diff --git a/libavformat/mpeg.c b/libavformat/mpeg.c
index e3c458c560..294c807b07 100644
--- a/libavformat/mpeg.c
+++ b/libavformat/mpeg.c
@@ -598,6 +598,9 @@ static int mpegps_read_packet(AVFormatContext *s,
     } else if (startcode >= 0x80 && startcode <= 0x9f) {
         type = CODEC_TYPE_AUDIO;
         codec_id = CODEC_ID_AC3;
+    } else if (startcode >= 0xa0 && startcode <= 0xbf) {
+        type = CODEC_TYPE_AUDIO;
+        codec_id = CODEC_ID_PCM_S16BE;
     } else {
     skip:
         /* skip packet */
@@ -611,6 +614,23 @@ static int mpegps_read_packet(AVFormatContext *s,
     st->codec.codec_type = type;
     st->codec.codec_id = codec_id;
  found:
+    if (startcode >= 0xa0 && startcode <= 0xbf) {
+        int b1, freq;
+        static const int lpcm_freq_tab[4] = { 48000, 96000, 44100, 32000 };
+
+        /* for LPCM, we just skip the header and consider it is raw
+           audio data */
+        if (len <= 3)
+            goto skip;
+        get_byte(&s->pb); /* emphasis (1), muse(1), reserved(1), frame number(5) */
+        b1 = get_byte(&s->pb); /* quant (2), freq(2), reserved(1), channels(3) */
+        get_byte(&s->pb); /* dynamic range control (0x80 = off) */
+        len -= 3;
+        freq = (b1 >> 4) & 3;
+        st->codec.sample_rate = lpcm_freq_tab[freq];
+        st->codec.channels = 1 + (b1 & 7);
+        st->codec.bit_rate = st->codec.channels * st->codec.sample_rate * 2;
+    }
     av_new_packet(pkt, len);
     //printf("\nRead Packet ID: %x PTS: %f Size: %d", startcode,
     //       (float)pts/90000, len);