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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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24
libavformat/aiff.c
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@@ -112,7 +112,7 @@ static unsigned int get_aiff_header(ByteIOContext *pb, AVCodecContext *codec,
codec->codec_type = CODEC_TYPE_AUDIO;
codec->channels = get_be16(pb);
num_frames = get_be32(pb);
codec->bits_per_sample = get_be16(pb);
codec->bits_per_coded_sample = get_be16(pb);
get_buffer(pb, (uint8_t*)&ext, sizeof(ext));/* Sample rate is in */
sample_rate = av_ext2dbl(ext); /* 80 bits BE IEEE extended float */
@@ -126,8 +126,8 @@ static unsigned int get_aiff_header(ByteIOContext *pb, AVCodecContext *codec,
switch (codec->codec_id) {
case CODEC_ID_PCM_S16BE:
codec->codec_id = aiff_codec_get_id(codec->bits_per_sample);
codec->bits_per_sample = av_get_bits_per_sample(codec->codec_id);
codec->codec_id = aiff_codec_get_id(codec->bits_per_coded_sample);
codec->bits_per_coded_sample = av_get_bits_per_sample(codec->codec_id);
break;
case CODEC_ID_ADPCM_IMA_QT:
codec->block_align = 34*codec->channels;
@@ -151,14 +151,14 @@ static unsigned int get_aiff_header(ByteIOContext *pb, AVCodecContext *codec,
size -= 4;
} else {
/* Need the codec type */
codec->codec_id = aiff_codec_get_id(codec->bits_per_sample);
codec->bits_per_sample = av_get_bits_per_sample(codec->codec_id);
codec->codec_id = aiff_codec_get_id(codec->bits_per_coded_sample);
codec->bits_per_coded_sample = av_get_bits_per_sample(codec->codec_id);
}
/* Block align needs to be computed in all cases, as the definition
* is specific to applications -> here we use the WAVE format definition */
if (!codec->block_align)
codec->block_align = (codec->bits_per_sample * codec->channels) >> 3;
codec->block_align = (codec->bits_per_coded_sample * codec->channels) >> 3;
codec->bit_rate = (codec->frame_size ? codec->sample_rate/codec->frame_size :
codec->sample_rate) * (codec->block_align << 3);
@@ -198,7 +198,7 @@ static int aiff_write_header(AVFormatContext *s)
put_tag(pb, aifc ? "AIFC" : "AIFF");
if (aifc) { // compressed audio
enc->bits_per_sample = 16;
enc->bits_per_coded_sample = 16;
if (!enc->block_align) {
av_log(s, AV_LOG_ERROR, "block align not set\n");
return -1;
@@ -217,16 +217,16 @@ static int aiff_write_header(AVFormatContext *s)
aiff->frames = url_ftell(pb);
put_be32(pb, 0); /* Number of frames */
if (!enc->bits_per_sample)
enc->bits_per_sample = av_get_bits_per_sample(enc->codec_id);
if (!enc->bits_per_sample) {
if (!enc->bits_per_coded_sample)
enc->bits_per_coded_sample = av_get_bits_per_sample(enc->codec_id);
if (!enc->bits_per_coded_sample) {
av_log(s, AV_LOG_ERROR, "could not compute bits per sample\n");
return -1;
}
if (!enc->block_align)
enc->block_align = (enc->bits_per_sample * enc->channels) >> 3;
enc->block_align = (enc->bits_per_coded_sample * enc->channels) >> 3;
put_be16(pb, enc->bits_per_sample); /* Sample size */
put_be16(pb, enc->bits_per_coded_sample); /* Sample size */
sample_rate = av_dbl2ext((double)enc->sample_rate);
put_buffer(pb, (uint8_t*)&sample_rate, sizeof(sample_rate));