Entries are sorted chronologically from oldest to youngest within each release,
releases are sorted from youngest to oldest.
-version 10:
-- av_strnstr
-- support ID3v2 tags in ASF files
+version <next>
+
+- HNM version 4 demuxer and video decoder
++- Live HDS muxer
+
+
+version 2.1:
+
+- aecho filter
+- perspective filter ported from libmpcodecs
+- ffprobe -show_programs option
+- compand filter
+- RTMP seek support
+- when transcoding with ffmpeg (i.e. not streamcopying), -ss is now accurate
+ even when used as an input option. Previous behavior can be restored with
+ the -noaccurate_seek option.
+- ffmpeg -t option can now be used for inputs, to limit the duration of
+ data read from an input file
+- incomplete Voxware MetaSound decoder
+- read EXIF metadata from JPEG
+- DVB teletext decoder
+- phase filter ported from libmpcodecs
+- w3fdif filter
+- Opus support in Matroska
+- FFV1 version 1.3 is stable and no longer experimental
+- FFV1: YUVA(444,422,420) 9, 10 and 16 bit support
+- changed DTS stream id in lavf mpeg ps muxer from 0x8a to 0x88, to be
+ more consistent with other muxers.
+- adelay filter
+- pullup filter ported from libmpcodecs
+- ffprobe -read_intervals option
+- Lossless and alpha support for WebP decoder
+- Error Resilient AAC syntax (ER AAC LC) decoding
+- Low Delay AAC (ER AAC LD) decoding
+- mux chapters in ASF files
+- SFTP protocol (via libssh)
+- libx264: add ability to encode in YUVJ422P and YUVJ444P
+- Fraps: use BT.709 colorspace by default for yuv, as reference fraps decoder does
+- make decoding alpha optional for prores, ffv1 and vp6 by setting
+ the skip_alpha flag.
+- ladspa wrapper filter
+- native VP9 decoder
+- dpx parser
+- max_error_rate parameter in ffmpeg
+- PulseAudio output device
+- ReplayGain scanner
+- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
+- Linux framebuffer output device
+- HEVC decoder
+- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
+- mergeplanes filter
+
+
+version 2.0:
+
+- curves filter
- reference-counting for AVFrame and AVPacket data
-- avconv now fails when input options are used for output file
+- ffmpeg now fails when input options are used for output file
or vice versa
-- new avconv options -filter_script and -filter_complex_script, which allow a
+- support for Monkey's Audio versions from 3.93
+- perms and aperms filters
+- audio filtering support in ffplay
+- 10% faster aac encoding on x86 and MIPS
+- sine audio filter source
+- WebP demuxing and decoding support
+- new ffmpeg options -filter_script and -filter_complex_script, which allow a
filtergraph description to be read from a file
+- OpenCL support
+- audio phaser filter
+- separatefields filter
+- libquvi demuxer
- uniform options syntax across all filters
+- telecine filter
- new interlace filter
-- JPEG 2000 decoder
-- new asetpts filter (same as setpts, but for audio)
-- new trim and atrim filters
-- avconv -t and -ss (output-only) options are now sample-accurate when
+- smptehdbars source
+- inverse telecine filters (fieldmatch and decimate)
+- colorbalance filter
+- colorchannelmixer filter
+- The matroska demuxer can now output proper verbatim ASS packets. It will
+ become the default at the next libavformat major bump.
+- decent native animated GIF encoding
+- asetrate filter
+- interleave filter
+- timeline editing with filters
+- vidstabdetect and vidstabtransform filters for video stabilization using
+ the vid.stab library
+- astats filter
+- trim and atrim filters
+- ffmpeg -t and -ss (output-only) options are now sample-accurate when
transcoding audio
- Matroska muxer can now put the index at the beginning of the file.
