mp4adepay: improve timestamps on outgoing packets

Improve parsing of the samplerate.
Parse the framelen so that we can calculate timestamps.
When interpollate the incomming timestamp on outgoing buffers when there are
multiple subframes.

fixes #625825

1 files changed, 77 insertions, 16 deletions

diff --git a/gst/rtp/gstrtpmp4adepay.c b/gst/rtp/gstrtpmp4adepay.c
index 836ff9f09..394fc0d28 100644
--- a/gst/rtp/gstrtpmp4adepay.c
+++ b/gst/rtp/gstrtpmp4adepay.c
@@ -21,6 +21,7 @@
 #  include "config.h"
 #endif
 
+#include <gst/base/gstbitreader.h>
 #include <gst/rtp/gstrtpbuffer.h>
 
 #include <string.h>
@@ -128,6 +129,10 @@ gst_rtp_mp4a_depay_finalize (GObject * object)
   G_OBJECT_CLASS (parent_class)->finalize (object);
 }
 
+static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000,
+  44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000
+};
+
 static gboolean
 gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
 {
@@ -165,10 +170,9 @@ gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
       guint8 *data;
       guint size;
       gint i;
-      guint sr_idx;
-      static const guint aac_sample_rates[] = { 96000, 88200, 64000, 48000,
-        44100, 32000, 24000, 22050, 16000, 12000, 11025, 8000
-      };
+      guint32 rate;
+      guint8 obj_type, sr_idx, channels;
+      GstBitReader br;
 
       buffer = gst_value_get_buffer (&v);
       gst_buffer_ref (buffer);
@@ -210,22 +214,68 @@ gst_rtp_mp4a_depay_setcaps (GstBaseRTPDepayload * depayload, GstCaps * caps)
       for (i = 0; i < size; i++) {
         data[i] = ((data[i + 1] & 1) << 7) | ((data[i + 2] & 0xfe) >> 1);
       }
+      /* ignore remaining bit, we're only interested in full bytes */
+      GST_BUFFER_SIZE (buffer) = size;
+
+      gst_bit_reader_init (&br, data, size);
+
+      /* any object type is fine, we need to copy it to the profile-level-id field. */
+      if (!gst_bit_reader_get_bits_uint8 (&br, &obj_type, 5))
+        goto bad_config;
+      if (obj_type == 0) {
+        GST_WARNING_OBJECT (depayload, "invalid object type 0");
+        goto bad_config;
+      }
+
+      if (!gst_bit_reader_get_bits_uint8 (&br, &sr_idx, 4))
+        goto bad_config;
+      if (sr_idx > 12 && sr_idx != 15) {
+        GST_WARNING_OBJECT (depayload, "invalid sample rate index %d", sr_idx);
+        goto bad_config;
+      }
+      GST_LOG_OBJECT (rtpmp4adepay, "sample rate index %u", sr_idx);
 
-      /* grab and set sampling rate */
-      sr_idx = ((data[0] & 0x07) << 1) | ((data[1] & 0x80) >> 7);
-      if (sr_idx < G_N_ELEMENTS (aac_sample_rates)) {
-        gst_caps_set_simple (srccaps,
-            "rate", G_TYPE_INT, (gint) aac_sample_rates[sr_idx], NULL);
-        GST_DEBUG_OBJECT (depayload, "sampling rate from stream-config %u",
-            aac_sample_rates[sr_idx]);
+      if (!gst_bit_reader_get_bits_uint8 (&br, &channels, 4))
+        goto bad_config;
+      if (channels > 7) {
+        GST_WARNING_OBJECT (depayload, "invalid channels %u", (guint) channels);
+        goto bad_config;
+      }
+
+      /* rtp rate depends on sampling rate of the audio */
+      if (sr_idx == 15) {
+        /* index of 15 means we get the rate in the next 24 bits */
+        if (!gst_bit_reader_get_bits_uint32 (&br, &rate, 24))
+          goto bad_config;
       } else {
-        GST_WARNING_OBJECT (depayload, "Invalid sample rate index %u", sr_idx);
+        /* else use the rate from the table */
+        rate = aac_sample_rates[sr_idx];
       }
 
-      /* ignore remaining bit, we're only interested in full bytes */
-      GST_BUFFER_SIZE (buffer) = size;
+      rtpmp4adepay->frame_len = 1024;
+
+      switch (obj_type) {
+        case 1:
+        case 2:
+        case 3:
+        case 4:
+        case 6:
+        case 7:
+        {
+          guint8 frameLenFlag = 0;
+
+          if (gst_bit_reader_get_bits_uint8 (&br, &frameLenFlag, 1))
+            if (frameLenFlag)
+              rtpmp4adepay->frame_len = 960;
+          break;
+        }
+        default:
+          break;
+      }
 
       gst_caps_set_simple (srccaps,
+          "channels", G_TYPE_INT, (gint) channels,
+          "rate", G_TYPE_INT, (gint) rate,
           "codec_data", GST_TYPE_BUFFER, buffer, NULL);
       gst_buffer_unref (buffer);
     } else {
@@ -254,6 +304,7 @@ gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
 
   outbuf = gst_rtp_buffer_get_payload_buffer (buf);
 
+  gst_buffer_copy_metadata (outbuf, buf, GST_BUFFER_COPY_TIMESTAMPS);
   gst_adapter_push (rtpmp4adepay->adapter, outbuf);
 
   /* RTP marker bit indicates the last packet of the AudioMuxElement => create
@@ -315,11 +366,19 @@ gst_rtp_mp4a_depay_process (GstBaseRTPDepayload * depayload, GstBuffer * buf)
       data += skip;
       avail -= skip;
 
+      if (offset > 0 && timestamp != -1 && depayload->clock_rate != 0) {
+        timestamp +=
+            gst_util_uint64_scale_int (offset, GST_SECOND,
+            depayload->clock_rate);
+      }
+
       GST_BUFFER_TIMESTAMP (tmp) = timestamp;
       gst_base_rtp_depayload_push (depayload, tmp);
 
-      /* only apply the timestamp for the first buffer */
-      timestamp = -1;
+      /* calculate offsets for next buffers */
+      if (rtpmp4adepay->frame_len) {
+        offset += rtpmp4adepay->frame_len;
+      }
     }
 
     /* just a check that lengths match */
@@ -355,6 +414,8 @@ gst_rtp_mp4a_depay_change_state (GstElement * element,
   switch (transition) {
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       gst_adapter_clear (rtpmp4adepay->adapter);
+      rtpmp4adepay->frame_len = 0;
+      rtpmp4adepay->numSubFrames = 0;
       break;
     default:
       break;