void ff_parse_specific_params(AVStream *st, int *au_rate,
int *au_ssize, int *au_scale)
{
AVCodecContext *codec = st->codec;
int gcd;
int audio_frame_size;
audio_frame_size = av_get_audio_frame_duration(codec, 0);
*au_ssize = codec->block_align;
if (audio_frame_size && codec->sample_rate) {
*au_scale = audio_frame_size;
*au_rate = codec->sample_rate;
} else if (codec->codec_type == AVMEDIA_TYPE_VIDEO ||
codec->codec_type == AVMEDIA_TYPE_DATA ||
codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
*au_scale = st->time_base.num;
*au_rate = st->time_base.den;
} else {
*au_scale = codec->block_align ? codec->block_align * 8 : 8;
*au_rate = codec->bit_rate ? codec->bit_rate :
8 * codec->sample_rate;
}
gcd = av_gcd(*au_scale, *au_rate);
*au_scale /= gcd;
*au_rate /= gcd;
}
void ff_parse_specific_params(AVStream *st, int *au_rate,
int *au_ssize, int *au_scale)
{
AVCodecContext *codec = st->codec;
int gcd;
int audio_frame_size;
/* We use the known constant frame size for the codec if known, otherwise
* fall back on using AVCodecContext.frame_size, which is not as reliable
* for indicating packet duration. */
audio_frame_size = av_get_audio_frame_duration(codec, 0);
if (!audio_frame_size)
audio_frame_size = codec->frame_size;
*au_ssize = codec->block_align;
if (audio_frame_size && codec->sample_rate) {
*au_scale = audio_frame_size;
*au_rate = codec->sample_rate;
} else if (codec->codec_type == AVMEDIA_TYPE_VIDEO ||
codec->codec_type == AVMEDIA_TYPE_DATA ||
codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
*au_scale = st->time_base.num;
*au_rate = st->time_base.den;
} else {
*au_scale = codec->block_align ? codec->block_align * 8 : 8;
*au_rate = codec->bit_rate ? codec->bit_rate :
8 * codec->sample_rate;
}
gcd = av_gcd(*au_scale, *au_rate);
*au_scale /= gcd;
*au_rate /= gcd;
}
static int rm_write_header(AVFormatContext *s)
{
RMMuxContext *rm = s->priv_data;
StreamInfo *stream;
int n;
AVCodecContext *codec;
if (s->nb_streams > 2) {
av_log(s, AV_LOG_ERROR, "At most 2 streams are currently supported for muxing in RM\n");
return AVERROR_PATCHWELCOME;
}
for(n=0;n<s->nb_streams;n++) {
AVStream *st = s->streams[n];
int frame_size;
s->streams[n]->id = n;
codec = s->streams[n]->codec;
stream = &rm->streams[n];
memset(stream, 0, sizeof(StreamInfo));
stream->num = n;
stream->bit_rate = codec->bit_rate;
stream->enc = codec;
switch(codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
rm->audio_stream = stream;
frame_size = av_get_audio_frame_duration(codec, 0);
stream->frame_rate = (AVRational){codec->sample_rate, frame_size};
/* XXX: dummy values */
stream->packet_max_size = 1024;
stream->nb_packets = 0;
stream->total_frames = stream->nb_packets;
break;
case AVMEDIA_TYPE_VIDEO:
rm->video_stream = stream;
// TODO: should be avg_frame_rate
stream->frame_rate = av_inv_q(st->time_base);
/* XXX: dummy values */
stream->packet_max_size = 4096;
stream->nb_packets = 0;
stream->total_frames = stream->nb_packets;
break;
default:
return -1;
}
}
if (rv10_write_header(s, 0, 0))
return AVERROR_INVALIDDATA;
avio_flush(s->pb);
return 0;
}
static int rm_write_header(AVFormatContext *s)
{
RMMuxContext *rm = s->priv_data;
StreamInfo *stream;
int n;
AVCodecContext *codec;
for(n=0;n<s->nb_streams;n++) {
AVStream *st = s->streams[n];
int frame_size;
s->streams[n]->id = n;
codec = s->streams[n]->codec;
stream = &rm->streams[n];
memset(stream, 0, sizeof(StreamInfo));
stream->num = n;
stream->bit_rate = codec->bit_rate;
stream->enc = codec;
switch(codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
rm->audio_stream = stream;
frame_size = av_get_audio_frame_duration(codec, 0);
stream->frame_rate = (float)codec->sample_rate / (float)frame_size;
/* XXX: dummy values */
stream->packet_max_size = 1024;
stream->nb_packets = 0;
stream->total_frames = stream->nb_packets;
break;
case AVMEDIA_TYPE_VIDEO:
rm->video_stream = stream;
// TODO: should be avg_frame_rate
stream->frame_rate = (float)st->time_base.den / (float)st->time_base.num;
/* XXX: dummy values */
stream->packet_max_size = 4096;
stream->nb_packets = 0;
stream->total_frames = stream->nb_packets;
break;
default:
return -1;
}
}
if (rv10_write_header(s, 0, 0))
return AVERROR_INVALIDDATA;
avio_flush(s->pb);
return 0;
}
static int adp_read_header(AVFormatContext *s)
{
AVStream *st;
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = AV_CODEC_ID_ADPCM_DTK;
st->codec->channel_layout = AV_CH_LAYOUT_STEREO;
st->codec->channels = 2;
st->codec->sample_rate = 48000;
st->start_time = 0;
if (s->pb->seekable)
st->duration = av_get_audio_frame_duration(st->codec, avio_size(s->pb));
avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
return 0;
}
static int msf_read_header(AVFormatContext *s)
{
unsigned codec, align, size;
AVStream *st;
avio_skip(s->pb, 4);
st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
codec = avio_rb32(s->pb);
st->codec->channels = avio_rb32(s->pb);
if (st->codec->channels <= 0)
return AVERROR_INVALIDDATA;
size = avio_rb32(s->pb);
st->codec->sample_rate = avio_rb32(s->pb);
if (st->codec->sample_rate <= 0)
return AVERROR_INVALIDDATA;
align = avio_rb32(s->pb) ;
if (align > INT_MAX / st->codec->channels)
return AVERROR_INVALIDDATA;
st->codec->block_align = align;
switch (codec) {
case 0: st->codec->codec_id = AV_CODEC_ID_PCM_S16BE; break;
case 3: st->codec->block_align = 16 * st->codec->channels;
st->codec->codec_id = AV_CODEC_ID_ADPCM_PSX; break;
case 7: st->need_parsing = AVSTREAM_PARSE_FULL_RAW;
st->codec->codec_id = AV_CODEC_ID_MP3; break;
default:
avpriv_request_sample(s, "Codec %d", codec);
return AVERROR_PATCHWELCOME;
}
st->duration = av_get_audio_frame_duration(st->codec, size);
avio_skip(s->pb, 0x40 - avio_tell(s->pb));
avpriv_set_pts_info(st, 64, 1, st->codec->sample_rate);
return 0;
}
static int swf_write_audio(AVFormatContext *s,
AVCodecContext *enc, uint8_t *buf, int size)
{
