最近遇到一个ffmpeg ts流估计时长问题 ， 此文对我帮助特别大，特别转载
本文转自: https://www.cnblogs.com/huaping-audio/p/3454781.html
ts流中的时间估计
我们知道ts流中是没有时间信息的，我门来看看ffmpeg是怎么估计其duration的
方法1.通过pts来估计
static void estimate_timings_from_pts(AVFormatContext *ic, int64_t old_offset)
{
AVPacket pkt1, *pkt = &pkt1;
AVStream *st;
int read_size, i, ret;
int64_t end_time;
int64_t filesize, offset, duration;
int retry=0;
/* flush packet queue */
flush_packet_queue(ic);
for (i=0; i<ic->nb_streams; i++) {
st = ic->streams[i];
if (st->start_time == AV_NOPTS_VALUE && st->first_dts == AV_NOPTS_VALUE)
av_log(st->codec, AV_LOG_WARNING, "start time is not set in estimate_timings_from_pts\n");
if (st->parser) {
av_parser_close(st->parser);
st->parser= NULL;
}
}
/* estimate the end time (duration) */
/* XXX: may need to support wrapping */
filesize = ic->pb ? avio_size(ic->pb) : 0;//得到文件大小
end_time = AV_NOPTS_VALUE;
do{
offset = filesize - (DURATION_MAX_READ_SIZE<<retry);
if (offset < 0)
offset = 0;
avio_seek(ic->pb, offset, SEEK_SET);//尽量往后查找，pts越靠近文件末尾，利用pts估计时长越准确
read_size = 0;
for(;;) {
if (read_size >= DURATION_MAX_READ_SIZE<<(FFMAX(retry-1,0)))
break;
do {
ret = ff_read_packet(ic, pkt);//从接近文件末尾的地方读取数据，直到最后一个合法的数据包
} while(ret == AVERROR(EAGAIN));
if (ret != 0)
break;
read_size += pkt->size;
st = ic->streams[pkt->stream_index];
if (pkt->pts != AV_NOPTS_VALUE &&
(st->start_time != AV_NOPTS_VALUE ||
st->first_dts != AV_NOPTS_VALUE)) {
duration = end_time = pkt->pts;//利用该包的pts数据得到比较接近的时长
if (st->start_time != AV_NOPTS_VALUE)
duration -= st->start_time;//减去初始时间
else
duration -= st->first_dts;
if (duration > 0) {
if (st->duration == AV_NOPTS_VALUE || st->duration < duration)
st->duration = duration;
}
}
av_free_packet(pkt);
}
}while( end_time==AV_NOPTS_VALUE
&& filesize > (DURATION_MAX_READ_SIZE<<retry)
&& ++retry <= DURATION_MAX_RETRY);//尝试 DURATION_MAX_RETRY这么多次
}
方法2：通过文件大小和码流来估计
static void estimate_timings_from_bit_rate(AVFormatContext *ic)
{
int64_t filesize, duration;
int bit_rate, i;
AVStream *st;
/* if bit_rate is already set, we believe it */
if (ic->bit_rate <= 0) {
bit_rate = 0;
for(i=0;i<ic->nb_streams;i++) {//通过累积各个子流的平均码率得到文件的平均码率
st = ic->streams[i];
if (st->codec->bit_rate > 0)
bit_rate += st->codec->bit_rate;
}
ic->bit_rate = bit_rate;
}
/* if duration is already set, we believe it */
if (ic->duration == AV_NOPTS_VALUE &&
ic->bit_rate != 0) {
filesize = ic->pb ? avio_size(ic->pb) : 0;
if (filesize > 0) {
for(i = 0; i < ic->nb_streams; i++) {
st = ic->streams[i];
duration= av_rescale(8*filesize, st->time_base.den, ic->bit_rate*(int64_t)st->time_base.num);//通过文件大小除以文件平均码率得到文件时长，之所以还有time_base信息，是因为最后要把秒转换为以time_base为单位的值
if (st->duration == AV_NOPTS_VALUE)
st->duration = duration;
}
}
}
}
对应的，也可以在android stagefright中加入类似的实现：
uint64_t MPEG2TSExtractor::estimateDuration() {
Mutex::Autolock autoLock(mLock);
int64_t filesize;
int64_t end_time;
int64_t offset, duration;
int retry=1;
status_t re=mDataSource->getSize(&filesize);//android中有类似的函数去得到文件的大小
