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Bump Major version, this commit is almost just renaming bits_per_sample to

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bits_per_coded_sample but that cannot be done seperately.
Patch by Luca Abeni
Also reset the minor version and fix the forgotton change to libfaad.
Note: The API/ABI should not be considered stable yet, there still may
be a change done here or there if some developer has some cleanup ideas and
patches!

Originally committed as revision 15262 to svn://svn.ffmpeg.org/ffmpeg/trunk
This commit is contained in:
Luca Abeni
2008-09-08 14:24:59 +00:00
committed by Michael Niedermayer
parent 71375e0500
commit dd1c8f3e6e
71 changed files with 206 additions and 207 deletions
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18
libavformat/rpl.c
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@@ -142,7 +142,7 @@ static int rpl_read_header(AVFormatContext *s, AVFormatParameters *ap)
vst->codec->codec_tag = read_line_and_int(pb, &error); // video format
vst->codec->width = read_line_and_int(pb, &error); // video width
vst->codec->height = read_line_and_int(pb, &error); // video height
vst->codec->bits_per_sample = read_line_and_int(pb, &error); // video bits per sample
vst->codec->bits_per_coded_sample = read_line_and_int(pb, &error); // video bits per sample
error |= read_line(pb, line, sizeof(line)); // video frames per second
fps = read_fps(line, &error);
av_set_pts_info(vst, 32, fps.den, fps.num);
@@ -157,7 +157,7 @@ static int rpl_read_header(AVFormatContext *s, AVFormatParameters *ap)
case 124:
vst->codec->codec_id = CODEC_ID_ESCAPE124;
// The header is wrong here, at least sometimes
vst->codec->bits_per_sample = 16;
vst->codec->bits_per_coded_sample = 16;
break;
#if 0
case 130:
@@ -184,20 +184,20 @@ static int rpl_read_header(AVFormatContext *s, AVFormatParameters *ap)
ast->codec->codec_tag = audio_format;
ast->codec->sample_rate = read_line_and_int(pb, &error); // audio bitrate
ast->codec->channels = read_line_and_int(pb, &error); // number of audio channels
ast->codec->bits_per_sample = read_line_and_int(pb, &error); // audio bits per sample
ast->codec->bits_per_coded_sample = read_line_and_int(pb, &error); // audio bits per sample
// At least one sample uses 0 for ADPCM, which is really 4 bits
// per sample.
if (ast->codec->bits_per_sample == 0)
ast->codec->bits_per_sample = 4;
if (ast->codec->bits_per_coded_sample == 0)
ast->codec->bits_per_coded_sample = 4;
ast->codec->bit_rate = ast->codec->sample_rate *
ast->codec->bits_per_sample *
ast->codec->bits_per_coded_sample *
ast->codec->channels;
ast->codec->codec_id = CODEC_ID_NONE;
switch (audio_format) {
case 1:
if (ast->codec->bits_per_sample == 16) {
if (ast->codec->bits_per_coded_sample == 16) {
// 16-bit audio is always signed
ast->codec->codec_id = CODEC_ID_PCM_S16LE;
break;
@@ -206,12 +206,12 @@ static int rpl_read_header(AVFormatContext *s, AVFormatParameters *ap)
// samples needed.
break;
case 101:
if (ast->codec->bits_per_sample == 8) {
if (ast->codec->bits_per_coded_sample == 8) {
// The samples with this kind of audio that I have
// are all unsigned.
ast->codec->codec_id = CODEC_ID_PCM_U8;
break;
} else if (ast->codec->bits_per_sample == 4) {
} else if (ast->codec->bits_per_coded_sample == 4) {
ast->codec->codec_id = CODEC_ID_ADPCM_IMA_EA_SEAD;
break;
}