@@ -51,8 +51,6 @@ typedef struct SmackerContext {
uint32_t flags;
uint32_t audio[7];
uint32_t treesize;
- uint8_t aflags[7];
- uint32_t rates[7];
uint32_t pad;
/* frame info */
uint32_t *frm_size;
@@ -107,7 +105,7 @@ static int smacker_read_header(AVFormatContext *s)
{
AVIOContext *pb = s->pb;
SmackerContext *smk = s->priv_data;
- AVStream *st, *ast[7];
+ AVStream *st;
int i, ret;
int tbase;
@@ -153,9 +151,45 @@ static int smacker_read_header(AVFormatContext *s)
if ((ret = ffio_read_size(pb, st->codecpar->extradata, 16)) < 0)
return ret;
+ /* handle possible audio streams */
for(i = 0; i < 7; i++) {
- smk->rates[i] = avio_rl24(pb);
- smk->aflags[i] = avio_r8(pb);
+ uint32_t rate = avio_rl24(pb);
+ uint8_t aflag = avio_r8(pb);
+
+ smk->indexes[i] = -1;
+
+ if (rate) {
+ AVStream *ast = avformat_new_stream(s, NULL);
+ if (!ast)
+ return AVERROR(ENOMEM);
+
+ smk->indexes[i] = ast->index;
+ ast->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
+ if (aflag & SMK_AUD_BINKAUD) {
+ ast->codecpar->codec_id = AV_CODEC_ID_BINKAUDIO_RDFT;
+ } else if (aflag & SMK_AUD_USEDCT) {
+ ast->codecpar->codec_id = AV_CODEC_ID_BINKAUDIO_DCT;
+ } else if (aflag & SMK_AUD_PACKED) {
+ ast->codecpar->codec_id = AV_CODEC_ID_SMACKAUDIO;
+ ast->codecpar->codec_tag = MKTAG('S', 'M', 'K', 'A');
+ } else {
+ ast->codecpar->codec_id = AV_CODEC_ID_PCM_U8;
+ }
+ if (aflag & SMK_AUD_STEREO) {
+ ast->codecpar->channels = 2;
+ ast->codecpar->channel_layout = AV_CH_LAYOUT_STEREO;
+ } else {
+ ast->codecpar->channels = 1;
+ ast->codecpar->channel_layout = AV_CH_LAYOUT_MONO;
+ }
+ ast->codecpar->sample_rate = rate;
+ ast->codecpar->bits_per_coded_sample = (aflag & SMK_AUD_16BITS) ? 16 : 8;
+ if (ast->codecpar->bits_per_coded_sample == 16 &&
+ ast->codecpar->codec_id == AV_CODEC_ID_PCM_U8)
+ ast->codecpar->codec_id = AV_CODEC_ID_PCM_S16LE;
+ avpriv_set_pts_info(ast, 64, 1, ast->codecpar->sample_rate
+ * ast->codecpar->channels * ast->codecpar->bits_per_coded_sample / 8);
+ }
}
smk->pad = avio_rl32(pb);
/* setup data */
@@ -194,44 +228,6 @@ static int smacker_read_header(AVFormatContext *s)
av_reduce(&tbase, &smk->pts_inc, tbase, smk->pts_inc, (1UL<<31)-1);
avpriv_set_pts_info(st, 33, smk->pts_inc, tbase);
st->duration = smk->frames;
- /* handle possible audio streams */
- for(i = 0; i < 7; i++) {
- smk->indexes[i] = -1;
- if (smk->rates[i]) {
- ast[i] = avformat_new_stream(s, NULL);
- if (!ast[i]) {
- av_freep(&smk->frm_size);
- av_freep(&smk->frm_flags);
- return AVERROR(ENOMEM);
- }
- smk->indexes[i] = ast[i]->index;
- ast[i]->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
- if (smk->aflags[i] & SMK_AUD_BINKAUD) {
- ast[i]->codecpar->codec_id = AV_CODEC_ID_BINKAUDIO_RDFT;
- } else if (smk->aflags[i] & SMK_AUD_USEDCT) {
- ast[i]->codecpar->codec_id = AV_CODEC_ID_BINKAUDIO_DCT;
- } else if (smk->aflags[i] & SMK_AUD_PACKED){
- ast[i]->codecpar->codec_id = AV_CODEC_ID_SMACKAUDIO;
- ast[i]->codecpar->codec_tag = MKTAG('S', 'M', 'K', 'A');
- } else {
- ast[i]->codecpar->codec_id = AV_CODEC_ID_PCM_U8;
- }
- if (smk->aflags[i] & SMK_AUD_STEREO) {
- ast[i]->codecpar->channels = 2;
- ast[i]->codecpar->channel_layout = AV_CH_LAYOUT_STEREO;
- } else {
- ast[i]->codecpar->channels = 1;
- ast[i]->codecpar->channel_layout = AV_CH_LAYOUT_MONO;
- }
- ast[i]->codecpar->sample_rate = smk->rates[i];
- ast[i]->codecpar->bits_per_coded_sample = (smk->aflags[i] & SMK_AUD_16BITS) ? 16 : 8;
- if(ast[i]->codecpar->bits_per_coded_sample == 16 && ast[i]->codecpar->codec_id == AV_CODEC_ID_PCM_U8)
- ast[i]->codecpar->codec_id = AV_CODEC_ID_PCM_S16LE;
- avpriv_set_pts_info(ast[i], 64, 1, ast[i]->codecpar->sample_rate
- * ast[i]->codecpar->channels * ast[i]->codecpar->bits_per_coded_sample / 8);
- }
- }
-
/* load trees to extradata, they will be unpacked by decoder */
ret = avio_read(pb, st->codecpar->extradata + 16, st->codecpar->extradata_size - 16);
The Smacker demuxer currently parses several fields that indicate how many audio streams a file contains. This data is parsed and stored into arrays in the demuxer's context and although the data is used only to initialize the audio streams, it is kept for the whole lifetime of the demuxer. This has been changed: The data is used directly to create the audio streams and no longer kept at all. This also simplifies error handling in case adding a new stream fails: Several arrays which until now have been allocated between parsing the data determining how many audio streams to create and actually creating them would need to be freed in this case. Now the streams are created first, so freeing is no longer an issue. Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com> --- libavformat/smacker.c | 82 ++++++++++++++++++++----------------------- 1 file changed, 39 insertions(+), 43 deletions(-)