SWFContext *swf = s->priv_data;
/* Flash Player limit */
if (swf->swf_frame_number == 16000)
av_log(enc, AV_LOG_INFO, "warning: Flash Player limit of 16000 frames reached\n");
if (av_fifo_size(swf->audio_fifo) + size > AUDIO_FIFO_SIZE) {
av_log(s, AV_LOG_ERROR, "audio fifo too small to mux audio essence\n");
return -1;
}
av_fifo_generic_write(swf->audio_fifo, buf, size, NULL);
swf->sound_samples += av_get_audio_frame_duration(enc, size);
/* if audio only stream make sure we add swf frames */
if (!swf->video_enc)
swf_write_video(s, enc, 0, 0);
return 0;
}
/* Returns the number of sound data frames or negative on error */
static unsigned int get_aiff_header(AVFormatContext *s, int size,
unsigned version)
{
AVIOContext *pb = s->pb;
AVCodecContext *codec = s->streams[0]->codec;
AIFFInputContext *aiff = s->priv_data;
int exp;
uint64_t val;
double sample_rate;
unsigned int num_frames;
if (size & 1)
size++;
codec->codec_type = AVMEDIA_TYPE_AUDIO;
codec->channels = avio_rb16(pb);
num_frames = avio_rb32(pb);
codec->bits_per_coded_sample = avio_rb16(pb);
exp = avio_rb16(pb);
val = avio_rb64(pb);
sample_rate = ldexp(val, exp - 16383 - 63);
codec->sample_rate = sample_rate;
size -= 18;
/* get codec id for AIFF-C */
if (version == AIFF_C_VERSION1) {
codec->codec_tag = avio_rl32(pb);
codec->codec_id = ff_codec_get_id(ff_codec_aiff_tags, codec->codec_tag);
size -= 4;
}
if (version != AIFF_C_VERSION1 || codec->codec_id == AV_CODEC_ID_PCM_S16BE) {
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);
aiff->block_duration = 1;
} else {
switch (codec->codec_id) {
case AV_CODEC_ID_PCM_F32BE:
case AV_CODEC_ID_PCM_F64BE:
case AV_CODEC_ID_PCM_S16LE:
case AV_CODEC_ID_PCM_ALAW:
case AV_CODEC_ID_PCM_MULAW:
aiff->block_duration = 1;
break;
case AV_CODEC_ID_ADPCM_IMA_QT:
codec->block_align = 34*codec->channels;
break;
case AV_CODEC_ID_MACE3:
codec->block_align = 2*codec->channels;
break;
case AV_CODEC_ID_MACE6:
codec->block_align = 1*codec->channels;
break;
case AV_CODEC_ID_GSM:
codec->block_align = 33;
break;
default:
aiff->block_duration = 1;
break;
}
if (codec->block_align > 0)
aiff->block_duration = av_get_audio_frame_duration(codec,
codec->block_align);
}
/* 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 = (av_get_bits_per_sample(codec->codec_id) * codec->channels) >> 3;
if (aiff->block_duration) {
codec->bit_rate = codec->sample_rate * (codec->block_align << 3) /
aiff->block_duration;
}
/* Chunk is over */
if (size)
avio_skip(pb, size);
return num_frames;
}
static int rsd_read_header(AVFormatContext *s)
{
AVIOContext *pb = s->pb;
int i, version, start = 0x800;
AVCodecContext *codec;
AVStream *st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
avio_skip(pb, 3); // "RSD"
version = avio_r8(pb) - '0';
codec = st->codec;
codec->codec_type = AVMEDIA_TYPE_AUDIO;
codec->codec_tag = avio_rl32(pb);
codec->codec_id = ff_codec_get_id(rsd_tags, codec->codec_tag);
if (!codec->codec_id) {
char tag_buf[5];
av_get_codec_tag_string(tag_buf, sizeof(tag_buf), codec->codec_tag);
for (i=0; i < FF_ARRAY_ELEMS(rsd_unsupported_tags); i++) {
if (codec->codec_tag == rsd_unsupported_tags[i]) {
avpriv_request_sample(s, "Codec tag: %s", tag_buf);
return AVERROR_PATCHWELCOME;
}
}
av_log(s, AV_LOG_ERROR, "Unknown codec tag: %s\n", tag_buf);
return AVERROR_INVALIDDATA;
}
codec->channels = avio_rl32(pb);
if (!codec->channels)
return AVERROR_INVALIDDATA;
avio_skip(pb, 4); // Bit depth
codec->sample_rate = avio_rl32(pb);
if (!codec->sample_rate)
return AVERROR_INVALIDDATA;
avio_skip(pb, 4); // Unknown
switch (codec->codec_id) {
case AV_CODEC_ID_ADPCM_IMA_RAD:
codec->block_align = 20 * codec->channels;
if (pb->seekable)
st->duration = av_get_audio_frame_duration(codec, avio_size(pb) - start);
break;
case AV_CODEC_ID_ADPCM_THP:
/* RSD3GADP is mono, so only alloc enough memory
to store the coeff table for a single channel. */
start = avio_rl32(pb);
if (ff_get_extradata(codec, s->pb, 32) < 0)
return AVERROR(ENOMEM);
for (i = 0; i < 16; i++)
AV_WB16(codec->extradata + i * 2, AV_RL16(codec->extradata + i * 2));
if (pb->seekable)
st->duration = (avio_size(pb) - start) / 8 * 14;
break;
case AV_CODEC_ID_PCM_S16LE:
case AV_CODEC_ID_PCM_S16BE:
if (version != 4)
start = avio_rl32(pb);
if (pb->seekable)
st->duration = (avio_size(pb) - start) / 2 / codec->channels;
break;
}
avio_skip(pb, start - avio_tell(pb));
avpriv_set_pts_info(st, 64, 1, codec->sample_rate);
return 0;
}
static int rsd_read_header(AVFormatContext *s)
{
AVIOContext *pb = s->pb;
int i, ret, version, start = 0x800;
AVCodecContext *codec;
AVStream *st = avformat_new_stream(s, NULL);
if (!st)
return AVERROR(ENOMEM);
avio_skip(pb, 3); // "RSD"
version = avio_r8(pb) - '0';
codec = st->codec;
codec->codec_type = AVMEDIA_TYPE_AUDIO;
codec->codec_tag = avio_rl32(pb);
codec->codec_id = ff_codec_get_id(rsd_tags, codec->codec_tag);
if (!codec->codec_id) {
char tag_buf[32];
av_get_codec_tag_string(tag_buf, sizeof(tag_buf), codec->codec_tag);
for (i=0; i < FF_ARRAY_ELEMS(rsd_unsupported_tags); i++) {
if (codec->codec_tag == rsd_unsupported_tags[i]) {
avpriv_request_sample(s, "Codec tag: %s", tag_buf);
return AVERROR_PATCHWELCOME;
}
}
av_log(s, AV_LOG_ERROR, "Unknown codec tag: %s\n", tag_buf);
return AVERROR_INVALIDDATA;
}
codec->channels = avio_rl32(pb);
if (codec->channels <= 0 || codec->channels > INT_MAX / 36) {
av_log(s, AV_LOG_ERROR, "Invalid number of channels: %d\n", codec->channels);
return AVERROR_INVALIDDATA;
}
avio_skip(pb, 4); // Bit depth
codec->sample_rate = avio_rl32(pb);
if (!codec->sample_rate)
return AVERROR_INVALIDDATA;
avio_skip(pb, 4); // Unknown
switch (codec->codec_id) {
case AV_CODEC_ID_XMA2:
codec->block_align = 2048;
ff_alloc_extradata(codec, 34);
if (!codec->extradata)
return AVERROR(ENOMEM);
memset(codec->extradata, 0, 34);
break;
case AV_CODEC_ID_ADPCM_PSX:
codec->block_align = 16 * codec->channels;
if (pb->seekable)