-- avconv -deinterlace option removed, the yadif filter should be used instead
+- extractplanes filter
+- avectorscope filter
+- ADPCM DTK decoder
+- ADP demuxer
+- RSD demuxer
+- RedSpark demuxer
+- ADPCM IMA Radical decoder
+- zmq filters
+- DCT denoiser filter (dctdnoiz)
+- Wavelet denoiser filter ported from libmpcodecs as owdenoise (formerly "ow")
- Apple Intermediate Codec decoder
- Escape 130 video decoder
+- FTP protocol support
+- V4L2 output device
+- 3D LUT filter (lut3d)
+- SMPTE 302M audio encoder
- support for slice multithreading in libavfilter
+- Hald CLUT support (generation and filtering)
- VC-1 interlaced B-frame support
- support for WavPack muxing (raw and in Matroska)
+- XVideo output device
+- vignette filter
+- True Audio (TTA) encoder
- Go2Webinar decoder
+- mcdeint filter ported from libmpcodecs
+- sab filter ported from libmpcodecs
+- ffprobe -show_chapters option
- WavPack encoding through libwavpack
-- Added the -n parameter to avconv
-- RTMP seek support
-- when transcoding with avconv (i.e. not streamcopying), -ss is now accurate
- even when used as an input option. Previous behavior can be restored with
- the -noaccurate_seek option.
-- avconv -t option can now be used for inputs, to limit the duration of
- data read from an input file
-- incomplete Voxware MetaSound decoder
-- WebP decoder
-- Error Resilient AAC syntax (ER AAC LC) decoding
-- Low Delay AAC (ER AAC LD) decoding
-- mux chapters in ASF files
-- Opus in Ogg demuxing
-- Enhanced Low Delay AAC (ER AAC ELD) decoding (no LD SBR support)
-- F4V muxer
-- HNM version 4 demuxer and video decoder
-- HEVC decoder
-- raw HEVC, HEVC in MOV/MP4, HEVC in Matroska, HEVC in MPEG-TS demuxing
-- remove avplay -vismv option, which has not worked for a long time
-- Live HDS muxer
+- rotate filter
+- spp filter ported from libmpcodecs
+- libgme support
+- psnr filter
-version 9:
-- av_basename and av_dirname
-- adobe and limelight publisher authentication in RTMP
+version 1.2:
+
- VDPAU hardware acceleration through normal hwaccel
- SRTP support
+- Error diffusion dither in Swscale
+- Chained Ogg support
+- Theora Midstream reconfiguration support
+- EVRC decoder
+- audio fade filter
+- filtering audio with unknown channel layout
+- allpass, bass, bandpass, bandreject, biquad, equalizer, highpass, lowpass
+ and treble audio filter
+- improved showspectrum filter, with multichannel support and sox-like colors
+- histogram filter
+- tee muxer
+- il filter ported from libmpcodecs
+- support ID3v2 tags in ASF files
+- encrypted TTA stream decoding support
+- RF64 support in WAV muxer
+- noise filter ported from libmpcodecs
+- Subtitles character encoding conversion
+- blend filter
+- stereo3d filter ported from libmpcodecs
+
+version 1.1:
-version 9_beta3:
-- ashowinfo audio filter
+- stream disposition information printing in ffprobe
+- filter for loudness analysis following EBU R128
+- Opus encoder using libopus
+- ffprobe -select_streams option
+- Pinnacle TARGA CineWave YUV16 decoder
+- TAK demuxer, decoder and parser
+- DTS-HD demuxer
+- remove -same_quant, it hasn't worked for years
+- FFM2 support
+- X-Face image encoder and decoder
- 24-bit FLAC encoding
-- audio volume filter
-- deprecated the avconv -vol option. the volume filter is to be used instead.
