mirror of
https://git.ffmpeg.org/ffmpeg.git
synced 2024-12-25 08:42:39 +00:00
adpcmdec: calculate actual number of output samples for each decoder.
This also allows for removing some of the buf_size checks and using the sample count for some of the decoding loops.
This commit is contained in:
parent
439998e18b
commit
a62c0f94ee
@ -315,6 +315,173 @@ static void xa_decode(short *out, const unsigned char *in,
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}
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}
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/**
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* Get the number of samples that will be decoded from the packet.
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* In one case, this is actually the maximum number of samples possible to
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* decode with the given buf_size.
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*
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* @param[out] coded_samples set to the number of samples as coded in the
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* packet, or 0 if the codec does not encode the
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* number of samples in each frame.
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*/
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static int get_nb_samples(AVCodecContext *avctx, const uint8_t *buf,
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int buf_size, int *coded_samples)
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{
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ADPCMDecodeContext *s = avctx->priv_data;
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int nb_samples = 0;
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int ch = avctx->channels;
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int has_coded_samples = 0;
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int header_size;
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*coded_samples = 0;
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switch (avctx->codec->id) {
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/* constant, only check buf_size */
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case CODEC_ID_ADPCM_EA_XAS:
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if (buf_size < 76 * ch)
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return 0;
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nb_samples = 128;
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break;
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case CODEC_ID_ADPCM_IMA_QT:
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if (buf_size < 34 * ch)
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return 0;
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nb_samples = 64;
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break;
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/* simple 4-bit adpcm */
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case CODEC_ID_ADPCM_CT:
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case CODEC_ID_ADPCM_IMA_EA_SEAD:
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case CODEC_ID_ADPCM_IMA_WS:
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case CODEC_ID_ADPCM_YAMAHA:
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nb_samples = buf_size * 2 / ch;
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break;
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}
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if (nb_samples)
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return nb_samples;
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/* simple 4-bit adpcm, with header */
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header_size = 0;
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switch (avctx->codec->id) {
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case CODEC_ID_ADPCM_4XM:
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case CODEC_ID_ADPCM_IMA_ISS: header_size = 4 * ch; break;
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case CODEC_ID_ADPCM_IMA_AMV: header_size = 8; break;
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case CODEC_ID_ADPCM_IMA_SMJPEG: header_size = 4; break;
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}
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if (header_size > 0)
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return (buf_size - header_size) * 2 / ch;
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/* more complex formats */
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switch (avctx->codec->id) {
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case CODEC_ID_ADPCM_EA:
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has_coded_samples = 1;
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if (buf_size < 4)
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return 0;
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*coded_samples = AV_RL32(buf);
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*coded_samples -= *coded_samples % 28;
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nb_samples = (buf_size - 12) / 30 * 28;
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break;
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case CODEC_ID_ADPCM_IMA_EA_EACS:
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has_coded_samples = 1;
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if (buf_size < 4)
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return 0;
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*coded_samples = AV_RL32(buf);
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nb_samples = (buf_size - (4 + 8 * ch)) * 2 / ch;
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break;
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case CODEC_ID_ADPCM_EA_MAXIS_XA:
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nb_samples = ((buf_size - ch) / (2 * ch)) * 2 * ch;
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break;
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case CODEC_ID_ADPCM_EA_R1:
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case CODEC_ID_ADPCM_EA_R2:
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case CODEC_ID_ADPCM_EA_R3:
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/* maximum number of samples */
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/* has internal offsets and a per-frame switch to signal raw 16-bit */
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has_coded_samples = 1;
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if (buf_size < 4)
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return 0;
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switch (avctx->codec->id) {
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case CODEC_ID_ADPCM_EA_R1:
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header_size = 4 + 9 * ch;
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*coded_samples = AV_RL32(buf);
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break;
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case CODEC_ID_ADPCM_EA_R2:
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header_size = 4 + 5 * ch;
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*coded_samples = AV_RL32(buf);
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break;
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case CODEC_ID_ADPCM_EA_R3:
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header_size = 4 + 5 * ch;
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*coded_samples = AV_RB32(buf);
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break;
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}
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*coded_samples -= *coded_samples % 28;
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nb_samples = (buf_size - header_size) * 2 / ch;
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nb_samples -= nb_samples % 28;
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break;
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case CODEC_ID_ADPCM_IMA_DK3:
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if (avctx->block_align > 0)
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buf_size = FFMIN(buf_size, avctx->block_align);
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nb_samples = ((buf_size - 16) * 8 / 3) / ch;
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break;
