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* WAV encoder and decoder
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* WAV muxer and demuxer
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* Copyright (c) 2001, 2002 Fabrice Bellard.
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* This library is free software; you can redistribute it and/or
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* This file is part of FFmpeg.
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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* version 2.1 of the License, or (at your option) any later version.
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* This library is distributed in the hope that it will be useful,
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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#include "avformat.h"
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const CodecTag codec_wav_tags[] = {
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{ CODEC_ID_MP2, 0x50 },
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{ CODEC_ID_MP3, 0x55 },
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{ CODEC_ID_AC3, 0x2000 },
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{ CODEC_ID_PCM_S16LE, 0x01 },
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{ CODEC_ID_PCM_U8, 0x01 }, /* must come after s16le in this list */
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{ CODEC_ID_PCM_ALAW, 0x06 },
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{ CODEC_ID_PCM_MULAW, 0x07 },
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{ CODEC_ID_ADPCM_MS, 0x02 },
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{ CODEC_ID_ADPCM_IMA_WAV, 0x11 },
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{ CODEC_ID_ADPCM_IMA_DK4, 0x61 }, /* rogue format number */
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{ CODEC_ID_ADPCM_IMA_DK3, 0x62 }, /* rogue format number */
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{ CODEC_ID_WMAV1, 0x160 },
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{ CODEC_ID_WMAV2, 0x161 },
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/* WAVEFORMATEX header */
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/* returns the size or -1 on error */
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int put_wav_header(ByteIOContext *pb, AVCodecContext *enc)
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int bps, blkalign, bytespersec;
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enc->codec_tag = codec_get_tag(codec_wav_tags, enc->codec_id);
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put_le16(pb, enc->codec_tag);
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put_le16(pb, enc->channels);
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put_le32(pb, enc->sample_rate);
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if (enc->codec_id == CODEC_ID_PCM_U8 ||
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enc->codec_id == CODEC_ID_PCM_ALAW ||
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enc->codec_id == CODEC_ID_PCM_MULAW) {
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} else if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3) {
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} else if (enc->codec_id == CODEC_ID_ADPCM_IMA_WAV || enc->codec_id == CODEC_ID_ADPCM_MS) {
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if (enc->codec_id == CODEC_ID_MP2 || enc->codec_id == CODEC_ID_MP3) {
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//blkalign = 144 * enc->bit_rate/enc->sample_rate;
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} else if (enc->block_align != 0) { /* specified by the codec */
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blkalign = enc->block_align;
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blkalign = enc->channels*bps >> 3;
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if (enc->codec_id == CODEC_ID_PCM_U8 ||
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enc->codec_id == CODEC_ID_PCM_S16LE) {
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bytespersec = enc->sample_rate * blkalign;
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bytespersec = enc->bit_rate / 8;
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put_le32(pb, bytespersec); /* bytes per second */
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put_le16(pb, blkalign); /* block align */
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put_le16(pb, bps); /* bits per sample */
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if (enc->codec_id == CODEC_ID_MP3) {
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put_le16(pb, 12); /* wav_extra_size */
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put_le16(pb, 1); /* wID */
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put_le32(pb, 2); /* fdwFlags */
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put_le16(pb, 1152); /* nBlockSize */
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put_le16(pb, 1); /* nFramesPerBlock */
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put_le16(pb, 1393); /* nCodecDelay */
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} else if (enc->codec_id == CODEC_ID_MP2) {
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put_le16(pb, 22); /* wav_extra_size */
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put_le16(pb, 2); /* fwHeadLayer */
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put_le32(pb, enc->bit_rate); /* dwHeadBitrate */
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put_le16(pb, enc->channels == 2 ? 1 : 8); /* fwHeadMode */
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put_le16(pb, 0); /* fwHeadModeExt */
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put_le16(pb, 1); /* wHeadEmphasis */
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put_le16(pb, 16); /* fwHeadFlags */
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put_le32(pb, 0); /* dwPTSLow */
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put_le32(pb, 0); /* dwPTSHigh */
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} else if (enc->codec_id == CODEC_ID_ADPCM_IMA_WAV) {
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put_le16(pb, 2); /* wav_extra_size */
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put_le16(pb, ((enc->block_align - 4 * enc->channels) / (4 * enc->channels)) * 8 + 1); /* wSamplesPerBlock */
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put_le16(pb, 0); /* wav_extra_size */
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/* We could be given one of the three possible structures here:
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* WAVEFORMAT, PCMWAVEFORMAT or WAVEFORMATEX. Each structure
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* is an expansion of the previous one with the fields added
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* at the bottom. PCMWAVEFORMAT adds 'WORD wBitsPerSample' and
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* WAVEFORMATEX adds 'WORD cbSize' and basically makes itself
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* an openended structure.