- multi-channel ALAC encoding up to 7.1
-- TAK demuxer, parser, and decoder
-- adaptive frame-level multithreading for H.264
-
-
-version 9_beta2:
- metadata (INFO tag) support in WAV muxer
+- subtitles raw text decoder
- support for building DLLs using MSVC
-- remove avserver daemon mode
+- LVF demuxer
+- ffescape tool
+- metadata (info chunk) support in CAF muxer
+- field filter ported from libmpcodecs
+- AVR demuxer
+- geq filter ported from libmpcodecs
+- remove ffserver daemon mode
+- AST muxer/demuxer
+- new expansion syntax for drawtext
+- BRender PIX image decoder
+- ffprobe -show_entries option
+- ffprobe -sections option
+- ADPCM IMA Dialogic decoder
+- BRSTM demuxer
+- animated GIF decoder and demuxer
+- PVF demuxer
+- subtitles filter
+- IRCAM muxer/demuxer
+- Paris Audio File demuxer
+- Virtual concatenation demuxer
+- VobSub demuxer
+- JSON captions for TED talks decoding support
+- SOX Resampler support in libswresample
+- aselect filter
+- SGI RLE 8-bit decoder
+- Silicon Graphics Motion Video Compressor 1 & 2 decoder
+- Silicon Graphics Movie demuxer
+- apad filter
+- Resolution & pixel format change support with multithreading for H.264
+- documentation split into per-component manuals
+- pp (postproc) filter ported from MPlayer
+- NIST Sphere demuxer
+- MPL2, VPlayer, MPlayer, AQTitle, PJS and SubViewer v1 subtitles demuxers and decoders
+- Sony Wave64 muxer
+- adobe and limelight publisher authentication in RTMP
+- data: URI scheme
- support building on the Plan 9 operating system
-- ffv1: support version 1.3
+- kerndeint filter ported from MPlayer
+- histeq filter ported from VirtualDub
+- Megalux Frame demuxer
+- 012v decoder
+- Improved AVC Intra decoding support
-version 9_beta1:
+version 1.0:
-- XWD encoder and decoder
-- Support for fragmentation in the mov/mp4 muxer
-- ISMV (Smooth Streaming) muxer
-- CDXL demuxer and decoder
-- Apple ProRes encoder
-- Sun Rasterfile Encoder
-- remove libpostproc
-- ID3v2 attached pictures reading and writing
-- WMA Lossless decoder
-- XBM encoder
-- RealAudio Lossless decoder
-- ZeroCodec decoder
-- drop support for avconv without libavfilter
-- add libavresample audio conversion library
-- audio filters support in libavfilter and avconv
-- add fps filter
-- audio split filter
-- audio mix filter
-- avprobe output is now standard INI or JSON. The old format can still
- be used with -of old.
+- INI and flat output in ffprobe
+- Scene detection in libavfilter
- Indeo Audio decoder
- channelsplit audio filter
+- setnsamples audio filter
+- atempo filter
+- ffprobe -show_data option
- RTMPT protocol support
- iLBC encoding/decoding via libilbc
- Microsoft Screen 1 decoder
--- /dev/null
- * This file is part of Libav.
+ /*
+ * Live HDS fragmenter
+ * Copyright (c) 2013 Martin Storsjo
+ *
- * Libav is free software; you can redistribute it and/or
++ * This file is part of FFmpeg.
+ *
- * Libav is distributed in the hope that it will be useful,
++ * 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.
+ *
- * License along with Libav; if not, write to the Free Software
++ * 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 "config.h"
+ #include <float.h>
+ #if HAVE_UNISTD_H
+ #include <unistd.h>
+ #endif
+
+ #include "avformat.h"
+ #include "internal.h"
