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https://github.com/FFmpeg/FFmpeg.git
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vmdaudio: convert to new channel layout API
Signed-off-by: Vittorio Giovara <vittorio.giovara@gmail.com> Signed-off-by: Anton Khirnov <anton@khirnov.net> Signed-off-by: James Almer <jamrial@gmail.com>
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@ -71,20 +71,20 @@ static const uint16_t vmdaudio_table[128] = {
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static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
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{
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VmdAudioContext *s = avctx->priv_data;
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int channels = avctx->ch_layout.nb_channels;
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if (avctx->channels < 1 || avctx->channels > 2) {
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if (channels < 1 || channels > 2) {
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av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n");
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return AVERROR(EINVAL);
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}
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if (avctx->block_align < 1 || avctx->block_align % avctx->channels ||
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avctx->block_align > INT_MAX - avctx->channels
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) {
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if (avctx->block_align < 1 || avctx->block_align % channels ||
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avctx->block_align > INT_MAX - channels) {
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av_log(avctx, AV_LOG_ERROR, "invalid block align\n");
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return AVERROR(EINVAL);
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}
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avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
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AV_CH_LAYOUT_STEREO;
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av_channel_layout_uninit(&avctx->ch_layout);
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av_channel_layout_default(&avctx->ch_layout, channels == 1);
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if (avctx->bits_per_coded_sample == 16)
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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@ -92,11 +92,11 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
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avctx->sample_fmt = AV_SAMPLE_FMT_U8;
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s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
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s->chunk_size = avctx->block_align + avctx->channels * (s->out_bps == 2);
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s->chunk_size = avctx->block_align + channels * (s->out_bps == 2);
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av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, "
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"block align = %d, sample rate = %d\n",
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avctx->channels, avctx->bits_per_coded_sample, avctx->block_align,
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channels, avctx->bits_per_coded_sample, avctx->block_align,
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avctx->sample_rate);
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return 0;
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@ -143,6 +143,7 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data,
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int ret;
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uint8_t *output_samples_u8;
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int16_t *output_samples_s16;
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int channels = avctx->ch_layout.nb_channels;
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if (buf_size < 16) {
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av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n");
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@ -186,7 +187,7 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data,
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/* get output buffer */
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frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) /
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avctx->channels;
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avctx->ch_layout.nb_channels;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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output_samples_u8 = frame->data[0];
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@ -195,7 +196,7 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data,
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/* decode silent chunks */
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if (silent_chunks > 0) {
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int silent_size = avctx->block_align * silent_chunks;
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av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->channels);
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av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->ch_layout.nb_channels);
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if (s->out_bps == 2) {
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memset(output_samples_s16, 0x00, silent_size * 2);
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@ -209,11 +210,10 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data,
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/* decode audio chunks */
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if (audio_chunks > 0) {
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buf_end = buf + buf_size;
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av_assert0((buf_size & (avctx->channels > 1)) == 0);
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av_assert0((buf_size & (avctx->ch_layout.nb_channels > 1)) == 0);
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while (buf_end - buf >= s->chunk_size) {
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if (s->out_bps == 2) {
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decode_audio_s16(output_samples_s16, buf, s->chunk_size,
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avctx->channels);
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decode_audio_s16(output_samples_s16, buf, s->chunk_size, channels);
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output_samples_s16 += avctx->block_align;
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} else {
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memcpy(output_samples_u8, buf, s->chunk_size);
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