)
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g_free (rtpmp4gpay->profile);
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rtpmp4gpay->profile = NULL;
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rtpmp4gpay->streamtype = NULL;
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rtpmp4gpay->mode = NULL;
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rtpmp4gpay->frame_len = 0;
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rtpmp4gpay->offset = 0;
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}
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static void
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gst_rtp_mp4g_pay_finalize (GObject * object)
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{
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GstRtpMP4GPay *rtpmp4gpay;
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rtpmp4gpay = GST_RTP_MP4G_PAY (object);
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gst_rtp_mp4g_pay_reset (rtpmp4gpay);
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g_object_unref (rtpmp4gpay->adapter);
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rtpmp4gpay->adapter = NULL;
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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@ -161,39 +182,66 @@ gst_rtp_mp4g_pay_parse_audio_config (GstRtpMP4GPay * rtpmp4gpay,
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guint8 objectType;
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guint8 samplingIdx;
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guint8 channelCfg;
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GstBitReader br;
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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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gst_bit_reader_init (&br, data, size);
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/* any object type is fine, we need to copy it to the profile-level-id field. */
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objectType = (data[0] & 0xf8) >> 3;
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if (!gst_bit_reader_get_bits_uint8 (&br, &objectType, 5))
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goto too_short;
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if (objectType == 0)
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goto invalid_object;
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samplingIdx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
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if (!gst_bit_reader_get_bits_uint8 (&br, &samplingIdx, 4))
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goto too_short;
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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 (!gst_bit_reader_get_bits_uint8 (&br, &channelCfg, 4))
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goto too_short;
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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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guint32 rate;
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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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if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24))
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goto too_short;
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rtpmp4gpay->rate = rate;
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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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rtpmp4gpay->frame_len = 1024;
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switch (objectType) {
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case 1:
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case 2:
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case 3:
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case 4:
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case 6:
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case 7:
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{
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guint8 frameLenFlag;
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if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1))
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if (frameLenFlag)
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rtpmp4gpay->frame_len = 960;
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break;
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}
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default:
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break;
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}
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/* extra rtp params contain the number of channels */
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g_free (rtpmp4gpay->params);
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rtpmp4gpay->params = g_strdup_printf ("%d", channelCfg);
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@ -206,8 +254,9 @@ gst_rtp_mp4g_pay_parse_audio_config (GstRtpMP4GPay * rtpmp4gpay,
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rtpmp4gpay->profile = g_strdup_printf ("%d", objectType);
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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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"objectType: %d, samplingIdx: %d (%d), channelCfg: %d, frame_len %d",
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objectType, samplingIdx, rtpmp4gpay->rate, channelCfg,
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rtpmp4gpay->frame_len);
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return TRUE;
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@ -215,13 +264,13 @@ gst_rtp_mp4g_pay_parse_audio_config (GstRtpMP4GPay * rtpmp4gpay,
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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, expected 2 bytes, got %d", size));
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(NULL), ("config string too short"));
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return FALSE;
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}
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invalid_object:
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{
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GST_ELEMENT_ERROR (rtpmp4gpay, STREAM, FORMAT,
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(NULL), ("invalid object type 0"));
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(NULL), ("invalid object type"));
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return FALSE;
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}
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wrong_freq:
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@ -479,6 +528,11 @@ gst_rtp_mp4g_pay_flush (GstRtpMP4GPay * rtpmp4gpay)
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GST_BUFFER_TIMESTAMP (outbuf) = rtpmp4gpay->first_timestamp;
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GST_BUFFER_DURATION (outbuf) = rtpmp4gpay->first_duration;
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if (rtpmp4gpay->frame_len) {
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GST_BUFFER_OFFSET (outbuf) = rtpmp4gpay->offset;
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rtpmp4gpay->offset += rtpmp4gpay->frame_len;
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}
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ret = gst_basertppayload_push (GST_BASE_RTP_PAYLOAD (rtpmp4gpay), outbuf);
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avail -= payload_len;
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@ -503,9 +557,31 @@ gst_rtp_mp4g_pay_handle_buffer (GstBaseRTPPayload * basepayload,
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/* we always encode and flush a full AU */
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gst_adapter_push (rtpmp4gpay->adapter, buffer);
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return gst_rtp_mp4g_pay_flush (rtpmp4gpay);
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}
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static GstStateChangeReturn
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gst_rtp_mp4g_pay_change_state (GstElement * element, GstStateChange transition)
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{
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GstStateChangeReturn ret;
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GstRtpMP4GPay *rtpmp4gpay;
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rtpmp4gpay = GST_RTP_MP4G_PAY (element);
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_PAUSED:
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gst_rtp_mp4g_pay_reset (rtpmp4gpay);
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break;
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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return ret;
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}
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gboolean
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gst_rtp_mp4g_pay_plugin_init (GstPlugin * plugin)
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{
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@ -48,6 +48,7 @@ struct _GstRtpMP4GPay
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GstClockTime first_timestamp;
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GstClockTime first_duration;
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GstClockTime duration;
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guint64 offset;
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gint rate;
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gchar *params;
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@ -55,6 +56,7 @@ struct _GstRtpMP4GPay
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const gchar *streamtype;
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const gchar *mode;
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GstBuffer *config;
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guint frame_len;
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};
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struct _GstRtpMP4GPayClass
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