st->duration = av_get_audio_frame_duration(codec, avio_size(pb) - start);
break;
case AV_CODEC_ID_ADPCM_IMA_RAD:
codec->block_align = 20 * codec->channels;
if (pb->seekable)
st->duration = av_get_audio_frame_duration(codec, avio_size(pb) - start);
break;
case AV_CODEC_ID_ADPCM_IMA_WAV:
if (version == 2)
start = avio_rl32(pb);
codec->bits_per_coded_sample = 4;
codec->block_align = 36 * codec->channels;
if (pb->seekable)
st->duration = av_get_audio_frame_duration(codec, avio_size(pb) - start);
break;
case AV_CODEC_ID_ADPCM_THP_LE:
/* RSD3GADP is mono, so only alloc enough memory
to store the coeff table for a single channel. */
start = avio_rl32(pb);
if ((ret = ff_get_extradata(codec, s->pb, 32)) < 0)
return ret;
if (pb->seekable)
st->duration = av_get_audio_frame_duration(codec, avio_size(pb) - start);
break;
case AV_CODEC_ID_ADPCM_THP:
codec->block_align = 8 * codec->channels;
avio_skip(s->pb, 0x1A4 - avio_tell(s->pb));
if ((ret = ff_alloc_extradata(st->codec, 32 * st->codec->channels)) < 0)
return ret;
for (i = 0; i < st->codec->channels; i++) {
avio_read(s->pb, st->codec->extradata + 32 * i, 32);
avio_skip(s->pb, 8);
}
if (pb->seekable)
st->duration = (avio_size(pb) - start) / (8 * st->codec->channels) * 14;
break;
case AV_CODEC_ID_PCM_S16LE:
case AV_CODEC_ID_PCM_S16BE:
if (version != 4)
start = avio_rl32(pb);
if (pb->seekable)
st->duration = (avio_size(pb) - start) / 2 / codec->channels;
break;
}
avio_skip(pb, start - avio_tell(pb));
if (codec->codec_id == AV_CODEC_ID_XMA2) {
avio_skip(pb, avio_rb32(pb) + avio_rb32(pb));
st->duration = avio_rb32(pb);
}
avpriv_set_pts_info(st, 64, 1, codec->sample_rate);
return 0;
}
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
int n;
AVStream *st;
if (s1->nb_streams != 1) {
av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
return AVERROR(EINVAL);
}
st = s1->streams[0];
if (!is_supported(st->codec->codec_id)) {
av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
return -1;
}
if (s->payload_type < 0)
s->payload_type = ff_rtp_get_payload_type(s1, st->codec);
s->base_timestamp = av_get_random_seed();
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
if (!s->ssrc)
s->ssrc = av_get_random_seed();
s->first_packet = 1;
s->first_rtcp_ntp_time = ff_ntp_time();
if (s1->start_time_realtime)
/* Round the NTP time to whole milliseconds. */
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
NTP_OFFSET_US;
if (s1->packet_size) {
if (s1->pb->max_packet_size)
s1->packet_size = FFMIN(s1->packet_size,
s1->pb->max_packet_size);
} else
s1->packet_size = s1->pb->max_packet_size;
if (s1->packet_size <= 12) {
av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
return AVERROR(EIO);
}
s->buf = av_malloc(s1->packet_size);
if (s->buf == NULL) {
return AVERROR(ENOMEM);
}
s->max_payload_size = s1->packet_size - 12;
s->max_frames_per_packet = 0;
if (s1->max_delay > 0) {
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
int frame_size = av_get_audio_frame_duration(st->codec, 0);
if (!frame_size)
frame_size = st->codec->frame_size;
if (frame_size == 0) {
av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
} else {
s->max_frames_per_packet =
av_rescale_q_rnd(s1->max_delay,
AV_TIME_BASE_Q,
(AVRational){ frame_size, st->codec->sample_rate },
AV_ROUND_DOWN);
}
}
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
/* FIXME: We should round down here... */
s->max_frames_per_packet = av_rescale_q(s1->max_delay, (AVRational){1, 1000000}, st->codec->time_base);
}
}
avpriv_set_pts_info(st, 32, 1, 90000);
switch(st->codec->codec_id) {
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
break;
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
break;
case AV_CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
case AV_CODEC_ID_H264:
/* check for H.264 MP4 syntax */
if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
}
break;
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
s->num_frames = 0;
goto defaultcase;
case AV_CODEC_ID_VP8:
av_log(s1, AV_LOG_ERROR, "RTP VP8 payload implementation is "
"incompatible with the latest spec drafts.\n");
break;
case AV_CODEC_ID_ADPCM_G722:
/* Due to a historical error, the clock rate for G722 in RTP is
* 8000, even if the sample rate is 16000. See RFC 3551. */
avpriv_set_pts_info(st, 32, 1, 8000);
break;
case AV_CODEC_ID_ILBC:
if (st->codec->block_align != 38 && st->codec->block_align != 50) {
av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
goto fail;
}
if (!s->max_frames_per_packet)
s->max_frames_per_packet = 1;
s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
s->max_payload_size / st->codec->block_align);
goto defaultcase;
case AV_CODEC_ID_AMR_NB:
case AV_CODEC_ID_AMR_WB:
if (!s->max_frames_per_packet)
s->max_frames_per_packet = 12;
if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
n = 31;
else
n = 61;
/* max_header_toc_size + the largest AMR payload must fit */
if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
goto fail;
}
if (st->codec->channels != 1) {
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
goto fail;
}
case AV_CODEC_ID_AAC:
s->num_frames = 0;
default:
defaultcase:
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
s->buf_ptr = s->buf;
break;
}
return 0;
fail:
av_freep(&s->buf);
return AVERROR(EINVAL);
}
int VideoStore::writeAudioFramePacket(AVPacket *ipkt, AVStream *input_st){
/*
See 01349 of http://www.ffmpeg.org/doxygen/trunk/ffmpeg_8c-source.html
do_streamcopy
*/
if(!audio_st)
return -1;//FIXME -ve return codes do not free packet in ffmpeg_camera at the moment
/*if(!keyframeMessage)
return -1;*/
char szErr[1024];
int64_t ost_tb_start_time = av_rescale_q(startTime, AV_TIME_BASE_Q, video_st->time_base);
AVPacket opkt;
AVPicture pict;//Not sure how much we need this
av_init_packet(&opkt);
//opkt.stream_index = audio_st->index;
//FIXME does the PTS/DTS code below still apply to our audio the same way?