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case CODEC_ID_ADPCM_IMA_DK4:
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nb_samples = 1 + (buf_size - 4 * ch) * 2 / ch;
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break;
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case CODEC_ID_ADPCM_IMA_WAV:
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if (avctx->block_align > 0)
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buf_size = FFMIN(buf_size, avctx->block_align);
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nb_samples = 1 + (buf_size - 4 * ch) / (4 * ch) * 8;
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break;
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case CODEC_ID_ADPCM_MS:
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if (avctx->block_align > 0)
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buf_size = FFMIN(buf_size, avctx->block_align);
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nb_samples = 2 + (buf_size - 7 * ch) * 2 / ch;
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break;
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case CODEC_ID_ADPCM_SBPRO_2:
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case CODEC_ID_ADPCM_SBPRO_3:
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case CODEC_ID_ADPCM_SBPRO_4:
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{
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int samples_per_byte;
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switch (avctx->codec->id) {
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case CODEC_ID_ADPCM_SBPRO_2: samples_per_byte = 4; break;
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case CODEC_ID_ADPCM_SBPRO_3: samples_per_byte = 3; break;
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case CODEC_ID_ADPCM_SBPRO_4: samples_per_byte = 2; break;
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}
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if (!s->status[0].step_index) {
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nb_samples++;
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buf_size -= ch;
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}
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nb_samples += buf_size * samples_per_byte / ch;
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break;
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}
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case CODEC_ID_ADPCM_SWF:
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{
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int buf_bits = buf_size * 8 - 2;
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int nbits = (buf[0] >> 6) + 2;
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int block_hdr_size = 22 * ch;
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int block_size = block_hdr_size + nbits * ch * 4095;
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int nblocks = buf_bits / block_size;
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int bits_left = buf_bits - nblocks * block_size;
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nb_samples = nblocks * 4096;
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if (bits_left >= block_hdr_size)
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nb_samples += 1 + (bits_left - block_hdr_size) / (nbits * ch);
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break;
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}
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case CODEC_ID_ADPCM_THP:
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has_coded_samples = 1;
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if (buf_size < 8)
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return 0;
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*coded_samples = AV_RB32(&buf[4]);
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*coded_samples -= *coded_samples % 14;
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nb_samples = (buf_size - 80) / (8 * ch) * 14;
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break;
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case CODEC_ID_ADPCM_XA:
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nb_samples = (buf_size / 128) * 224 / ch;
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break;
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}
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/* validate coded sample count */
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if (has_coded_samples && (*coded_samples <= 0 || *coded_samples > nb_samples))
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return AVERROR_INVALIDDATA;
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return nb_samples;
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}
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/* DK3 ADPCM support macro */
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#define DK3_GET_NEXT_NIBBLE() \
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@ -344,20 +511,33 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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ADPCMChannelStatus *cs;
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int n, m, channel, i;
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short *samples;
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short *samples_end;
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const uint8_t *src;
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int st; /* stereo */
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uint32_t samples_in_chunk;
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int count1, count2;
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int nb_samples, coded_samples, out_bps, out_size;
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//should protect all 4bit ADPCM variants
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//8 is needed for CODEC_ID_ADPCM_IMA_WAV with 2 channels
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//
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if(*data_size/4 < buf_size + 8)
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return -1;
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nb_samples = get_nb_samples(avctx, buf, buf_size, &coded_samples);
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if (nb_samples <= 0) {
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av_log(avctx, AV_LOG_ERROR, "invalid number of samples in packet\n");
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return AVERROR_INVALIDDATA;
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}
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out_bps = av_get_bytes_per_sample(avctx->sample_fmt);
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out_size = nb_samples * avctx->channels * out_bps;
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if (*data_size < out_size) {
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av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n");
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return AVERROR(EINVAL);
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}
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/* use coded_samples when applicable */
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/* it is always <= nb_samples, so the output buffer will be large enough */
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if (coded_samples) {
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if (coded_samples != nb_samples)
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av_log(avctx, AV_LOG_WARNING, "mismatch in coded sample count\n");
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nb_samples = coded_samples;
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out_size = nb_samples * avctx->channels * out_bps;
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}
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samples = data;
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samples_end= samples + *data_size/2;
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src = buf;
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st = avctx->channels == 2 ? 1 : 0;
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@ -366,10 +546,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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case CODEC_ID_ADPCM_IMA_QT:
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/* In QuickTime, IMA is encoded by chunks of 34 bytes (=64 samples).
