[FFmpeg-devel] [PATCH v6 3/3] avformat/hlsenc:addition of CODECS attribute in the master playlist
Steven Liu
lq at chinaffmpeg.org
Thu Dec 14 16:53:14 EET 2017
> 在 2017年12月14日，下午6:55，vdixit at akamai.com 写道：
>
> From: Vishwanath Dixit <vdixit at akamai.com>
>
> ---
> libavformat/Makefile | 2 +-
> libavformat/dashenc.c | 2 +-
> libavformat/hlsenc.c | 65 +++++++++++++++++++++++++++++++++++++++++++++--
> libavformat/hlsplaylist.c | 5 +++-
> libavformat/hlsplaylist.h | 3 ++-
> libavformat/reverse.c | 1 +
> tests/ref/fate/source | 1 +
> 7 files changed, 73 insertions(+), 6 deletions(-)
> create mode 100644 libavformat/reverse.c
>
> diff --git a/libavformat/Makefile b/libavformat/Makefile
> index 734b703..b7e042d 100644
> --- a/libavformat/Makefile
> +++ b/libavformat/Makefile
> @@ -61,7 +61,7 @@ OBJS-$(CONFIG_RTPDEC) += rdt.o \
> rtpdec_vp9.o \
> rtpdec_xiph.o
> OBJS-$(CONFIG_RTPENC_CHAIN) += rtpenc_chain.o rtp.o
> -OBJS-$(CONFIG_SHARED) += log2_tab.o golomb_tab.o
> +OBJS-$(CONFIG_SHARED) += log2_tab.o golomb_tab.o reverse.o
> OBJS-$(CONFIG_SRTP) += srtp.o
>
> # muxers/demuxers
> diff --git a/libavformat/dashenc.c b/libavformat/dashenc.c
> index f363418..016ada3 100644
> --- a/libavformat/dashenc.c
> +++ b/libavformat/dashenc.c
> @@ -760,7 +760,7 @@ static int write_manifest(AVFormatContext *s, int final)
> AVStream *st = s->streams[i];
> get_hls_playlist_name(playlist_file, sizeof(playlist_file), NULL, i);
> ff_hls_write_stream_info(st, out, st->codecpar->bit_rate,
> - playlist_file, NULL);
> + playlist_file, NULL, NULL);
> }
> avio_close(out);
> if (use_rename)
> diff --git a/libavformat/hlsenc.c b/libavformat/hlsenc.c
> index 273dd8a..ed64847 100644
> --- a/libavformat/hlsenc.c
> +++ b/libavformat/hlsenc.c
> @@ -39,6 +39,7 @@
> #include "libavutil/avstring.h"
> #include "libavutil/intreadwrite.h"
> #include "libavutil/random_seed.h"
> +#include "libavutil/reverse.h"
> #include "libavutil/opt.h"
> #include "libavutil/log.h"
> #include "libavutil/time_internal.h"
> @@ -1078,6 +1079,63 @@ static int get_relative_url(const char *master_url, const char *media_url,
> return 0;
> }
>
> +static char *get_codec_str(AVStream *vid_st, AVStream *aud_st) {
> + size_t codec_str_size = 64;
> + char *codec_str = av_malloc(codec_str_size);
> + int video_str_len = 0;
> +
> + if (!codec_str)
> + return NULL;
> +
> + if (!vid_st && !aud_st) {
> + goto fail;
> + }
> +
> + if (vid_st) {
> + if (vid_st->codecpar->profile != FF_PROFILE_UNKNOWN &&
> + vid_st->codecpar->level != FF_LEVEL_UNKNOWN &&
> + vid_st->codecpar->codec_id == AV_CODEC_ID_H264) {
> + snprintf(codec_str, codec_str_size, "avc1.%02x%02x%02x",
> + vid_st->codecpar->profile & 0xFF,
> + ff_reverse[(vid_st->codecpar->profile >> 8) & 0xFF],
> + vid_st->codecpar->level);
> + } else {
> + goto fail;
> + }
> + video_str_len = strlen(codec_str);
> + }
> +
> + if (aud_st) {
> + char *audio_str = codec_str;
> + if (video_str_len) {
> + codec_str[video_str_len] = ',';
> + video_str_len += 1;
> + audio_str += video_str_len;
> + codec_str_size -= video_str_len;
> + }
> + if (aud_st->codecpar->codec_id == AV_CODEC_ID_MP2) {
> + snprintf(audio_str, codec_str_size, "mp4a.40.33");
> + } else if (aud_st->codecpar->codec_id == AV_CODEC_ID_MP3) {
> + snprintf(audio_str, codec_str_size, "mp4a.40.34");
> + } else if (aud_st->codecpar->codec_id == AV_CODEC_ID_AAC) {
> + /* TODO : For HE-AAC, HE-AACv2, the last digit needs to be set to 5 and 29 respectively */
> + snprintf(audio_str, codec_str_size, "mp4a.40.2");