//Scale the PTS of the outgoing packet to be the correct time base
if (ipkt->pts != AV_NOPTS_VALUE)
opkt.pts = av_rescale_q(ipkt->pts-startPts, input_st->time_base, audio_st->time_base) - ost_tb_start_time;
else
opkt.pts = AV_NOPTS_VALUE;
//Scale the DTS of the outgoing packet to be the correct time base
if(ipkt->dts == AV_NOPTS_VALUE)
opkt.dts = av_rescale_q(input_st->cur_dts-startDts, AV_TIME_BASE_Q, audio_st->time_base);
else
opkt.dts = av_rescale_q(ipkt->dts-startDts, input_st->time_base, audio_st->time_base);
opkt.dts -= ost_tb_start_time;
if (audio_st->codec->codec_type == AVMEDIA_TYPE_AUDIO && ipkt->dts != AV_NOPTS_VALUE) {
int duration = av_get_audio_frame_duration(input_st->codec, ipkt->size);
if(!duration)
duration = input_st->codec->frame_size;
//FIXME where to get filter_in_rescale_delta_last
opkt.dts = opkt.pts = av_rescale_delta(input_st->time_base, ipkt->dts,
(AVRational){1, input_st->codec->sample_rate}, duration, &filter_in_rescale_delta_last,
audio_st->time_base) - ost_tb_start_time;
}
opkt.duration = av_rescale_q(ipkt->duration, input_st->time_base, audio_st->time_base);
opkt.pos=-1;
opkt.flags = ipkt->flags;
//TODO: Should be checking if not H264, mpeg1, etc
//Maybe the check isn't needed if we're only going to do this for H264 video incoming
opkt.data = ipkt->data;
opkt.size = ipkt->size;
opkt.stream_index = ipkt->stream_index;
/*opkt.flags |= AV_PKT_FLAG_KEY;*/
int ret;
ret = av_interleaved_write_frame(oc, &opkt);
if(ret<0){
Fatal("Error encoding audio frame packet: %s\n", av_make_error_string(szErr,1024,ret));
}
av_free_packet(&opkt);
return 0;
}
static int rv10_write_header(AVFormatContext *ctx,
int data_size, int index_pos)
{
RMMuxContext *rm = ctx->priv_data;
AVIOContext *s = ctx->pb;
StreamInfo *stream;
unsigned char *data_offset_ptr, *start_ptr;
const char *desc, *mimetype;
int nb_packets, packet_total_size, packet_max_size, size, packet_avg_size, i;
int bit_rate, v, duration, flags, data_pos;
AVDictionaryEntry *tag;
start_ptr = s->buf_ptr;
ffio_wfourcc(s, ".RMF");
avio_wb32(s,18); /* header size */
avio_wb16(s,0);
avio_wb32(s,0);
avio_wb32(s,4 + ctx->nb_streams); /* num headers */
ffio_wfourcc(s,"PROP");
avio_wb32(s, 50);
avio_wb16(s, 0);
packet_max_size = 0;
packet_total_size = 0;
nb_packets = 0;
bit_rate = 0;
duration = 0;
for(i=0;i<ctx->nb_streams;i++) {
StreamInfo *stream = &rm->streams[i];
bit_rate += stream->bit_rate;
if (stream->packet_max_size > packet_max_size)
packet_max_size = stream->packet_max_size;
nb_packets += stream->nb_packets;
packet_total_size += stream->packet_total_size;
/* select maximum duration */
v = (int) (1000.0 * (float)stream->total_frames / stream->frame_rate);
if (v > duration)
duration = v;
}
avio_wb32(s, bit_rate); /* max bit rate */
avio_wb32(s, bit_rate); /* avg bit rate */
avio_wb32(s, packet_max_size); /* max packet size */
if (nb_packets > 0)
packet_avg_size = packet_total_size / nb_packets;
else
packet_avg_size = 0;
avio_wb32(s, packet_avg_size); /* avg packet size */
avio_wb32(s, nb_packets); /* num packets */
avio_wb32(s, duration); /* duration */
avio_wb32(s, BUFFER_DURATION); /* preroll */
avio_wb32(s, index_pos); /* index offset */
/* computation of data the data offset */
data_offset_ptr = s->buf_ptr;
avio_wb32(s, 0); /* data offset : will be patched after */
avio_wb16(s, ctx->nb_streams); /* num streams */
flags = 1 | 2; /* save allowed & perfect play */
if (!s->seekable)
flags |= 4; /* live broadcast */
avio_wb16(s, flags);
/* comments */
ffio_wfourcc(s,"CONT");
size = 4 * 2 + 10;
for(i=0; i<FF_ARRAY_ELEMS(ff_rm_metadata); i++) {
tag = av_dict_get(ctx->metadata, ff_rm_metadata[i], NULL, 0);
if(tag) size += strlen(tag->value);
}
avio_wb32(s,size);
avio_wb16(s,0);
for(i=0; i<FF_ARRAY_ELEMS(ff_rm_metadata); i++) {
tag = av_dict_get(ctx->metadata, ff_rm_metadata[i], NULL, 0);
put_str(s, tag ? tag->value : "");
}
for(i=0;i<ctx->nb_streams;i++) {
int codec_data_size;
stream = &rm->streams[i];
if (stream->enc->codec_type == AVMEDIA_TYPE_AUDIO) {
desc = "The Audio Stream";
mimetype = "audio/x-pn-realaudio";
codec_data_size = 73;
} else {
desc = "The Video Stream";
mimetype = "video/x-pn-realvideo";
codec_data_size = 34;
}
ffio_wfourcc(s,"MDPR");
size = 10 + 9 * 4 + strlen(desc) + strlen(mimetype) + codec_data_size;
avio_wb32(s, size);
avio_wb16(s, 0);
avio_wb16(s, i); /* stream number */
avio_wb32(s, stream->bit_rate); /* max bit rate */
avio_wb32(s, stream->bit_rate); /* avg bit rate */
avio_wb32(s, stream->packet_max_size); /* max packet size */
if (stream->nb_packets > 0)
packet_avg_size = stream->packet_total_size /
stream->nb_packets;
else
packet_avg_size = 0;
avio_wb32(s, packet_avg_size); /* avg packet size */
avio_wb32(s, 0); /* start time */
avio_wb32(s, BUFFER_DURATION); /* preroll */
/* duration */
if (!s->seekable || !stream->total_frames)
avio_wb32(s, (int)(3600 * 1000));
else
avio_wb32(s, (int)(stream->total_frames * 1000 / stream->frame_rate));
put_str8(s, desc);
put_str8(s, mimetype);
avio_wb32(s, codec_data_size);
if (stream->enc->codec_type == AVMEDIA_TYPE_AUDIO) {
int coded_frame_size, fscode, sample_rate;
int frame_size = av_get_audio_frame_duration(stream->enc, 0);
sample_rate = stream->enc->sample_rate;
coded_frame_size = (stream->enc->bit_rate *
frame_size) / (8 * sample_rate);
/* audio codec info */
avio_write(s, ".ra", 3);
avio_w8(s, 0xfd);
avio_wb32(s, 0x00040000); /* version */
ffio_wfourcc(s, ".ra4");
avio_wb32(s, 0x01b53530); /* stream length */
avio_wb16(s, 4); /* unknown */
avio_wb32(s, 0x39); /* header size */
switch(sample_rate) {
case 48000:
case 24000:
case 12000:
fscode = 1;
break;
default:
case 44100:
case 22050:
case 11025:
fscode = 2;
break;
case 32000:
case 16000:
case 8000:
fscode = 3;
}
avio_wb16(s, fscode); /* codec additional info, for AC-3, seems
to be a frequency code */
/* special hack to compensate rounding errors... */
