Re: [libav-devel] [libav-commits] r3d: do not create the audio stream until we know the sample rate

[Martin Storsjö]

On Sat, 12 Dec 2015, Anton Khirnov  wrote:

Module: libav
Branch: master
Commit: 6bf4c1d71199b92894f24db6386ed5070e590a16
Author:    Anton Khirnov <an...@khirnov.net>
Committer: Anton Khirnov <an...@khirnov.net>
Date:      Wed Oct 21 11:22:13 2015 +0200

r3d: do not create the audio stream until we know the sample rate

---

libavformat/r3d.c |   37 +++++++++++++++++++++++++------------
1 file changed, 25 insertions(+), 12 deletions(-)

This broke fate-redcode-demux on a number of configurations

diff --git a/libavformat/r3d.c b/libavformat/r3d.c
index c56bdc6..fb066a6 100644
--- a/libavformat/r3d.c
+++ b/libavformat/r3d.c
@@ -29,6 +29,8 @@ typedef struct R3DContext {
    unsigned video_offsets_count;
    unsigned *video_offsets;
    unsigned rdvo_offset;
+
+    int audio_channels;
} R3DContext;

typedef struct Atom {
@@ -52,6 +54,7 @@ static int read_atom(AVFormatContext *s, Atom *atom)
static int r3d_read_red1(AVFormatContext *s)
{
    AVStream *st = avformat_new_stream(s, NULL);
+    R3DContext *r3d = s->priv_data;
    char filename[258];
    int tmp;
    int av_unused tmp2;
@@ -89,17 +92,8 @@ static int r3d_read_red1(AVFormatContext *s)
        st->avg_frame_rate = framerate;
    }

-    tmp = avio_r8(s->pb); // audio channels
+    r3d->audio_channels = avio_r8(s->pb); // audio channels
    av_log(s, AV_LOG_TRACE, "audio channels %d\n", tmp);
-    if (tmp > 0) {
-        AVStream *ast = avformat_new_stream(s, NULL);
-        if (!ast)
-            return AVERROR(ENOMEM);
-        ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
-        ast->codec->codec_id = AV_CODEC_ID_PCM_S32BE;
-        ast->codec->channels = tmp;
-        avpriv_set_pts_info(ast, 32, 1, st->time_base.den);
-    }

    avio_read(s->pb, filename, 257);
    filename[sizeof(filename)-1] = 0;
@@ -183,6 +177,11 @@ static int r3d_read_header(AVFormatContext *s)
        return -1;
    }

+    /* we cannot create the audio stream now because we do not know the
+     * sample rate */
+    if (r3d->audio_channels)
+        s->ctx_flags |= AVFMTCTX_NOHEADER;
+
    s->internal->data_offset = avio_tell(s->pb);
    av_log(s, AV_LOG_TRACE, "data offset %#"PRIx64"\n", 
s->internal->data_offset);
    if (!s->pb->seekable)
@@ -271,13 +270,26 @@ static int r3d_read_redv(AVFormatContext *s, AVPacket 
*pkt, Atom *atom)

static int r3d_read_reda(AVFormatContext *s, AVPacket *pkt, Atom *atom)
{
-    AVStream *st = s->streams[1];
+    R3DContext *r3d = s->priv_data;
+    AVStream *st;
    int av_unused tmp, tmp2;
    int samples, size;
    int64_t pos = avio_tell(s->pb);
    unsigned dts;
    int ret;

+    if (s->nb_streams < 2) {
+        st = avformat_new_stream(s, NULL);
+        if (!st)
+            return AVERROR(ENOMEM);
+        st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
+        st->codec->codec_id = AV_CODEC_ID_PCM_S32BE;
+        st->codec->channels = r3d->audio_channels;
+        avpriv_set_pts_info(st, 32, 1, s->streams[0]->time_base.den);
+    } else {
+        st = s->streams[1];
+    }
+
    dts = avio_rb32(s->pb);

    st->codec->sample_rate = avio_rb32(s->pb);
@@ -321,6 +333,7 @@ static int r3d_read_reda(AVFormatContext *s, AVPacket *pkt, 
Atom *atom)

static int r3d_read_packet(AVFormatContext *s, AVPacket *pkt)
{
+    R3DContext *r3d = s->priv_data;
    Atom atom;
    int err = 0;

@@ -337,7 +350,7 @@ static int r3d_read_packet(AVFormatContext *s, AVPacket 
*pkt)
                return 0;
            break;
        case MKTAG('R','E','D','A'):
-            if (s->nb_streams < 2)
+            if (!r3d->audio_channels)
                return -1;
            if (s->streams[1]->discard == AVDISCARD_ALL)
                goto skip;

In fate-redcode-demux, this dereferences s->streams[1] even if 
s->nb_streams == 1.

// Martin
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