+ #include "os_support.h"
+
+ #include "libavutil/avstring.h"
+ #include "libavutil/base64.h"
+ #include "libavutil/intreadwrite.h"
+ #include "libavutil/mathematics.h"
+ #include "libavutil/opt.h"
+
+ typedef struct Fragment {
+ char file[1024];
+ int64_t start_time, duration;
+ int n;
+ } Fragment;
+
+ typedef struct OutputStream {
+ int bitrate;
+ int first_stream;
+ AVFormatContext *ctx;
+ int ctx_inited;
+ uint8_t iobuf[32768];
+ char temp_filename[1024];
+ int64_t frag_start_ts, last_ts;
+ AVIOContext *out;
+ int packets_written;
+ int nb_fragments, fragments_size, fragment_index;
+ Fragment **fragments;
+
+ int has_audio, has_video;
+
+ uint8_t *metadata;
+ int metadata_size;
+
+ uint8_t *extra_packets[2];
+ int extra_packet_sizes[2];
+ int nb_extra_packets;
+ } OutputStream;
+
+ typedef struct HDSContext {
+ const AVClass *class; /* Class for private options. */
+ int window_size;
+ int extra_window_size;
+ int min_frag_duration;
+ int remove_at_exit;
+
+ OutputStream *streams;
+ int nb_streams;
+ } HDSContext;
+
+ static int parse_header(OutputStream *os, const uint8_t *buf, int buf_size)
+ {
+ if (buf_size < 13)
+ return AVERROR_INVALIDDATA;
+ if (memcmp(buf, "FLV", 3))
+ return AVERROR_INVALIDDATA;
+ buf += 13;
+ buf_size -= 13;
+ while (buf_size >= 11 + 4) {
+ int type = buf[0];
+ int size = AV_RB24(&buf[1]) + 11 + 4;
+ if (size > buf_size)
+ return AVERROR_INVALIDDATA;
+ if (type == 8 || type == 9) {
+ if (os->nb_extra_packets > FF_ARRAY_ELEMS(os->extra_packets))
+ return AVERROR_INVALIDDATA;
+ os->extra_packet_sizes[os->nb_extra_packets] = size;
+ os->extra_packets[os->nb_extra_packets] = av_malloc(size);
+ if (!os->extra_packets[os->nb_extra_packets])
+ return AVERROR(ENOMEM);
+ memcpy(os->extra_packets[os->nb_extra_packets], buf, size);
+ os->nb_extra_packets++;
+ } else if (type == 0x12) {
+ if (os->metadata)
+ return AVERROR_INVALIDDATA;
+ os->metadata_size = size - 11 - 4;
+ os->metadata = av_malloc(os->metadata_size);
+ if (!os->metadata)
+ return AVERROR(ENOMEM);
+ memcpy(os->metadata, buf + 11, os->metadata_size);
+ }
+ buf += size;
+ buf_size -= size;
+ }
+ if (!os->metadata)
+ return AVERROR_INVALIDDATA;
+ return 0;
+ }
+
+ static int hds_write(void *opaque, uint8_t *buf, int buf_size)
+ {
+ OutputStream *os = opaque;
+ if (os->out) {
+ avio_write(os->out, buf, buf_size);
+ } else {
+ if (!os->metadata_size) {
+ int ret;
+ // Assuming the IO buffer is large enough to fit the
+ // FLV header and all metadata and extradata packets
+ if ((ret = parse_header(os, buf, buf_size)) < 0)
+ return ret;
+ }
+ }
+ return buf_size;
+ }
+
+ static void hds_free(AVFormatContext *s)
+ {
+ HDSContext *c = s->priv_data;
+ int i, j;
+ if (!c->streams)
+ return;
+ for (i = 0; i < s->nb_streams; i++) {
+ OutputStream *os = &c->streams[i];
+ if (os->out)
+ avio_close(os->out);
+ os->out = NULL;
+ if (os->ctx && os->ctx_inited)
+ av_write_trailer(os->ctx);
+ if (os->ctx && os->ctx->pb)
+ av_free(os->ctx->pb);
+ if (os->ctx)
+ avformat_free_context(os->ctx);
+ av_free(os->metadata);
+ for (j = 0; j < os->nb_extra_packets; j++)
+ av_free(os->extra_packets[j]);
+ for (j = 0; j < os->nb_fragments; j++)
+ av_free(os->fragments[j]);
+ av_free(os->fragments);
+ }
+ av_freep(&c->streams);
+ }
+
+ static int write_manifest(AVFormatContext *s, int final)
