segafilm: convert to new channel layout API

Signed-off-by: James Almer <jamrial@gmail.com>

libavformat/segafilm.c

@@ -177,20 +177,20 @@ static int film_read_header(AVFormatContext *s)
         st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
         st->codecpar->codec_id = film->audio_type;
         st->codecpar->codec_tag = 1;
-        st->codecpar->channels = film->audio_channels;
+        st->codecpar->ch_layout.nb_channels = film->audio_channels;
         st->codecpar->sample_rate = film->audio_samplerate;
 
         if (film->audio_type == AV_CODEC_ID_ADPCM_ADX) {
             st->codecpar->bits_per_coded_sample = 18 * 8 / 32;
-            st->codecpar->block_align = st->codecpar->channels * 18;
+            st->codecpar->block_align = film->audio_channels * 18;
             ffstream(st)->need_parsing = AVSTREAM_PARSE_FULL;
         } else {
             st->codecpar->bits_per_coded_sample = film->audio_bits;
-            st->codecpar->block_align = st->codecpar->channels *
+            st->codecpar->block_align = film->audio_channels *
                 st->codecpar->bits_per_coded_sample / 8;
         }
 
-        st->codecpar->bit_rate = st->codecpar->channels * st->codecpar->sample_rate *
+        st->codecpar->bit_rate = film->audio_channels * st->codecpar->sample_rate *
             st->codecpar->bits_per_coded_sample;
     }
 

libavformat/segafilmenc.c

@@ -236,7 +236,7 @@ static int film_write_header(AVFormatContext *format_context)
         AVStream *audio = format_context->streams[film->audio_index];
         int audio_codec = get_audio_codec_id(audio->codecpar->codec_id);
 
-        bytestream_put_byte(&ptr, audio->codecpar->channels); /* Audio channels */
+        bytestream_put_byte(&ptr, audio->codecpar->ch_layout.nb_channels); /* Audio channels */
         bytestream_put_byte(&ptr, audio->codecpar->bits_per_coded_sample); /* Audio bit depth */
         bytestream_put_byte(&ptr, audio_codec); /* Compression - 0 is PCM, 2 is ADX */
         bytestream_put_be16(&ptr, audio->codecpar->sample_rate); /* Audio sampling rate */