if (coded_frame_size == 557)
coded_frame_size--;
avio_wb32(s, coded_frame_size); /* frame length */
avio_wb32(s, 0x51540); /* unknown */
avio_wb32(s, 0x249f0); /* unknown */
avio_wb32(s, 0x249f0); /* unknown */
avio_wb16(s, 0x01);
/* frame length : seems to be very important */
avio_wb16(s, coded_frame_size);
avio_wb32(s, 0); /* unknown */
avio_wb16(s, stream->enc->sample_rate); /* sample rate */
avio_wb32(s, 0x10); /* unknown */
avio_wb16(s, stream->enc->channels);
put_str8(s, "Int0"); /* codec name */
if (stream->enc->codec_tag) {
avio_w8(s, 4); /* tag length */
avio_wl32(s, stream->enc->codec_tag);
} else {
av_log(ctx, AV_LOG_ERROR, "Invalid codec tag\n");
return -1;
}
avio_wb16(s, 0); /* title length */
avio_wb16(s, 0); /* author length */
avio_wb16(s, 0); /* copyright length */
avio_w8(s, 0); /* end of header */
} else {
/* video codec info */
avio_wb32(s,34); /* size */
ffio_wfourcc(s, "VIDO");
if(stream->enc->codec_id == AV_CODEC_ID_RV10)
ffio_wfourcc(s,"RV10");
else
ffio_wfourcc(s,"RV20");
avio_wb16(s, stream->enc->width);
avio_wb16(s, stream->enc->height);
avio_wb16(s, (int) stream->frame_rate); /* frames per seconds ? */
avio_wb32(s,0); /* unknown meaning */
avio_wb16(s, (int) stream->frame_rate); /* unknown meaning */
avio_wb32(s,0); /* unknown meaning */
avio_wb16(s, 8); /* unknown meaning */
/* Seems to be the codec version: only use basic H263. The next
versions seems to add a diffential DC coding as in
MPEG... nothing new under the sun */
if(stream->enc->codec_id == AV_CODEC_ID_RV10)
avio_wb32(s,0x10000000);
else
avio_wb32(s,0x20103001);
//avio_wb32(s,0x10003000);
}
}
/* patch data offset field */
data_pos = s->buf_ptr - start_ptr;
rm->data_pos = data_pos;
data_offset_ptr[0] = data_pos >> 24;
data_offset_ptr[1] = data_pos >> 16;
data_offset_ptr[2] = data_pos >> 8;
data_offset_ptr[3] = data_pos;
/* data stream */
ffio_wfourcc(s, "DATA");
avio_wb32(s,data_size + 10 + 8);
avio_wb16(s,0);
avio_wb32(s, nb_packets); /* number of packets */
avio_wb32(s,0); /* next data header */
return 0;
}
int
ff_voc_get_packet(AVFormatContext *s, AVPacket *pkt, AVStream *st, int max_size)
{
VocDecContext *voc = s->priv_data;
AVCodecContext *dec = st->codec;
AVIOContext *pb = s->pb;
VocType type;
int size, tmp_codec=-1;
int sample_rate = 0;
int channels = 1;
int64_t duration;
int ret;
av_add_index_entry(st,
avio_tell(pb),
voc->pts,
voc->remaining_size,
0,
AVINDEX_KEYFRAME);
while (!voc->remaining_size) {
type = avio_r8(pb);
if (type == VOC_TYPE_EOF)
return AVERROR_EOF;
voc->remaining_size = avio_rl24(pb);
if (!voc->remaining_size) {
if (!s->pb->seekable)
return AVERROR(EIO);
voc->remaining_size = avio_size(pb) - avio_tell(pb);
}
max_size -= 4;
switch (type) {
case VOC_TYPE_VOICE_DATA:
if (!dec->sample_rate) {
dec->sample_rate = 1000000 / (256 - avio_r8(pb));
if (sample_rate)
dec->sample_rate = sample_rate;
avpriv_set_pts_info(st, 64, 1, dec->sample_rate);
dec->channels = channels;
dec->bits_per_coded_sample = av_get_bits_per_sample(dec->codec_id);
} else
avio_skip(pb, 1);
tmp_codec = avio_r8(pb);
voc->remaining_size -= 2;
max_size -= 2;
channels = 1;
break;
case VOC_TYPE_VOICE_DATA_CONT:
break;
case VOC_TYPE_EXTENDED:
sample_rate = avio_rl16(pb);
avio_r8(pb);
channels = avio_r8(pb) + 1;
sample_rate = 256000000 / (channels * (65536 - sample_rate));
voc->remaining_size = 0;
max_size -= 4;
break;
case VOC_TYPE_NEW_VOICE_DATA:
if (!dec->sample_rate) {
dec->sample_rate = avio_rl32(pb);
avpriv_set_pts_info(st, 64, 1, dec->sample_rate);
dec->bits_per_coded_sample = avio_r8(pb);
dec->channels = avio_r8(pb);
} else
avio_skip(pb, 6);
tmp_codec = avio_rl16(pb);
avio_skip(pb, 4);
voc->remaining_size -= 12;
max_size -= 12;
break;
default:
avio_skip(pb, voc->remaining_size);
max_size -= voc->remaining_size;
voc->remaining_size = 0;
break;
}
}
if (tmp_codec >= 0) {
tmp_codec = ff_codec_get_id(ff_voc_codec_tags, tmp_codec);
if (dec->codec_id == AV_CODEC_ID_NONE)
dec->codec_id = tmp_codec;
else if (dec->codec_id != tmp_codec)
av_log(s, AV_LOG_WARNING, "Ignoring mid-stream change in audio codec\n");
if (dec->codec_id == AV_CODEC_ID_NONE) {
if (s->audio_codec_id == AV_CODEC_ID_NONE) {
av_log(s, AV_LOG_ERROR, "unknown codec tag\n");
return AVERROR(EINVAL);
}
av_log(s, AV_LOG_WARNING, "unknown codec tag\n");
}
}
dec->bit_rate = dec->sample_rate * dec->channels * dec->bits_per_coded_sample;
if (max_size <= 0)
max_size = 2048;
size = FFMIN(voc->remaining_size, max_size);
voc->remaining_size -= size;
ret = av_get_packet(pb, pkt, size);
pkt->dts = pkt->pts = voc->pts;
duration = av_get_audio_frame_duration(st->codec, size);
if (duration > 0 && voc->pts != AV_NOPTS_VALUE)
voc->pts += duration;
else
voc->pts = AV_NOPTS_VALUE;
return ret;
}
static int rtp_write_header(AVFormatContext *s1)
{
RTPMuxContext *s = s1->priv_data;
int n;
AVStream *st;
if (s1->nb_streams != 1) {
av_log(s1, AV_LOG_ERROR, "Only one stream supported in the RTP muxer\n");
return AVERROR(EINVAL);
}
st = s1->streams[0];
if (!is_supported(st->codec->codec_id)) {
av_log(s1, AV_LOG_ERROR, "Unsupported codec %s\n", avcodec_get_name(st->codec->codec_id));
return -1;
}
if (s->payload_type < 0) {
/* Re-validate non-dynamic payload types */
if (st->id < RTP_PT_PRIVATE)
st->id = ff_rtp_get_payload_type(s1, st->codec, -1);
s->payload_type = st->id;
} else {
/* private option takes priority */
st->id = s->payload_type;
}
s->base_timestamp = av_get_random_seed();
s->timestamp = s->base_timestamp;
s->cur_timestamp = 0;
if (!s->ssrc)
s->ssrc = av_get_random_seed();
s->first_packet = 1;
s->first_rtcp_ntp_time = ff_ntp_time();
if (s1->start_time_realtime)
/* Round the NTP time to whole milliseconds. */
s->first_rtcp_ntp_time = (s1->start_time_realtime / 1000) * 1000 +
NTP_OFFSET_US;
// Pick a random sequence start number, but in the lower end of the
// available range, so that any wraparound doesn't happen immediately.
