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mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00

lavf/swfdec: factorize the creation of a new stream.

This also makes the changes of a3949fe11 applicable in both cases.
This commit is contained in:
Clément Bœsch 2013-02-20 21:40:52 +01:00
parent da7672b20a
commit 9a0076f50c

View File

@ -138,6 +138,29 @@ static int swf_read_header(AVFormatContext *s)
return 0;
}
static AVStream *create_new_audio_stream(AVFormatContext *s, int id, int info)
{
int sample_rate_code;
AVStream *ast = avformat_new_stream(s, NULL);
if (!ast)
return NULL;
ast->id = id;
if (info & 1) {
ast->codec->channels = 2;
ast->codec->channel_layout = AV_CH_LAYOUT_STEREO;
} else {
ast->codec->channels = 1;
ast->codec->channel_layout = AV_CH_LAYOUT_MONO;
}
ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
ast->codec->codec_id = ff_codec_get_id(swf_audio_codec_tags, info>>4 & 15);
ast->need_parsing = AVSTREAM_PARSE_FULL;
sample_rate_code = info>>2 & 3;
ast->codec->sample_rate = 44100 >> (3 - sample_rate_code);
avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate);
return ast;
}
static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
{
SWFContext *swf = s->priv_data;
@ -184,7 +207,6 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
len -= 8;
} else if (tag == TAG_STREAMHEAD || tag == TAG_STREAMHEAD2) {
/* streaming found */
int sample_rate_code;
for (i=0; i<s->nb_streams; i++) {
st = s->streams[i];
@ -195,27 +217,12 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
avio_r8(pb);
v = avio_r8(pb);
swf->samples_per_frame = avio_rl16(pb);
ast = avformat_new_stream(s, NULL);
ast = create_new_audio_stream(s, -1, v); /* -1 to avoid clash with video stream ch_id */
if (!ast)
return AVERROR(ENOMEM);
ast->id = -1; /* -1 to avoid clash with video stream ch_id */
if (v & 1) {
ast->codec->channels = 2;
ast->codec->channel_layout = AV_CH_LAYOUT_STEREO;
} else {
ast->codec->channels = 1;
ast->codec->channel_layout = AV_CH_LAYOUT_MONO;
}
ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
ast->codec->codec_id = ff_codec_get_id(swf_audio_codec_tags, (v>>4) & 15);
ast->need_parsing = AVSTREAM_PARSE_FULL;
sample_rate_code= (v>>2) & 3;
ast->codec->sample_rate = 44100 >> (3 - sample_rate_code);
avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate);
len -= 4;
} else if (tag == TAG_DEFINESOUND) {
/* audio stream */
int sample_rate_code;
int ch_id = avio_rl16(pb);
for (i=0; i<s->nb_streams; i++) {
@ -229,17 +236,9 @@ static int swf_read_packet(AVFormatContext *s, AVPacket *pkt)
// these are smaller audio streams in DEFINESOUND tags, but it's technically
// possible they could be huge. Break it up into multiple packets if it's big.
v = avio_r8(pb);
ast = avformat_new_stream(s, NULL);
ast = create_new_audio_stream(s, ch_id, v);
if (!ast)
return AVERROR(ENOMEM);
ast->id = ch_id;
ast->codec->channels = 1 + (v&1);
ast->codec->codec_type = AVMEDIA_TYPE_AUDIO;
ast->codec->codec_id = ff_codec_get_id(swf_audio_codec_tags, (v>>4) & 15);
ast->need_parsing = AVSTREAM_PARSE_FULL;
sample_rate_code= (v>>2) & 3;
ast->codec->sample_rate = 44100 >> (3 - sample_rate_code);
avpriv_set_pts_info(ast, 64, 1, ast->codec->sample_rate);
ast->duration = avio_rl32(pb); // number of samples
if (((v>>4) & 15) == 2) { // MP3 sound data record
ast->skip_samples = avio_rl16(pb);