Mercurial > libavformat.hg
diff aiff.c @ 3908:1d3d17de20ba libavformat
Bump Major version, this commit is almost just renaming bits_per_sample to
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!
author | michael |
---|---|
date | Mon, 08 Sep 2008 14:24:59 +0000 |
parents | fbe8704f513a |
children | 4fd67f05bad9 |
line wrap: on
line diff
--- a/aiff.c Mon Sep 08 00:58:24 2008 +0000 +++ b/aiff.c Mon Sep 08 14:24:59 2008 +0000 @@ -112,7 +112,7 @@ 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 @@ 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 @@ 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 @@ 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 @@ 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));