// (Immediate wraparound would be an issue for SRTP.)
if (s->seq < 0) {
if (st->codec->flags & CODEC_FLAG_BITEXACT) {
s->seq = 0;
} else
s->seq = av_get_random_seed() & 0x0fff;
} else
s->seq &= 0xffff; // Use the given parameter, wrapped to the right interval
if (s1->packet_size) {
if (s1->pb->max_packet_size)
s1->packet_size = FFMIN(s1->packet_size,
s1->pb->max_packet_size);
} else
s1->packet_size = s1->pb->max_packet_size;
if (s1->packet_size <= 12) {
av_log(s1, AV_LOG_ERROR, "Max packet size %d too low\n", s1->packet_size);
return AVERROR(EIO);
}
s->buf = av_malloc(s1->packet_size);
if (s->buf == NULL) {
return AVERROR(ENOMEM);
}
s->max_payload_size = s1->packet_size - 12;
s->max_frames_per_packet = 0;
if (s1->max_delay > 0) {
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
int frame_size = av_get_audio_frame_duration(st->codec, 0);
if (!frame_size)
frame_size = st->codec->frame_size;
if (frame_size == 0) {
av_log(s1, AV_LOG_ERROR, "Cannot respect max delay: frame size = 0\n");
} else {
AVRational avr1 ={ frame_size, st->codec->sample_rate };
s->max_frames_per_packet = av_rescale_q_rnd(s1->max_delay,AV_TIME_BASE_Q,avr1, AV_ROUND_DOWN);
}
}
if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
/* FIXME: We should round down here... */
AVRational avr2 = {1, 1000000};
s->max_frames_per_packet = av_rescale_q(s1->max_delay, avr2, st->codec->time_base);
}
}
avpriv_set_pts_info(st, 32, 1, 90000);
switch(st->codec->codec_id) {
case AV_CODEC_ID_MP2:
case AV_CODEC_ID_MP3:
s->buf_ptr = s->buf + 4;
break;
case AV_CODEC_ID_MPEG1VIDEO:
case AV_CODEC_ID_MPEG2VIDEO:
break;
case AV_CODEC_ID_MPEG2TS:
n = s->max_payload_size / TS_PACKET_SIZE;
if (n < 1)
n = 1;
s->max_payload_size = n * TS_PACKET_SIZE;
s->buf_ptr = s->buf;
break;
case AV_CODEC_ID_H264:
/* check for H.264 MP4 syntax */
if (st->codec->extradata_size > 4 && st->codec->extradata[0] == 1) {
s->nal_length_size = (st->codec->extradata[4] & 0x03) + 1;
}
break;
case AV_CODEC_ID_VORBIS:
case AV_CODEC_ID_THEORA:
if (!s->max_frames_per_packet) s->max_frames_per_packet = 15;
s->max_frames_per_packet = av_clip(s->max_frames_per_packet, 1, 15);
s->max_payload_size -= 6; // ident+frag+tdt/vdt+pkt_num+pkt_length
s->num_frames = 0;
goto defaultcase;
case AV_CODEC_ID_ADPCM_G722:
/* Due to a historical error, the clock rate for G722 in RTP is
* 8000, even if the sample rate is 16000. See RFC 3551. */
avpriv_set_pts_info(st, 32, 1, 8000);
break;
case AV_CODEC_ID_OPUS:
if (st->codec->channels > 2) {
av_log(s1, AV_LOG_ERROR, "Multistream opus not supported in RTP\n");
goto fail;
}
/* The opus RTP RFC says that all opus streams should use 48000 Hz
* as clock rate, since all opus sample rates can be expressed in
* this clock rate, and sample rate changes on the fly are supported. */
avpriv_set_pts_info(st, 32, 1, 48000);
break;
case AV_CODEC_ID_ILBC:
if (st->codec->block_align != 38 && st->codec->block_align != 50) {
av_log(s1, AV_LOG_ERROR, "Incorrect iLBC block size specified\n");
goto fail;
}
if (!s->max_frames_per_packet)
s->max_frames_per_packet = 1;
s->max_frames_per_packet = FFMIN(s->max_frames_per_packet,
s->max_payload_size / st->codec->block_align);
goto defaultcase;
case AV_CODEC_ID_AMR_NB:
case AV_CODEC_ID_AMR_WB:
if (!s->max_frames_per_packet)
s->max_frames_per_packet = 12;
if (st->codec->codec_id == AV_CODEC_ID_AMR_NB)
n = 31;
else
n = 61;
/* max_header_toc_size + the largest AMR payload must fit */
if (1 + s->max_frames_per_packet + n > s->max_payload_size) {
av_log(s1, AV_LOG_ERROR, "RTP max payload size too small for AMR\n");
goto fail;
}
if (st->codec->channels != 1) {
av_log(s1, AV_LOG_ERROR, "Only mono is supported\n");
goto fail;
}
case AV_CODEC_ID_AAC:
s->num_frames = 0;
default:
defaultcase:
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
avpriv_set_pts_info(st, 32, 1, st->codec->sample_rate);
}
s->buf_ptr = s->buf;
break;
}
return 0;
fail:
av_freep(&s->buf);
return AVERROR(EINVAL);
}
/* returns the size or -1 on error */
int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc, int flags)
{
int bps, blkalign, bytespersec, frame_size;
int hdrsize;
int64_t hdrstart = avio_tell(pb);
int waveformatextensible;
uint8_t temp[256];
uint8_t *riff_extradata = temp;
uint8_t *riff_extradata_start = temp;
if (!enc->codec_tag || enc->codec_tag > 0xffff)
return -1;
/* We use the known constant frame size for the codec if known, otherwise
* fall back on using AVCodecContext.frame_size, which is not as reliable
* for indicating packet duration. */
frame_size = av_get_audio_frame_duration(enc, enc->block_align);
waveformatextensible = (enc->channels > 2 && enc->channel_layout) ||
enc->sample_rate > 48000 ||
enc->codec_id == AV_CODEC_ID_EAC3 ||
av_get_bits_per_sample(enc->codec_id) > 16;
if (waveformatextensible)
avio_wl16(pb, 0xfffe);
else
avio_wl16(pb, enc->codec_tag);
avio_wl16(pb, enc->channels);
avio_wl32(pb, enc->sample_rate);
if (enc->codec_id == AV_CODEC_ID_ATRAC3 ||
enc->codec_id == AV_CODEC_ID_G723_1 ||
enc->codec_id == AV_CODEC_ID_MP2 ||
enc->codec_id == AV_CODEC_ID_MP3 ||
enc->codec_id == AV_CODEC_ID_GSM_MS) {
bps = 0;
} else {
if (!(bps = av_get_bits_per_sample(enc->codec_id))) {
if (enc->bits_per_coded_sample)
bps = enc->bits_per_coded_sample;
else
bps = 16; // default to 16
}
}
if (bps != enc->bits_per_coded_sample && enc->bits_per_coded_sample) {