+ {
+ HDSContext *c = s->priv_data;
+ AVIOContext *out;
+ char filename[1024], temp_filename[1024];
+ int ret, i;
+ float duration = 0;
+
+ if (c->nb_streams > 0)
+ duration = c->streams[0].last_ts * av_q2d(s->streams[0]->time_base);
+
+ snprintf(filename, sizeof(filename), "%s/index.f4m", s->filename);
+ snprintf(temp_filename, sizeof(temp_filename), "%s/index.f4m.tmp", s->filename);
+ ret = avio_open2(&out, temp_filename, AVIO_FLAG_WRITE,
+ &s->interrupt_callback, NULL);
+ if (ret < 0) {
+ av_log(s, AV_LOG_ERROR, "Unable to open %s for writing\n", filename);
+ return ret;
+ }
+ avio_printf(out, "<?xml version=\"1.0\" encoding=\"utf-8\"?>\n");
+ avio_printf(out, "<manifest xmlns=\"http://ns.adobe.com/f4m/1.0\">\n");
+ avio_printf(out, "\t<id>%s</id>\n", av_basename(s->filename));
+ avio_printf(out, "\t<streamType>%s</streamType>\n",
+ final ? "recorded" : "live");
+ avio_printf(out, "\t<deliveryType>streaming</deliveryType>\n");
+ if (final)
+ avio_printf(out, "\t<duration>%f</duration>\n", duration);
+ for (i = 0; i < c->nb_streams; i++) {
+ OutputStream *os = &c->streams[i];
+ int b64_size = AV_BASE64_SIZE(os->metadata_size);
+ char *base64 = av_malloc(b64_size);
+ if (!base64) {
+ avio_close(out);
+ return AVERROR(ENOMEM);
+ }
+ av_base64_encode(base64, b64_size, os->metadata, os->metadata_size);
+
+ avio_printf(out, "\t<bootstrapInfo profile=\"named\" url=\"stream%d.abst\" id=\"bootstrap%d\" />\n", i, i);
+ avio_printf(out, "\t<media bitrate=\"%d\" url=\"stream%d\" bootstrapInfoId=\"bootstrap%d\">\n", os->bitrate/1000, i, i);
+ avio_printf(out, "\t\t<metadata>%s</metadata>\n", base64);
+ avio_printf(out, "\t</media>\n");
+ av_free(base64);
+ }
+ avio_printf(out, "</manifest>\n");
+ avio_flush(out);
+ avio_close(out);
+ rename(temp_filename, filename);
+ return 0;
+ }
+
+ static void update_size(AVIOContext *out, int64_t pos)
+ {
+ int64_t end = avio_tell(out);
+ avio_seek(out, pos, SEEK_SET);
+ avio_wb32(out, end - pos);
+ avio_seek(out, end, SEEK_SET);
+ }
+
+ /* Note, the .abst files need to be served with the "binary/octet"
+ * mime type, otherwise at least the OSMF player can easily fail
+ * with "stream not found" when polling for the next fragment. */
+ static int write_abst(AVFormatContext *s, OutputStream *os, int final)
+ {
+ HDSContext *c = s->priv_data;
+ AVIOContext *out;
+ char filename[1024], temp_filename[1024];
+ int i, ret;
+ int64_t asrt_pos, afrt_pos;
+ int start = 0, fragments;
+ int index = s->streams[os->first_stream]->id;
+ int64_t cur_media_time = 0;
+ if (c->window_size)
+ start = FFMAX(os->nb_fragments - c->window_size, 0);
+ fragments = os->nb_fragments - start;
+ if (final)
+ cur_media_time = os->last_ts;
+ else if (os->nb_fragments)
+ cur_media_time = os->fragments[os->nb_fragments - 1]->start_time;
+
+ snprintf(filename, sizeof(filename),
+ "%s/stream%d.abst", s->filename, index);
+ snprintf(temp_filename, sizeof(temp_filename),
+ "%s/stream%d.abst.tmp", s->filename, index);
+ ret = avio_open2(&out, temp_filename, AVIO_FLAG_WRITE,
+ &s->interrupt_callback, NULL);
+ if (ret < 0) {
+ av_log(s, AV_LOG_ERROR, "Unable to open %s for writing\n", temp_filename);
+ return ret;
+ }
+ avio_wb32(out, 0); // abst size
+ avio_wl32(out, MKTAG('a','b','s','t'));
