[FFmpeg-devel] [PATCH 091/281] matroska: convert to new channel layout API

[Andreas Rheinhardt]

James Almer:
> From: Vittorio Giovara <vittorio.giovara at gmail.com>
> 
> Signed-off-by: Vittorio Giovara <vittorio.giovara at gmail.com>
> Signed-off-by: Anton Khirnov <anton at khirnov.net>
> Signed-off-by: James Almer <jamrial at gmail.com>
> ---
>  libavformat/matroskadec.c |  8 ++++++--
>  libavformat/matroskaenc.c | 10 +++++-----
>  2 files changed, 11 insertions(+), 7 deletions(-)
> 
> diff --git a/libavformat/matroskadec.c b/libavformat/matroskadec.c
> index 78e5a4a203..62a36cbb8c 100644
> --- a/libavformat/matroskadec.c
> +++ b/libavformat/matroskadec.c
> @@ -2083,7 +2083,7 @@ static int matroska_parse_flac(AVFormatContext *s,
>                      av_log(s, AV_LOG_WARNING,
>                             "Invalid value of WAVEFORMATEXTENSIBLE_CHANNEL_MASK\n");
>                  } else
> -                    st->codecpar->channel_layout = mask;
> +                    av_channel_layout_from_mask(&st->codecpar->ch_layout, mask);
>              }
>              av_dict_free(&dict);
>          }
> @@ -2911,7 +2911,11 @@ static int matroska_parse_tracks(AVFormatContext *s)
>              st->codecpar->codec_type  = AVMEDIA_TYPE_AUDIO;
>              st->codecpar->codec_tag   = fourcc;
>              st->codecpar->sample_rate = track->audio.out_samplerate;
> -            st->codecpar->channels    = track->audio.channels;
> +            // channel layout may be already set by codec private checks above
> +            if (st->codecpar->ch_layout.order == AV_CHANNEL_ORDER_NATIVE &&
> +                !st->codecpar->ch_layout.u.mask)
> +                st->codecpar->ch_layout.order = AV_CHANNEL_ORDER_UNSPEC;
> +            st->codecpar->ch_layout.nb_channels = track->audio.channels;
>              if (!st->codecpar->bits_per_coded_sample)
>                  st->codecpar->bits_per_coded_sample = track->audio.bitdepth;
>              if (st->codecpar->codec_id == AV_CODEC_ID_MP3 ||
> diff --git a/libavformat/matroskaenc.c b/libavformat/matroskaenc.c
> index 152312102a..5278166e8c 100644
> --- a/libavformat/matroskaenc.c
> +++ b/libavformat/matroskaenc.c
> @@ -657,9 +657,9 @@ static int put_wv_codecpriv(AVIOContext *pb, const AVCodecParameters *par)
>  static int put_flac_codecpriv(AVFormatContext *s, AVIOContext *pb,
>                                const AVCodecParameters *par)
>  {
> -    int write_comment = (par->channel_layout &&
> -                         !(par->channel_layout & ~0x3ffffULL) &&
> -                         !ff_flac_is_native_layout(par->channel_layout));
> +    int write_comment = (par->ch_layout.order == AV_CHANNEL_ORDER_NATIVE && par->ch_layout.u.mask &&
> +                         !av_channel_layout_subset(&par->ch_layout, ~0x3ffffULL) &&

Why are you using this function although you already know that it is a
native layout, so that the typical & works fine?

> +                         !ff_flac_is_native_layout(par->ch_layout.u.mask));
>      int ret = ff_flac_write_header(pb, par->extradata, par->extradata_size,
>                                     !write_comment);
>  
> @@ -673,7 +673,7 @@ static int put_flac_codecpriv(AVFormatContext *s, AVIOContext *pb,
>          uint8_t buf[32];
>          int64_t len;
>  
> -        snprintf(buf, sizeof(buf), "0x%"PRIx64, par->channel_layout);
> +        snprintf(buf, sizeof(buf), "0x%"PRIx64, par->ch_layout.u.mask);
>          av_dict_set(&dict, "WAVEFORMATEXTENSIBLE_CHANNEL_MASK", buf, 0);
>  
>          len = ff_vorbiscomment_length(dict, vendor, NULL, 0);
> @@ -1396,7 +1396,7 @@ static int mkv_write_track(AVFormatContext *s, MatroskaMuxContext *mkv,
>              put_ebml_string(pb, MATROSKA_ID_CODECID, "A_MS/ACM");
>  
>          subinfo = start_ebml_master(pb, MATROSKA_ID_TRACKAUDIO, 6 + 4 * 9);
> -        put_ebml_uint  (pb, MATROSKA_ID_AUDIOCHANNELS    , par->channels);
> +        put_ebml_uint(pb, MATROSKA_ID_AUDIOCHANNELS, par->ch_layout.nb_channels);
>  
>          track->sample_rate_offset = avio_tell(pb);
>          put_ebml_float (pb, MATROSKA_ID_AUDIOSAMPLINGFREQ, sample_rate);
>