av_log(enc, AV_LOG_WARNING,
"requested bits_per_coded_sample (%d) "
"and actually stored (%d) differ\n",
enc->bits_per_coded_sample, bps);
}
if (enc->codec_id == AV_CODEC_ID_MP2) {
blkalign = (144 * enc->bit_rate - 1)/enc->sample_rate + 1;
} else if (enc->codec_id == AV_CODEC_ID_MP3) {
blkalign = 576 * (enc->sample_rate <= (24000 + 32000)/2 ? 1 : 2);
} else if (enc->codec_id == AV_CODEC_ID_AC3) {
blkalign = 3840; /* maximum bytes per frame */
} else if (enc->codec_id == AV_CODEC_ID_AAC) {
blkalign = 768 * enc->channels; /* maximum bytes per frame */
} else if (enc->codec_id == AV_CODEC_ID_G723_1) {
blkalign = 24;
} else if (enc->block_align != 0) { /* specified by the codec */
blkalign = enc->block_align;
} else
blkalign = bps * enc->channels / av_gcd(8, bps);
if (enc->codec_id == AV_CODEC_ID_PCM_U8 ||
enc->codec_id == AV_CODEC_ID_PCM_S24LE ||
enc->codec_id == AV_CODEC_ID_PCM_S32LE ||
enc->codec_id == AV_CODEC_ID_PCM_F32LE ||
enc->codec_id == AV_CODEC_ID_PCM_F64LE ||
enc->codec_id == AV_CODEC_ID_PCM_S16LE) {
bytespersec = enc->sample_rate * blkalign;
} else if (enc->codec_id == AV_CODEC_ID_G723_1) {
bytespersec = 800;
} else {
bytespersec = enc->bit_rate / 8;
}
avio_wl32(pb, bytespersec); /* bytes per second */
avio_wl16(pb, blkalign); /* block align */
avio_wl16(pb, bps); /* bits per sample */
if (enc->codec_id == AV_CODEC_ID_MP3) {
bytestream_put_le16(&riff_extradata, 1); /* wID */
bytestream_put_le32(&riff_extradata, 2); /* fdwFlags */
bytestream_put_le16(&riff_extradata, 1152); /* nBlockSize */
bytestream_put_le16(&riff_extradata, 1); /* nFramesPerBlock */
bytestream_put_le16(&riff_extradata, 1393); /* nCodecDelay */
} else if (enc->codec_id == AV_CODEC_ID_MP2) {
/* fwHeadLayer */
bytestream_put_le16(&riff_extradata, 2);
/* dwHeadBitrate */
bytestream_put_le32(&riff_extradata, enc->bit_rate);
/* fwHeadMode */
bytestream_put_le16(&riff_extradata, enc->channels == 2 ? 1 : 8);
/* fwHeadModeExt */
bytestream_put_le16(&riff_extradata, 0);
/* wHeadEmphasis */
bytestream_put_le16(&riff_extradata, 1);
/* fwHeadFlags */
bytestream_put_le16(&riff_extradata, 16);
/* dwPTSLow */
bytestream_put_le32(&riff_extradata, 0);
/* dwPTSHigh */
bytestream_put_le32(&riff_extradata, 0);
} else if (enc->codec_id == AV_CODEC_ID_G723_1) {
bytestream_put_le32(&riff_extradata, 0x9ace0002); /* extradata needed for msacm g723.1 codec */
bytestream_put_le32(&riff_extradata, 0xaea2f732);
bytestream_put_le16(&riff_extradata, 0xacde);
} else if (enc->codec_id == AV_CODEC_ID_GSM_MS ||
enc->codec_id == AV_CODEC_ID_ADPCM_IMA_WAV) {
/* wSamplesPerBlock */
bytestream_put_le16(&riff_extradata, frame_size);
} else if (enc->extradata_size) {
riff_extradata_start = enc->extradata;
riff_extradata = enc->extradata + enc->extradata_size;
}
/* write WAVEFORMATEXTENSIBLE extensions */
if (waveformatextensible) {
int write_channel_mask = enc->strict_std_compliance < FF_COMPLIANCE_NORMAL ||
enc->channel_layout < 0x40000;
/* 22 is WAVEFORMATEXTENSIBLE size */
avio_wl16(pb, riff_extradata - riff_extradata_start + 22);
/* ValidBitsPerSample || SamplesPerBlock || Reserved */
avio_wl16(pb, bps);
/* dwChannelMask */
avio_wl32(pb, write_channel_mask ? enc->channel_layout : 0);
/* GUID + next 3 */
if (enc->codec_id == AV_CODEC_ID_EAC3) {
ff_put_guid(pb, ff_get_codec_guid(enc->codec_id, ff_codec_wav_guids));
} else {
avio_wl32(pb, enc->codec_tag);
avio_wl32(pb, 0x00100000);
avio_wl32(pb, 0xAA000080);
avio_wl32(pb, 0x719B3800);
}
} else if ((flags & FF_PUT_WAV_HEADER_FORCE_WAVEFORMATEX) ||
enc->codec_tag != 0x0001 /* PCM */ ||
riff_extradata - riff_extradata_start) {
/* WAVEFORMATEX */
avio_wl16(pb, riff_extradata - riff_extradata_start); /* cbSize */
} /* else PCMWAVEFORMAT */
avio_write(pb, riff_extradata_start, riff_extradata - riff_extradata_start);
hdrsize = avio_tell(pb) - hdrstart;
if (hdrsize & 1) {
hdrsize++;
avio_w8(pb, 0);
}
return hdrsize;
}
/* returns the size or -1 on error */
int ff_put_wav_header(AVIOContext *pb, AVCodecContext *enc)
{
int bps, blkalign, bytespersec, frame_size;
int hdrsize = 18;
int waveformatextensible;
uint8_t temp[256];
uint8_t *riff_extradata = temp;
uint8_t *riff_extradata_start = temp;
if (!enc->codec_tag || enc->codec_tag > 0xffff)
return -1;
/* We use the known constant frame size for the codec if known, otherwise
* fall back on using AVCodecContext.frame_size, which is not as reliable
* for indicating packet duration. */
frame_size = av_get_audio_frame_duration(enc, enc->block_align);
waveformatextensible = (enc->channels > 2 && enc->channel_layout) ||
enc->sample_rate > 48000 ||
av_get_bits_per_sample(enc->codec_id) > 16;
if (waveformatextensible)
avio_wl16(pb, 0xfffe);
else
avio_wl16(pb, enc->codec_tag);
avio_wl16(pb, enc->channels);
avio_wl32(pb, enc->sample_rate);
if (enc->codec_id == AV_CODEC_ID_MP2 ||
enc->codec_id == AV_CODEC_ID_MP3 ||
enc->codec_id == AV_CODEC_ID_GSM_MS) {
bps = 0;
} else {
if (!(bps = av_get_bits_per_sample(enc->codec_id))) {
if (enc->bits_per_coded_sample)
bps = enc->bits_per_coded_sample;
else
bps = 16; // default to 16
}
}
if (bps != enc->bits_per_coded_sample && enc->bits_per_coded_sample) {
av_log(enc, AV_LOG_WARNING,
"requested bits_per_coded_sample (%d) "
"and actually stored (%d) differ\n",
enc->bits_per_coded_sample, bps);
}
