virtualenv/FFmpeg
1
0
Fork 0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2025-10-30 23:18:11 +02:00
Code Issues Releases Activity

segafilm: convert to new channel layout API

Browse Source 
Signed-off-by: James Almer <jamrial@gmail.com>
This commit is contained in:
Vittorio Giovara
2017-04-04 15:41:55 +02:00
committed by James Almer
parent 81b3ebbae9
commit 525ccbe8f1
2 changed files with 5 additions and 5 deletions
Download Patch File Download Diff File Expand all files Collapse all files

8
libavformat/segafilm.c
View File

@@ -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;
}

2
libavformat/segafilmenc.c
View File

@@ -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 */