+ avio_wb32(out, 0); // version + flags
+ avio_wb32(out, os->fragment_index - 1); // BootstrapinfoVersion
+ avio_w8(out, final ? 0 : 0x20); // profile, live, update
+ avio_wb32(out, 1000); // timescale
+ avio_wb64(out, cur_media_time);
+ avio_wb64(out, 0); // SmpteTimeCodeOffset
+ avio_w8(out, 0); // MovieIdentifer (null string)
+ avio_w8(out, 0); // ServerEntryCount
+ avio_w8(out, 0); // QualityEntryCount
+ avio_w8(out, 0); // DrmData (null string)
+ avio_w8(out, 0); // MetaData (null string)
+ avio_w8(out, 1); // SegmentRunTableCount
+ asrt_pos = avio_tell(out);
+ avio_wb32(out, 0); // asrt size
+ avio_wl32(out, MKTAG('a','s','r','t'));
+ avio_wb32(out, 0); // version + flags
+ avio_w8(out, 0); // QualityEntryCount
+ avio_wb32(out, 1); // SegmentRunEntryCount
+ avio_wb32(out, 1); // FirstSegment
+ avio_wb32(out, final ? (os->fragment_index - 1) : 0xffffffff); // FragmentsPerSegment
+ update_size(out, asrt_pos);
+ avio_w8(out, 1); // FragmentRunTableCount
+ afrt_pos = avio_tell(out);
+ avio_wb32(out, 0); // afrt size
+ avio_wl32(out, MKTAG('a','f','r','t'));
+ avio_wb32(out, 0); // version + flags
+ avio_wb32(out, 1000); // timescale
+ avio_w8(out, 0); // QualityEntryCount
+ avio_wb32(out, fragments); // FragmentRunEntryCount
+ for (i = start; i < os->nb_fragments; i++) {
+ avio_wb32(out, os->fragments[i]->n);
+ avio_wb64(out, os->fragments[i]->start_time);
+ avio_wb32(out, os->fragments[i]->duration);
+ }
+ update_size(out, afrt_pos);
+ update_size(out, 0);
+ avio_close(out);
+ rename(temp_filename, filename);
+ return 0;
+ }
+
+ static int init_file(AVFormatContext *s, OutputStream *os, int64_t start_ts)
+ {
+ int ret, i;
+ ret = avio_open2(&os->out, os->temp_filename, AVIO_FLAG_WRITE,
+ &s->interrupt_callback, NULL);
+ if (ret < 0)
+ return ret;
+ avio_wb32(os->out, 0);
+ avio_wl32(os->out, MKTAG('m','d','a','t'));
+ for (i = 0; i < os->nb_extra_packets; i++) {
+ AV_WB24(os->extra_packets[i] + 4, start_ts);
+ os->extra_packets[i][7] = (start_ts >> 24) & 0x7f;
+ avio_write(os->out, os->extra_packets[i], os->extra_packet_sizes[i]);
+ }
+ return 0;
+ }
+
+ static void close_file(OutputStream *os)
+ {
+ int64_t pos = avio_tell(os->out);
+ avio_seek(os->out, 0, SEEK_SET);
+ avio_wb32(os->out, pos);
+ avio_flush(os->out);
+ avio_close(os->out);
+ os->out = NULL;
+ }
+
+ static int hds_write_header(AVFormatContext *s)
+ {
+ HDSContext *c = s->priv_data;
+ int ret = 0, i;
+ AVOutputFormat *oformat;
+
+ mkdir(s->filename, 0777);
+
+ oformat = av_guess_format("flv", NULL, NULL);
+ if (!oformat) {
+ ret = AVERROR_MUXER_NOT_FOUND;
+ goto fail;
+ }
+
+ c->streams = av_mallocz(sizeof(*c->streams) * s->nb_streams);
+ if (!c->streams) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+
+ for (i = 0; i < s->nb_streams; i++) {
+ OutputStream *os = &c->streams[c->nb_streams];
+ AVFormatContext *ctx;
+ AVStream *st = s->streams[i];
+
+ if (!st->codec->bit_rate) {
+ av_log(s, AV_LOG_ERROR, "No bit rate set for stream %d\n", i);
+ ret = AVERROR(EINVAL);
+ goto fail;
+ }
+ if (st->codec->codec_type == AVMEDIA_TYPE_VIDEO) {
+ if (os->has_video) {
+ c->nb_streams++;
+ os++;
+ }
+ os->has_video = 1;
+ } else if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
+ if (os->has_audio) {
+ c->nb_streams++;
+ os++;
+ }
+ os->has_audio = 1;