if (enc->codec_id == AV_CODEC_ID_MP2) {
blkalign = frame_size;
} else if (enc->codec_id == AV_CODEC_ID_MP3) {
blkalign = 576 * (enc->sample_rate <= 24000 ? 1 : 2);
} else if (enc->codec_id == AV_CODEC_ID_AC3) {
blkalign = 3840; /* maximum bytes per frame */
} else if (enc->block_align != 0) { /* specified by the codec */
blkalign = enc->block_align;
} else
blkalign = bps * enc->channels / av_gcd(8, bps);
if (enc->codec_id == AV_CODEC_ID_PCM_U8 ||
enc->codec_id == AV_CODEC_ID_PCM_S24LE ||
enc->codec_id == AV_CODEC_ID_PCM_S32LE ||
enc->codec_id == AV_CODEC_ID_PCM_F32LE ||
enc->codec_id == AV_CODEC_ID_PCM_F64LE ||
enc->codec_id == AV_CODEC_ID_PCM_S16LE) {
bytespersec = enc->sample_rate * blkalign;
} else {
bytespersec = enc->bit_rate / 8;
}
avio_wl32(pb, bytespersec); /* bytes per second */
avio_wl16(pb, blkalign); /* block align */
avio_wl16(pb, bps); /* bits per sample */
if (enc->codec_id == AV_CODEC_ID_MP3) {
hdrsize += 12;
bytestream_put_le16(&riff_extradata, 1); /* wID */
bytestream_put_le32(&riff_extradata, 2); /* fdwFlags */
bytestream_put_le16(&riff_extradata, 1152); /* nBlockSize */
bytestream_put_le16(&riff_extradata, 1); /* nFramesPerBlock */
bytestream_put_le16(&riff_extradata, 1393); /* nCodecDelay */
} else if (enc->codec_id == AV_CODEC_ID_MP2) {
hdrsize += 22;
/* fwHeadLayer */
bytestream_put_le16(&riff_extradata, 2);
/* dwHeadBitrate */
bytestream_put_le32(&riff_extradata, enc->bit_rate);
/* fwHeadMode */
bytestream_put_le16(&riff_extradata, enc->channels == 2 ? 1 : 8);
/* fwHeadModeExt */
bytestream_put_le16(&riff_extradata, 0);
/* wHeadEmphasis */
bytestream_put_le16(&riff_extradata, 1);
/* fwHeadFlags */
bytestream_put_le16(&riff_extradata, 16);
/* dwPTSLow */
bytestream_put_le32(&riff_extradata, 0);
/* dwPTSHigh */
bytestream_put_le32(&riff_extradata, 0);
} else if (enc->codec_id == AV_CODEC_ID_GSM_MS ||
enc->codec_id == AV_CODEC_ID_ADPCM_IMA_WAV) {
hdrsize += 2;
/* wSamplesPerBlock */
bytestream_put_le16(&riff_extradata, frame_size);
} else if (enc->extradata_size) {
riff_extradata_start = enc->extradata;
riff_extradata = enc->extradata + enc->extradata_size;
hdrsize += enc->extradata_size;
}
/* write WAVEFORMATEXTENSIBLE extensions */
if (waveformatextensible) {
hdrsize += 22;
/* 22 is WAVEFORMATEXTENSIBLE size */
avio_wl16(pb, riff_extradata - riff_extradata_start + 22);
/* ValidBitsPerSample || SamplesPerBlock || Reserved */
avio_wl16(pb, bps);
/* dwChannelMask */
avio_wl32(pb, enc->channel_layout);
/* GUID + next 3 */
avio_wl32(pb, enc->codec_tag);
avio_wl32(pb, 0x00100000);
avio_wl32(pb, 0xAA000080);
avio_wl32(pb, 0x719B3800);
} else {
avio_wl16(pb, riff_extradata - riff_extradata_start); /* cbSize */
}
avio_write(pb, riff_extradata_start, riff_extradata - riff_extradata_start);
if (hdrsize & 1) {
hdrsize++;
avio_w8(pb, 0);
}
return hdrsize;
}
/* Returns the number of sound data frames or negative on error */
static int get_aiff_header(AVFormatContext *s, int size,
unsigned version)
{
AVIOContext *pb = s->pb;
AVCodecContext *codec = s->streams[0]->codec;
AIFFInputContext *aiff = s->priv_data;
int exp;
uint64_t val;
int sample_rate;
unsigned int num_frames;
if (size & 1)
size++;
codec->codec_type = AVMEDIA_TYPE_AUDIO;
codec->channels = avio_rb16(pb);
num_frames = avio_rb32(pb);
codec->bits_per_coded_sample = avio_rb16(pb);
exp = avio_rb16(pb) - 16383 - 63;
val = avio_rb64(pb);
if (exp <-63 || exp >63) {
av_log(s, AV_LOG_ERROR, "exp %d is out of range\n", exp);
return AVERROR_INVALIDDATA;
}
if (exp >= 0)
sample_rate = val << exp;
else
sample_rate = (val + (1ULL<<(-exp-1))) >> -exp;
codec->sample_rate = sample_rate;
size -= 18;
/* get codec id for AIFF-C */
if (size < 4) {
version = AIFF;
} else if (version == AIFF_C_VERSION1) {
codec->codec_tag = avio_rl32(pb);
codec->codec_id = ff_codec_get_id(ff_codec_aiff_tags, codec->codec_tag);
if (codec->codec_id == AV_CODEC_ID_NONE) {
char tag[32];
av_get_codec_tag_string(tag, sizeof(tag), codec->codec_tag);
avpriv_request_sample(s, "unknown or unsupported codec tag: %s", tag);
}
size -= 4;
}
if (version != AIFF_C_VERSION1 || codec->codec_id == AV_CODEC_ID_PCM_S16BE) {
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);
aiff->block_duration = 1;
} else {
switch (codec->codec_id) {
case AV_CODEC_ID_PCM_F32BE:
case AV_CODEC_ID_PCM_F64BE:
case AV_CODEC_ID_PCM_S16LE:
case AV_CODEC_ID_PCM_ALAW:
case AV_CODEC_ID_PCM_MULAW:
aiff->block_duration = 1;
break;
case AV_CODEC_ID_ADPCM_IMA_QT:
codec->block_align = 34*codec->channels;
break;
case AV_CODEC_ID_MACE3:
codec->block_align = 2*codec->channels;
break;
case AV_CODEC_ID_ADPCM_G726LE:
codec->bits_per_coded_sample = 5;
case AV_CODEC_ID_ADPCM_IMA_WS:
case AV_CODEC_ID_ADPCM_G722:
case AV_CODEC_ID_MACE6:
case AV_CODEC_ID_SDX2_DPCM:
codec->block_align = 1*codec->channels;
break;
case AV_CODEC_ID_GSM:
codec->block_align = 33;
break;
default:
aiff->block_duration = 1;
break;
}
if (codec->block_align > 0)
aiff->block_duration = av_get_audio_frame_duration(codec,
codec->block_align);
}
/* 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 = (av_get_bits_per_sample(codec->codec_id) * codec->channels) >> 3;
if (aiff->block_duration) {
codec->bit_rate = codec->sample_rate * (codec->block_align << 3) /
aiff->block_duration;
}
/* Chunk is over */
if (size)
avio_skip(pb, size);
return num_frames;
}
