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https://github.com/xenia-project/FFmpeg.git
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riffenc: take an AVStream instead of an AVCodecContext
It will be useful in the following commits. Also, rename the AVCodecContext pointer name from 'stream' to 'codec'.
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commit
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@ -120,7 +120,7 @@ static int avi_write_counters(AVFormatContext *s, int riff_id)
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assert(avist->frames_hdr_strm);
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stream = s->streams[n]->codec;
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avio_seek(pb, avist->frames_hdr_strm, SEEK_SET);
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ff_parse_specific_params(stream, &au_byterate, &au_ssize, &au_scale);
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ff_parse_specific_params(s->streams[n], &au_byterate, &au_ssize, &au_scale);
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if (au_ssize == 0)
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avio_wl32(pb, avist->packet_count);
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else
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@ -248,7 +248,7 @@ static int avi_write_header(AVFormatContext *s)
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avio_wl16(pb, 0); /* language */
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avio_wl32(pb, 0); /* initial frame */
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ff_parse_specific_params(enc, &au_byterate, &au_ssize, &au_scale);
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ff_parse_specific_params(st, &au_byterate, &au_ssize, &au_scale);
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avio_wl32(pb, au_scale); /* scale */
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avio_wl32(pb, au_byterate); /* rate */
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@ -674,7 +674,7 @@ static int nut_write_header(AVFormatContext *s)
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AVStream *st = s->streams[i];
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int ssize;
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AVRational time_base;
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ff_parse_specific_params(st->codec, &time_base.den, &ssize,
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ff_parse_specific_params(st, &time_base.den, &ssize,
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&time_base.num);
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avpriv_set_pts_info(st, 64, time_base.num, time_base.den);
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@ -53,7 +53,7 @@ int ff_get_wav_header(AVIOContext *pb, AVCodecContext *codec, int size);
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extern const AVCodecTag ff_codec_bmp_tags[];
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extern const AVCodecTag ff_codec_wav_tags[];
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void ff_parse_specific_params(AVCodecContext *stream, int *au_rate, int *au_ssize, int *au_scale);
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void ff_parse_specific_params(AVStream *st, int *au_rate, int *au_ssize, int *au_scale);
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int ff_read_riff_info(AVFormatContext *s, int64_t size);
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@ -209,32 +209,33 @@ void ff_put_bmp_header(AVIOContext *pb, AVCodecContext *enc,
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avio_w8(pb, 0);
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}
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void ff_parse_specific_params(AVCodecContext *stream, int *au_rate,
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void ff_parse_specific_params(AVStream *st, int *au_rate,
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int *au_ssize, int *au_scale)
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{
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AVCodecContext *codec = st->codec;
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int gcd;
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int audio_frame_size;
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/* We use the known constant frame size for the codec if known, otherwise
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* fall back on using AVCodecContext.frame_size, which is not as reliable
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* for indicating packet duration. */
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audio_frame_size = av_get_audio_frame_duration(stream, 0);
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audio_frame_size = av_get_audio_frame_duration(codec, 0);
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if (!audio_frame_size)
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audio_frame_size = stream->frame_size;
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audio_frame_size = codec->frame_size;
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*au_ssize = stream->block_align;
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if (audio_frame_size && stream->sample_rate) {
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*au_ssize = codec->block_align;
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if (audio_frame_size && codec->sample_rate) {
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*au_scale = audio_frame_size;
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*au_rate = stream->sample_rate;
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} else if (stream->codec_type == AVMEDIA_TYPE_VIDEO ||
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stream->codec_type == AVMEDIA_TYPE_DATA ||
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stream->codec_type == AVMEDIA_TYPE_SUBTITLE) {
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*au_scale = stream->time_base.num;
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*au_rate = stream->time_base.den;
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*au_rate = codec->sample_rate;
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} else if (codec->codec_type == AVMEDIA_TYPE_VIDEO ||
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codec->codec_type == AVMEDIA_TYPE_DATA ||
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codec->codec_type == AVMEDIA_TYPE_SUBTITLE) {
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*au_scale = codec->time_base.num;
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*au_rate = codec->time_base.den;
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} else {
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*au_scale = stream->block_align ? stream->block_align * 8 : 8;
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*au_rate = stream->bit_rate ? stream->bit_rate :
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8 * stream->sample_rate;
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*au_scale = codec->block_align ? codec->block_align * 8 : 8;
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*au_rate = codec->bit_rate ? codec->bit_rate :
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8 * codec->sample_rate;
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}
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gcd = av_gcd(*au_scale, *au_rate);
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*au_scale /= gcd;
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