+ } else {
+ av_log(s, AV_LOG_ERROR, "Unsupported stream type in stream %d\n", i);
+ ret = AVERROR(EINVAL);
+ goto fail;
+ }
+ os->bitrate += s->streams[i]->codec->bit_rate;
+
+ if (!os->ctx) {
+ os->first_stream = i;
+ ctx = avformat_alloc_context();
+ if (!ctx) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+ os->ctx = ctx;
+ ctx->oformat = oformat;
+ ctx->interrupt_callback = s->interrupt_callback;
+
+ ctx->pb = avio_alloc_context(os->iobuf, sizeof(os->iobuf),
+ AVIO_FLAG_WRITE, os,
+ NULL, hds_write, NULL);
+ if (!ctx->pb) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+ } else {
+ ctx = os->ctx;
+ }
+ s->streams[i]->id = c->nb_streams;
+
+ if (!(st = avformat_new_stream(ctx, NULL))) {
+ ret = AVERROR(ENOMEM);
+ goto fail;
+ }
+ avcodec_copy_context(st->codec, s->streams[i]->codec);
+ st->sample_aspect_ratio = s->streams[i]->sample_aspect_ratio;
+ }
+ if (c->streams[c->nb_streams].ctx)
+ c->nb_streams++;
+
+ for (i = 0; i < c->nb_streams; i++) {
+ OutputStream *os = &c->streams[i];
+ int j;
+ if ((ret = avformat_write_header(os->ctx, NULL)) < 0) {
+ goto fail;
+ }
+ os->ctx_inited = 1;
+ avio_flush(os->ctx->pb);
+ for (j = 0; j < os->ctx->nb_streams; j++)
+ s->streams[os->first_stream + j]->time_base = os->ctx->streams[j]->time_base;
+
+ snprintf(os->temp_filename, sizeof(os->temp_filename),
+ "%s/stream%d_temp", s->filename, i);
+ init_file(s, os, 0);
+
+ if (!os->has_video && c->min_frag_duration <= 0) {
+ av_log(s, AV_LOG_WARNING,
+ "No video stream in output stream %d and no min frag duration set\n", i);
+ ret = AVERROR(EINVAL);
+ }
+ os->fragment_index = 1;
+ write_abst(s, os, 0);
+ }
+ ret = write_manifest(s, 0);
+
+ fail:
+ if (ret)
+ hds_free(s);
+ return ret;
+ }
+
+ static int add_fragment(OutputStream *os, const char *file,
+ int64_t start_time, int64_t duration)
+ {
+ Fragment *frag;
+ if (duration == 0)
+ duration = 1;
+ if (os->nb_fragments >= os->fragments_size) {
+ int ret;
+ os->fragments_size = (os->fragments_size + 1) * 2;
+ if ((ret = av_reallocp_array(&os->fragments, os->fragments_size,
+ sizeof(*os->fragments))) < 0) {
+ os->fragments_size = 0;
+ os->nb_fragments = 0;
+ return ret;
+ }
+ }
+ frag = av_mallocz(sizeof(*frag));
+ if (!frag)
+ return AVERROR(ENOMEM);
+ av_strlcpy(frag->file, file, sizeof(frag->file));
+ frag->start_time = start_time;
+ frag->duration = duration;
+ frag->n = os->fragment_index;
+ os->fragments[os->nb_fragments++] = frag;
+ os->fragment_index++;
+ return 0;
+ }
+
+ static int hds_flush(AVFormatContext *s, OutputStream *os, int final,
+ int64_t end_ts)
+ {
+ HDSContext *c = s->priv_data;
+ int i, ret = 0;
+ char target_filename[1024];
+ int index = s->streams[os->first_stream]->id;
+
+ if (!os->packets_written)
+ return 0;
+
+ avio_flush(os->ctx->pb);
+ os->packets_written = 0;
+ close_file(os);
+
+ snprintf(target_filename, sizeof(target_filename),
+ "%s/stream%dSeg1-Frag%d", s->filename, index, os->fragment_index);
+ rename(os->temp_filename, target_filename);
+ add_fragment(os, target_filename, os->frag_start_ts, end_ts - os->frag_start_ts);
+
+ if (!final) {
+ ret = init_file(s, os, end_ts);
+ if (ret < 0)
+ return ret;
+ }
+
+ if (c->window_size || (final && c->remove_at_exit)) {
+ int remove = os->nb_fragments - c->window_size - c->extra_window_size;