if (re != OK) {
ALOGE("Failed to get file size");
return ERROR_MALFORMED;
}
uint8_t packet[kTSPacketSize];
unsigned payload_unit_start_indicator = 0;
unsigned PID = 0;
unsigned adaptation_field_control = 0;
//实现思想：从文件末尾开始读取188个字节，直到找到第一个pes包的边界，并且跳过adp filed的包，这里认为adp field不含有合法的pts
while (1) {
offset = filesize - kTSPacketSize*retry;
ssize_t n = mDataSource->readAt(offset, packet, kTSPacketSize);
if (n < (ssize_t)kTSPacketSize) {
return (n < 0) ? (status_t)n : ERROR_END_OF_STREAM;
}
ABitReader* br= new ABitReader((const uint8_t *)packet, kTSPacketSize);//主要此类实现了一个bit读取器，对于码流的解析非常方便，类似于ffmpeg中的get_bits.h中实现的功能
unsigned sync_byte = br->getBits(8);
CHECK_EQ(sync_byte, 0x47u);
br->skipBits(1);
unsigned payload_unit_start_indicator = br->getBits(1);
br->skipBits(1);
PID = br->getBits(13);
br->skipBits(2);
adaptation_field_control = br->getBits(2);
if ((payload_unit_start_indicator == 1) && (adaptation_field_control == 1) &&
(PID != 0x00u) && (PID != 0x01u) && (PID != 0x02u) ) {
break;
}
retry++;
delete br;
} ;
ABitReader* br= new ABitReader((const uint8_t *)packet, kTSPacketSize);
br->skipBits(8 + 3 + 13);
br->skipBits(2);
adaptation_field_control = br->getBits(2);
ALOGV("adaptation_field_control = %u", adaptation_field_control);
br->skipBits(4);
if (adaptation_field_control == 2 || adaptation_field_control == 3) {
unsigned adaptation_field_length = br->getBits(8);
if (adaptation_field_length > 0) {
br->skipBits(adaptation_field_length * 8);
}
ALOGV("adaptation_field_length = %u", adaptation_field_length);
}
if (adaptation_field_control == 1 || adaptation_field_control == 3) {
unsigned packet_startcode_prefix = br->getBits(24);
ALOGV("packet_startcode_prefix = 0x%08x", packet_startcode_prefix);
CHECK_EQ(packet_startcode_prefix, 0x000001u);
unsigned stream_id = br->getBits(8);
ALOGV("stream_id = 0x%02x", stream_id);
br->skipBits(16);
//以下可以参考标准，标准上解释的很详细，应该不难理解
if (stream_id != 0xbc // program_stream_map
&& stream_id != 0xbe // padding_stream
&& stream_id != 0xbf // private_stream_2
&& stream_id != 0xf0 // ECM
&& stream_id != 0xf1 // EMM
&& stream_id != 0xff // program_stream_directory
&& stream_id != 0xf2 // DSMCC
&& stream_id != 0xf8) { // H.222.1 type E
CHECK_EQ(br->getBits(2), 2u);
br->skipBits(6);
unsigned PTS_DTS_flags = br->getBits(2);
ALOGV("PTS_DTS_flags = %u", PTS_DTS_flags);
br->skipBits(6);
unsigned PES_header_data_length = br->getBits(8);
unsigned optional_bytes_remaining = PES_header_data_length;
uint64_t PTS = 0, DTS = 0;
if (PTS_DTS_flags == 2 || PTS_DTS_flags == 3) {
CHECK_GE(optional_bytes_remaining, 5u);
CHECK_EQ(br->getBits(4), PTS_DTS_flags);
PTS = ((uint64_t)br->getBits(3)) << 30;
CHECK_EQ(br->getBits(1), 1u);
PTS |= ((uint64_t)br->getBits(15)) << 15;
CHECK_EQ(br->getBits(1), 1u);
PTS |= br->getBits(15);
CHECK_EQ(br->getBits(1), 1u);
ALOGV("PTS = %llu", PTS);
ALOGV("PTS = %.2f secs", PTS / 90000.0f);
PTS = (PTS * 1000 * 1000ll) / 90000;
return PTS;
}
// ES data follows.
}
}
delete br;
return 0;
}