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>
release/5.1
Anton Khirnov 12 years ago committed by James Almer
parent 972586344a
commit eb81c946c1
  1. 26
      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);

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