From 6ec688e1bc76dd93151cbca1c340162ae4b10d77 Mon Sep 17 00:00:00 2001
From: nu774 <honeycomb77@gmail.com>
Date: Fri, 9 May 2014 23:05:42 +0900
Subject: [PATCH] mp3: enable packed main_data decoding in MP4
14496-3 suggests packing main_data of MP3 that is usually scattered
into multiple frames due to bit reservoir.
However, after packing main_data into a access unit, bitrate index
in the MPEG audio frame header doesn't match with actual frame size.
In order to accept this, this patch removes unnecessary frame size
checking on mp3 decoder.
Also, mov demuxer was changed to use MP3 parser only on special cases
(QT MOV with specific sample description) to avoid re-packetizing.
Signed-off-by: Luca Barbato <lu_zero@gentoo.org>
---
libavcodec/mpegaudiodec_template.c | 7 -------
libavformat/mov.c | 10 +++++++++-
tests/fate/mp3.mak | 2 +-
3 files changed, 10 insertions(+), 9 deletions(-)
diff --git a/libavcodec/mpegaudiodec_template.c b/libavcodec/mpegaudiodec_template.c
index 08dd18bf2ef..293316b9441 100644
--- a/libavcodec/mpegaudiodec_template.c
+++ b/libavcodec/mpegaudiodec_template.c
@@ -1647,13 +1647,6 @@ static int decode_frame(AVCodecContext * avctx, void *data, int *got_frame_ptr,
if (!avctx->bit_rate)
avctx->bit_rate = s->bit_rate;
- if (s->frame_size <= 0 || s->frame_size > buf_size) {
- av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
- return AVERROR_INVALIDDATA;
- } else if (s->frame_size < buf_size) {
- buf_size= s->frame_size;
- }
-
s->frame = data;
ret = mp_decode_frame(s, NULL, buf, buf_size);
diff --git a/libavformat/mov.c b/libavformat/mov.c
index f603446d98b..b922579ac0a 100644
--- a/libavformat/mov.c
+++ b/libavformat/mov.c
@@ -1519,6 +1519,15 @@ static void mov_parse_stsd_audio(MOVContext *c, AVIOContext *pb,
ff_mov_get_lpcm_codec_id(st->codec->bits_per_coded_sample,
flags);
}
+ if (version == 0 || (version == 1 && sc->audio_cid != -2)) {
+ /* can't correctly handle variable sized packet as audio unit */
+ switch (st->codec->codec_id) {
+ case AV_CODEC_ID_MP2:
+ case AV_CODEC_ID_MP3:
+ st->need_parsing = AVSTREAM_PARSE_FULL;
+ break;
+ }
+ }
}
switch (st->codec->codec_id) {
@@ -1695,7 +1704,6 @@ static int mov_finalize_stsd_codec(MOVContext *c, AVIOContext *pb,
case AV_CODEC_ID_MP3:
/* force type after stsd for m1a hdlr */
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
- st->need_parsing = AVSTREAM_PARSE_FULL;
break;
case AV_CODEC_ID_GSM:
case AV_CODEC_ID_ADPCM_MS:
diff --git a/tests/fate/mp3.mak b/tests/fate/mp3.mak
index fe6a0e12eb5..3442ce15b93 100644
--- a/tests/fate/mp3.mak
+++ b/tests/fate/mp3.mak
@@ -1,6 +1,6 @@
FATE_MP3 += fate-mp3-float-conf-compl
fate-mp3-float-conf-compl: CMD = pcm -acodec mp3float -i $(TARGET_SAMPLES)/mp3-conformance/compl.bit
-fate-mp3-float-conf-compl: REF = $(SAMPLES)/mp3-conformance/compl.pcm
+fate-mp3-float-conf-compl: REF = $(SAMPLES)/mp3-conformance/compl_2.pcm
FATE_MP3 += fate-mp3-float-conf-he_32khz
fate-mp3-float-conf-he_32khz: CMD = pcm -acodec mp3float -i $(TARGET_SAMPLES)/mp3-conformance/he_32khz.bit -fs 343296
--
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