ffmpeg/libavformat/aiffenc.c
Michael Niedermayer 9d76cf0b18 Merge remote-tracking branch 'qatar/master'
* qatar/master:
  rtpdec: Templatize the code for different g726 bitrate variants
  rv40: move loop filter to rv34dsp context
  lavf: make av_set_pts_info private.
  rtpdec: Add support for G726 audio
  rtpdec: Add an init function that can do custom codec context initialization
  avconv: make copy_tb on by default.
  matroskadec: don't set codec timebase.
  rmdec: don't set codec timebase.
  avconv: compute next_pts from input packet duration when possible.
  lavf: estimate frame duration from r_frame_rate.
  avconv: update InputStream.pts in the streamcopy case.

Conflicts:
	avconv.c
	libavdevice/alsa-audio-dec.c
	libavdevice/bktr.c
	libavdevice/fbdev.c
	libavdevice/libdc1394.c
	libavdevice/oss_audio.c
	libavdevice/v4l.c
	libavdevice/v4l2.c
	libavdevice/vfwcap.c
	libavdevice/x11grab.c
	libavformat/au.c
	libavformat/eacdata.c
	libavformat/flvdec.c
	libavformat/mpegts.c
	libavformat/mxfenc.c
	libavformat/rtpdec_g726.c
	libavformat/wtv.c
	libavformat/xmv.c

Merged-by: Michael Niedermayer <michaelni@gmx.at>
2011-12-01 02:54:24 +01:00

171 lines
5.0 KiB
C

/*
* AIFF/AIFF-C muxer
* Copyright (c) 2006 Patrick Guimond
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "libavutil/intfloat_readwrite.h"
#include "avformat.h"
#include "internal.h"
#include "aiff.h"
#include "avio_internal.h"
#include "isom.h"
typedef struct {
int64_t form;
int64_t frames;
int64_t ssnd;
} AIFFOutputContext;
static int aiff_write_header(AVFormatContext *s)
{
AIFFOutputContext *aiff = s->priv_data;
AVIOContext *pb = s->pb;
AVCodecContext *enc = s->streams[0]->codec;
AVExtFloat sample_rate;
int aifc = 0;
/* First verify if format is ok */
if (!enc->codec_tag)
return -1;
if (enc->codec_tag != MKTAG('N','O','N','E'))
aifc = 1;
/* FORM AIFF header */
ffio_wfourcc(pb, "FORM");
aiff->form = avio_tell(pb);
avio_wb32(pb, 0); /* file length */
ffio_wfourcc(pb, aifc ? "AIFC" : "AIFF");
if (aifc) { // compressed audio
enc->bits_per_coded_sample = 16;
if (!enc->block_align) {
av_log(s, AV_LOG_ERROR, "block align not set\n");
return -1;
}
/* Version chunk */
ffio_wfourcc(pb, "FVER");
avio_wb32(pb, 4);
avio_wb32(pb, 0xA2805140);
}
if (enc->channels > 2 && enc->channel_layout) {
ffio_wfourcc(pb, "CHAN");
avio_wb32(pb, 12);
ff_mov_write_chan(pb, enc->channel_layout);
}
/* Common chunk */
ffio_wfourcc(pb, "COMM");
avio_wb32(pb, aifc ? 24 : 18); /* size */
avio_wb16(pb, enc->channels); /* Number of channels */
aiff->frames = avio_tell(pb);
avio_wb32(pb, 0); /* Number of frames */
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_coded_sample * enc->channels) >> 3;
avio_wb16(pb, enc->bits_per_coded_sample); /* Sample size */
sample_rate = av_dbl2ext((double)enc->sample_rate);
avio_write(pb, (uint8_t*)&sample_rate, sizeof(sample_rate));
if (aifc) {
avio_wl32(pb, enc->codec_tag);
avio_wb16(pb, 0);
}
/* Sound data chunk */
ffio_wfourcc(pb, "SSND");
aiff->ssnd = avio_tell(pb); /* Sound chunk size */
avio_wb32(pb, 0); /* Sound samples data size */
avio_wb32(pb, 0); /* Data offset */
avio_wb32(pb, 0); /* Block-size (block align) */
avpriv_set_pts_info(s->streams[0], 64, 1, s->streams[0]->codec->sample_rate);
/* Data is starting here */
avio_flush(pb);
return 0;
}
static int aiff_write_packet(AVFormatContext *s, AVPacket *pkt)
{
AVIOContext *pb = s->pb;
avio_write(pb, pkt->data, pkt->size);
return 0;
}
static int aiff_write_trailer(AVFormatContext *s)
{
AVIOContext *pb = s->pb;
AIFFOutputContext *aiff = s->priv_data;
AVCodecContext *enc = s->streams[0]->codec;
/* Chunks sizes must be even */
int64_t file_size, end_size;
end_size = file_size = avio_tell(pb);
if (file_size & 1) {
avio_w8(pb, 0);
end_size++;
}
if (s->pb->seekable) {
/* File length */
avio_seek(pb, aiff->form, SEEK_SET);
avio_wb32(pb, file_size - aiff->form - 4);
/* Number of sample frames */
avio_seek(pb, aiff->frames, SEEK_SET);
avio_wb32(pb, (file_size-aiff->ssnd-12)/enc->block_align);
/* Sound Data chunk size */
avio_seek(pb, aiff->ssnd, SEEK_SET);
avio_wb32(pb, file_size - aiff->ssnd - 4);
/* return to the end */
avio_seek(pb, end_size, SEEK_SET);
avio_flush(pb);
}
return 0;
}
AVOutputFormat ff_aiff_muxer = {
.name = "aiff",
.long_name = NULL_IF_CONFIG_SMALL("Audio IFF"),
.mime_type = "audio/aiff",
.extensions = "aif,aiff,afc,aifc",
.priv_data_size = sizeof(AIFFOutputContext),
.audio_codec = CODEC_ID_PCM_S16BE,
.video_codec = CODEC_ID_NONE,
.write_header = aiff_write_header,
.write_packet = aiff_write_packet,
.write_trailer = aiff_write_trailer,
.codec_tag= (const AVCodecTag* const []){ff_codec_aiff_tags, 0},
};