mirror of https://git.ffmpeg.org/ffmpeg.git
vmdaudio: fix decoding of 16-bit audio format.
The initial sample of each block is raw 16-bit PCM, not DPCM. Fixes decoding of all samples in: http://streams.videolan.org/samples/game-formats/sierra-vmd/Lighthouse/
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@ -417,9 +417,8 @@ static av_cold int vmdvideo_decode_end(AVCodecContext *avctx)
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#define BLOCK_TYPE_SILENCE 3
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typedef struct VmdAudioContext {
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AVCodecContext *avctx;
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int out_bps;
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int predictors[2];
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int chunk_size;
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} VmdAudioContext;
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static const uint16_t vmdaudio_table[128] = {
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@ -442,13 +441,23 @@ 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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s->avctx = avctx;
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if (avctx->channels < 1 || avctx->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) {
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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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if (avctx->bits_per_coded_sample == 16)
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avctx->sample_fmt = AV_SAMPLE_FMT_S16;
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else
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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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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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@ -457,52 +466,47 @@ static av_cold int vmdaudio_decode_init(AVCodecContext *avctx)
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return 0;
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}
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static void vmdaudio_decode_audio(VmdAudioContext *s, unsigned char *data,
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const uint8_t *buf, int buf_size, int stereo)
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static void decode_audio_s16(int16_t *out, const uint8_t *buf, int buf_size,
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int channels)
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{
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int i;
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int chan = 0;
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int16_t *out = (int16_t*)data;
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int ch;
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const uint8_t *buf_end = buf + buf_size;
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int predictor[2];
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int st = channels - 1;
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for(i = 0; i < buf_size; i++) {
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if(buf[i] & 0x80)
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s->predictors[chan] -= vmdaudio_table[buf[i] & 0x7F];
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/* decode initial raw sample */
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for (ch = 0; ch < channels; ch++) {
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predictor[ch] = (int16_t)AV_RL16(buf);
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buf += 2;
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*out++ = predictor[ch];
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}
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/* decode DPCM samples */
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ch = 0;
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while (buf < buf_end) {
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uint8_t b = *buf++;
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if (b & 0x80)
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predictor[ch] -= vmdaudio_table[b & 0x7F];
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else
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s->predictors[chan] += vmdaudio_table[buf[i]];
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s->predictors[chan] = av_clip_int16(s->predictors[chan]);
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out[i] = s->predictors[chan];
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chan ^= stereo;
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predictor[ch] += vmdaudio_table[b];
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predictor[ch] = av_clip_int16(predictor[ch]);
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*out++ = predictor[ch];
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ch ^= st;
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}
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}
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static int vmdaudio_loadsound(VmdAudioContext *s, unsigned char *data,
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const uint8_t *buf, int silent_chunks, int data_size)
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{
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int silent_size = s->avctx->block_align * silent_chunks * s->out_bps;
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if (silent_chunks) {
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memset(data, s->out_bps == 2 ? 0x00 : 0x80, silent_size);
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data += silent_size;
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}
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if (s->avctx->bits_per_coded_sample == 16)
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vmdaudio_decode_audio(s, data, buf, data_size, s->avctx->channels == 2);
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else {
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/* just copy the data */
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memcpy(data, buf, data_size);
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}
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return silent_size + data_size * s->out_bps;
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}
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static int vmdaudio_decode_frame(AVCodecContext *avctx,
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void *data, int *data_size,
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AVPacket *avpkt)
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{
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const uint8_t *buf = avpkt->data;
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const uint8_t *buf_end;
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int buf_size = avpkt->size;
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VmdAudioContext *s = avctx->priv_data;
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int block_type, silent_chunks;
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unsigned char *output_samples = (unsigned char *)data;
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int block_type, silent_chunks, audio_chunks;
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int nb_samples, out_size;
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uint8_t *output_samples_u8 = data;
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int16_t *output_samples_s16 = data;
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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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@ -518,9 +522,15 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
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buf += 16;
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buf_size -= 16;
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/* get number of silent chunks */
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silent_chunks = 0;
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if (block_type == BLOCK_TYPE_INITIAL) {
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uint32_t flags = AV_RB32(buf);
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uint32_t flags;
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if (buf_size < 4) {
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av_log(avctx, AV_LOG_ERROR, "packet is too small\n");
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return AVERROR(EINVAL);
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}
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flags = AV_RB32(buf);
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silent_chunks = av_popcount(flags);
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buf += 4;
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buf_size -= 4;
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@ -530,11 +540,41 @@ static int vmdaudio_decode_frame(AVCodecContext *avctx,
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}
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/* ensure output buffer is large enough */
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if (*data_size < (avctx->block_align*silent_chunks + buf_size) * s->out_bps)
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audio_chunks = buf_size / s->chunk_size;
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nb_samples = ((silent_chunks + audio_chunks) * avctx->block_align) / avctx->channels;
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out_size = nb_samples * avctx->channels * s->out_bps;
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if (*data_size < out_size)
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return -1;
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*data_size = vmdaudio_loadsound(s, output_samples, buf, silent_chunks, buf_size);
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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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if (s->out_bps == 2) {
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memset(output_samples_s16, 0x00, silent_size * 2);
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output_samples_s16 += silent_size;
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} else {
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memset(output_samples_u8, 0x80, silent_size);
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output_samples_u8 += silent_size;
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}
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}
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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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while (buf < buf_end) {
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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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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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output_samples_u8 += avctx->block_align;
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}
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buf += s->chunk_size;
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}
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}
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*data_size = out_size;
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return avpkt->size;
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}
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