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void get_wav_header(ByteIOContext *pb, AVCodecContext *codec, int size)
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codec->codec_type = CODEC_TYPE_AUDIO;
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codec->codec_tag = id;
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codec->channels = get_le16(pb);
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codec->sample_rate = get_le32(pb);
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codec->bit_rate = get_le32(pb) * 8;
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codec->block_align = get_le16(pb);
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if (size == 14) { /* We're dealing with plain vanilla WAVEFORMAT */
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codec->bits_per_sample = 8;
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codec->bits_per_sample = get_le16(pb);
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codec->codec_id = wav_codec_get_id(id, codec->bits_per_sample);
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if (size > 16) { /* We're obviously dealing with WAVEFORMATEX */
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codec->extradata_size = get_le16(pb);
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if (codec->extradata_size > 0) {
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if (codec->extradata_size > size - 18)
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codec->extradata_size = size - 18;
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codec->extradata = av_mallocz(codec->extradata_size);
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get_buffer(pb, codec->extradata, codec->extradata_size);
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codec->extradata_size = 0;
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/* It is possible for the chunk to contain garbage at the end */
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if (size - codec->extradata_size - 18 > 0)
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url_fskip(pb, size - codec->extradata_size - 18);
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int wav_codec_get_id(unsigned int tag, int bps)
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id = codec_get_id(codec_wav_tags, tag);
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/* handle specific u8 codec */
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if (id == CODEC_ID_PCM_S16LE && bps == 8)
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id = CODEC_ID_PCM_U8;
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#include "allformats.h"
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static int wav_write_header(AVFormatContext *s)
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WAVContext *wav = s->priv_data;
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ByteIOContext *pb = &s->pb;
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put_tag(pb, "RIFF");
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put_le32(pb, 0); /* file length */
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/* format header */
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fmt = start_tag(pb, "fmt ");
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if (put_wav_header(pb, &s->streams[0]->codec) < 0) {
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if (put_wav_header(pb, s->streams[0]->codec) < 0) {
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if(s->streams[0]->codec->codec_tag != 0x01 /* hence for all other than PCM */
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&& !url_is_streamed(&s->pb)) {
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fact = start_tag(pb, "fact");
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av_set_pts_info(s->streams[0], 64, 1, s->streams[0]->codec->sample_rate);
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wav->maxpts = wav->last_duration = 0;
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wav->minpts = INT64_MAX;
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wav->data = start_tag(pb, "data");
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put_flush_packet(pb);
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static int wav_write_packet(AVFormatContext *s, int stream_index_ptr,
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const uint8_t *buf, int size, int64_t pts)
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static int wav_write_packet(AVFormatContext *s, AVPacket *pkt)
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ByteIOContext *pb = &s->pb;
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put_buffer(pb, buf, size);
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WAVContext *wav = s->priv_data;
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put_buffer(pb, pkt->data, pkt->size);
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if(pkt->pts != AV_NOPTS_VALUE) {
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wav->minpts = FFMIN(wav->minpts, pkt->pts);
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wav->maxpts = FFMAX(wav->maxpts, pkt->pts);
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wav->last_duration = pkt->duration;
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av_log(s, AV_LOG_ERROR, "wav_write_packet: NOPTS\n");