diff --git a/libavcodec/vmdaudio.c b/libavcodec/vmdaudio.c index 53aef660ef..5e04686cb1 100644 --- a/libavcodec/vmdaudio.c +++ b/libavcodec/vmdaudio.c @@ -71,20 +71,20 @@ static const uint16_t vmdaudio_table[128] = { static av_cold int vmdaudio_decode_init(AVCodecContext *avctx) { VmdAudioContext *s = avctx->priv_data; + int channels = avctx->ch_layout.nb_channels; - if (avctx->channels < 1 || avctx->channels > 2) { + if (channels < 1 || channels > 2) { av_log(avctx, AV_LOG_ERROR, "invalid number of channels\n"); return AVERROR(EINVAL); } - if (avctx->block_align < 1 || avctx->block_align % avctx->channels || - avctx->block_align > INT_MAX - avctx->channels - ) { + if (avctx->block_align < 1 || avctx->block_align % channels || + avctx->block_align > INT_MAX - channels) { av_log(avctx, AV_LOG_ERROR, "invalid block align\n"); return AVERROR(EINVAL); } - avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO : - AV_CH_LAYOUT_STEREO; + av_channel_layout_uninit(&avctx->ch_layout); + av_channel_layout_default(&avctx->ch_layout, channels == 1); if (avctx->bits_per_coded_sample == 16) avctx->sample_fmt = AV_SAMPLE_FMT_S16; @@ -92,11 +92,11 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx) avctx->sample_fmt = AV_SAMPLE_FMT_U8; s->out_bps = av_get_bytes_per_sample(avctx->sample_fmt); - s->chunk_size = avctx->block_align + avctx->channels * (s->out_bps == 2); + s->chunk_size = avctx->block_align + channels * (s->out_bps == 2); av_log(avctx, AV_LOG_DEBUG, "%d channels, %d bits/sample, " "block align = %d, sample rate = %d\n", - avctx->channels, avctx->bits_per_coded_sample, avctx->block_align, + channels, avctx->bits_per_coded_sample, avctx->block_align, avctx->sample_rate); return 0; @@ -143,6 +143,7 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data, int ret; uint8_t *output_samples_u8; int16_t *output_samples_s16; + int channels = avctx->ch_layout.nb_channels; if (buf_size < 16) { av_log(avctx, AV_LOG_WARNING, "skipping small junk packet\n"); @@ -186,7 +187,7 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data, /* get output buffer */ frame->nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) / - avctx->channels; + avctx->ch_layout.nb_channels; if ((ret = ff_get_buffer(avctx, frame, 0)) < 0) return ret; output_samples_u8 = frame->data[0]; @@ -195,7 +196,7 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data, /* decode silent chunks */ if (silent_chunks > 0) { int silent_size = avctx->block_align * silent_chunks; - av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->channels); + av_assert0(avctx->block_align * silent_chunks <= frame->nb_samples * avctx->ch_layout.nb_channels); if (s->out_bps == 2) { memset(output_samples_s16, 0x00, silent_size * 2); @@ -209,11 +210,10 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx, void *data, /* decode audio chunks */ if (audio_chunks > 0) { buf_end = buf + buf_size; - av_assert0((buf_size & (avctx->channels > 1)) == 0); + av_assert0((buf_size & (avctx->ch_layout.nb_channels > 1)) == 0); while (buf_end - buf >= s->chunk_size) { if (s->out_bps == 2) { - decode_audio_s16(output_samples_s16, buf, s->chunk_size, - avctx->channels); + decode_audio_s16(output_samples_s16, buf, s->chunk_size, channels); output_samples_s16 += avctx->block_align; } else { memcpy(output_samples_u8, buf, s->chunk_size);