> + } else if (aud_st->codecpar->codec_id == AV_CODEC_ID_AC3) {
> + snprintf(audio_str, codec_str_size, "mp4a.A5");
> + } else if (aud_st->codecpar->codec_id == AV_CODEC_ID_EAC3) {
> + snprintf(audio_str, codec_str_size, "mp4a.A6");
> + } else {
> + goto fail;
> + }
> + }
> +
> + return codec_str;
> +
> +fail:
> + av_free(codec_str);
> + return NULL;
> +}
> +
> static int create_master_playlist(AVFormatContext *s,
> VariantStream * const input_vs)
> {
> @@ -1088,7 +1146,7 @@ static int create_master_playlist(AVFormatContext *s,
> AVDictionary *options = NULL;
> unsigned int i, j;
> int m3u8_name_size, ret, bandwidth;
> - char *m3u8_rel_name;
> + char *m3u8_rel_name, *codec_str;
>
> input_vs->m3u8_created = 1;
> if (!hls->master_m3u8_created) {
> @@ -1202,9 +1260,12 @@ static int create_master_playlist(AVFormatContext *s,
> bandwidth += aud_st->codecpar->bit_rate;
> bandwidth += bandwidth / 10;
>
> + codec_str = get_codec_str(vid_st, aud_st);
> +
> ff_hls_write_stream_info(vid_st, master_pb, bandwidth, m3u8_rel_name,
> - aud_st ? vs->agroup : NULL);
> + codec_str, aud_st ? vs->agroup : NULL);
>
> + av_freep(&codec_str);
> av_freep(&m3u8_rel_name);
> }
> fail:
> diff --git a/libavformat/hlsplaylist.c b/libavformat/hlsplaylist.c
> index 42f059a..b1b1ec6 100644
> --- a/libavformat/hlsplaylist.c
> +++ b/libavformat/hlsplaylist.c
> @@ -36,7 +36,8 @@ void ff_hls_write_playlist_version(AVIOContext *out, int version) {
> }
>
> void ff_hls_write_stream_info(AVStream *st, AVIOContext *out,
> - int bandwidth, char *filename, char *agroup) {
> + int bandwidth, char *filename, char *codec_str,
> + char *agroup) {
> if (!out || !filename)
> return;
>
> @@ -50,6 +51,8 @@ void ff_hls_write_stream_info(AVStream *st, AVIOContext *out,
> if (st && st->codecpar->width > 0 && st->codecpar->height > 0)
> avio_printf(out, ",RESOLUTION=%dx%d", st->codecpar->width,
> st->codecpar->height);
> + if (codec_str && strlen(codec_str) > 0)
> + avio_printf(out, ",CODECS=\"%s\"", codec_str);
> if (agroup && strlen(agroup) > 0)
> avio_printf(out, ",AUDIO=\"group_%s\"", agroup);
> avio_printf(out, "\n%s\n\n", filename);
> diff --git a/libavformat/hlsplaylist.h b/libavformat/hlsplaylist.h
> index a3ce26c..e807c6e 100644
> --- a/libavformat/hlsplaylist.h
> +++ b/libavformat/hlsplaylist.h
> @@ -43,7 +43,8 @@ static inline int hls_get_int_from_double(double val)
>
> void ff_hls_write_playlist_version(AVIOContext *out, int version);
> void ff_hls_write_stream_info(AVStream *st, AVIOContext *out,
> - int bandwidth, char *filename, char *agroup);
> + int bandwidth, char *filename, char *codec_str,
> + char *agroup);
> void ff_hls_write_playlist_header(AVIOContext *out, int version, int allowcache,
> int target_duration, int64_t sequence,
> uint32_t playlist_type);
> diff --git a/libavformat/reverse.c b/libavformat/reverse.c
> new file mode 100644
> index 0000000..440bada
> --- /dev/null
> +++ b/libavformat/reverse.c
> @@ -0,0 +1 @@
> +#include "libavutil/reverse.c"
> diff --git a/tests/ref/fate/source b/tests/ref/fate/source
> index 2def034..b68873b 100644
> --- a/tests/ref/fate/source
> +++ b/tests/ref/fate/source
> @@ -11,6 +11,7 @@ libavfilter/log2_tab.c
> libavformat/file_open.c
> libavformat/golomb_tab.c
> libavformat/log2_tab.c
> +libavformat/reverse.c
this need double check for a better way
> libswresample/log2_tab.c
> libswscale/log2_tab.c
> tools/uncoded_frame.c
> --
> 1.9.1
>
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Thanks
Steven
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