+ if (final && c->remove_at_exit)
+ remove = os->nb_fragments;
+ if (remove > 0) {
+ for (i = 0; i < remove; i++) {
+ unlink(os->fragments[i]->file);
+ av_free(os->fragments[i]);
+ }
+ os->nb_fragments -= remove;
+ memmove(os->fragments, os->fragments + remove,
+ os->nb_fragments * sizeof(*os->fragments));
+ }
+ }
+
+ if (ret >= 0)
+ ret = write_abst(s, os, final);
+ return ret;
+ }
+
+ static int hds_write_packet(AVFormatContext *s, AVPacket *pkt)
+ {
+ HDSContext *c = s->priv_data;
+ AVStream *st = s->streams[pkt->stream_index];
+ OutputStream *os = &c->streams[s->streams[pkt->stream_index]->id];
+ int64_t end_dts = (os->fragment_index) * c->min_frag_duration;
+ int ret;
+
+ if (st->first_dts == AV_NOPTS_VALUE)
+ st->first_dts = pkt->dts;
+
+ if ((!os->has_video || st->codec->codec_type == AVMEDIA_TYPE_VIDEO) &&
+ av_compare_ts(pkt->dts - st->first_dts, st->time_base,
+ end_dts, AV_TIME_BASE_Q) >= 0 &&
+ pkt->flags & AV_PKT_FLAG_KEY && os->packets_written) {
+
+ if ((ret = hds_flush(s, os, 0, pkt->dts)) < 0)
+ return ret;
+ }
+
+ // Note, these fragment start timestamps, that represent a whole
+ // OutputStream, assume all streams in it have the same time base.
+ if (!os->packets_written)
+ os->frag_start_ts = pkt->dts;
+ os->last_ts = pkt->dts;
+
+ os->packets_written++;
+ return ff_write_chained(os->ctx, pkt->stream_index - os->first_stream, pkt, s);
+ }
+
+ static int hds_write_trailer(AVFormatContext *s)
+ {
+ HDSContext *c = s->priv_data;
+ int i;
+
+ for (i = 0; i < c->nb_streams; i++)
+ hds_flush(s, &c->streams[i], 1, c->streams[i].last_ts);
+ write_manifest(s, 1);
+
+ if (c->remove_at_exit) {
+ char filename[1024];
+ snprintf(filename, sizeof(filename), "%s/index.f4m", s->filename);
+ unlink(filename);
+ for (i = 0; i < c->nb_streams; i++) {
+ snprintf(filename, sizeof(filename), "%s/stream%d.abst", s->filename, i);
+ unlink(filename);
+ }
+ rmdir(s->filename);
+ }
+
+ hds_free(s);
+ return 0;
+ }
+
+ #define OFFSET(x) offsetof(HDSContext, x)
+ #define E AV_OPT_FLAG_ENCODING_PARAM
+ static const AVOption options[] = {
+ { "window_size", "number of fragments kept in the manifest", OFFSET(window_size), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, INT_MAX, E },
+ { "extra_window_size", "number of fragments kept outside of the manifest before removing from disk", OFFSET(extra_window_size), AV_OPT_TYPE_INT, { .i64 = 5 }, 0, INT_MAX, E },
+ { "min_frag_duration", "minimum fragment duration (in microseconds)", OFFSET(min_frag_duration), AV_OPT_TYPE_INT64, { .i64 = 10000000 }, 0, INT_MAX, E },
+ { "remove_at_exit", "remove all fragments when finished", OFFSET(remove_at_exit), AV_OPT_TYPE_INT, { .i64 = 0 }, 0, 1, E },
+ { NULL },
+ };
+
+ static const AVClass hds_class = {
+ .class_name = "HDS muxer",
+ .item_name = av_default_item_name,
+ .option = options,
+ .version = LIBAVUTIL_VERSION_INT,
+ };
+
+ AVOutputFormat ff_hds_muxer = {
+ .name = "hds",
+ .long_name = NULL_IF_CONFIG_SMALL("HDS Muxer"),
+ .priv_data_size = sizeof(HDSContext),
+ .audio_codec = AV_CODEC_ID_AAC,
+ .video_codec = AV_CODEC_ID_H264,
+ .flags = AVFMT_GLOBALHEADER | AVFMT_NOFILE,
+ .write_header = hds_write_header,
+ .write_packet = hds_write_packet,
+ .write_trailer = hds_write_trailer,
+ .priv_class = &hds_class,
+ };
OSDN Git Service