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Channel data is interleaved per-chunk. */
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if (buf_size / 34 < avctx->channels) {
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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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for (channel = 0; channel < avctx->channels; channel++) {
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int16_t predictor;
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int step_index;
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@ -410,15 +586,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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src ++;
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}
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}
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if (st)
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samples--;
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break;
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case CODEC_ID_ADPCM_IMA_WAV:
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if (avctx->block_align != 0 && buf_size > avctx->block_align)
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buf_size = avctx->block_align;
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// samples_per_block= (block_align-4*chanels)*8 / (bits_per_sample * chanels) + 1;
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for(i=0; i<avctx->channels; i++){
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cs = &(c->status[i]);
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cs->predictor = *samples++ = (int16_t)bytestream_get_le16(&src);
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@ -431,7 +603,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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if (*src++) av_log(avctx, AV_LOG_ERROR, "unused byte should be null but is %d!!\n", src[-1]); /* unused */
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}
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while (src <= buf + buf_size - (avctx->channels * 4)) {
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for (n = (nb_samples - 1) / 8; n > 0; n--) {
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for (i = 0; i < avctx->channels; i++) {
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cs = &c->status[i];
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for (m = 0; m < 4; m++) {
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@ -455,20 +627,17 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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c->status[i].step_index = av_clip(c->status[i].step_index, 0, 88);
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}
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m= (buf_size - (src - buf))>>st;
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for (i = 0; i < avctx->channels; i++) {
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samples = (short*)data + i;
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cs = &c->status[i];
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for (n = 0; n < m; n++) {
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uint8_t v = *src++;
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for (n = nb_samples >> 1; n > 0; n--, src++) {
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uint8_t v = *src;
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*samples = adpcm_ima_expand_nibble(cs, v & 0x0F, 4);
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samples += avctx->channels;
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*samples = adpcm_ima_expand_nibble(cs, v >> 4 , 4);
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samples += avctx->channels;
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}
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}
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samples -= (avctx->channels - 1);
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break;
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case CODEC_ID_ADPCM_MS:
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{
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@ -476,9 +645,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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if (avctx->block_align != 0 && buf_size > avctx->block_align)
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buf_size = avctx->block_align;
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n = buf_size - 7 * avctx->channels;
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if (n < 0)
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return -1;
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block_predictor = av_clip(*src++, 0, 6);
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c->status[0].coeff1 = ff_adpcm_AdaptCoeff1[block_predictor];
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@ -502,10 +668,9 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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if (st) *samples++ = c->status[1].sample2;
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*samples++ = c->status[0].sample1;
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if (st) *samples++ = c->status[1].sample1;
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for(;n>0;n--) {
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for(n = (nb_samples - 2) >> (1 - st); n > 0; n--, src++) {
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*samples++ = adpcm_ms_expand_nibble(&c->status[0 ], src[0] >> 4 );
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*samples++ = adpcm_ms_expand_nibble(&c->status[st], src[0] & 0x0F);
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src ++;
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}
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break;
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}
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@ -513,12 +678,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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if (avctx->block_align != 0 && buf_size > avctx->block_align)
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buf_size = avctx->block_align;
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n = buf_size - 4 * avctx->channels;
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if (n < 0) {
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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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for (channel = 0; channel < avctx->channels; channel++) {
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cs = &c->status[channel];
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cs->predictor = (int16_t)bytestream_get_le16(&src);
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@ -526,8 +685,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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src++;
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*samples++ = cs->predictor;
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}
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while (n-- > 0) {
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uint8_t v = *src++;
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for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
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uint8_t v = *src;
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*samples++ = adpcm_ima_expand_nibble(&c->status[0 ], v >> 4 , 3);
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*samples++ = adpcm_ima_expand_nibble(&c->status[st], v & 0x0F, 3);
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}
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@ -543,9 +702,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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if (avctx->block_align != 0 && buf_size > avctx->block_align)
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buf_size = avctx->block_align;
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if(buf_size + 16 > (samples_end - samples)*3/8)
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return -1;
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c->status[0].predictor = (int16_t)AV_RL16(src + 10);
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c->status[1].predictor = (int16_t)AV_RL16(src + 12);
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c->status[0].step_index = src[14];
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@ -586,12 +742,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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break;
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}
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case CODEC_ID_ADPCM_IMA_ISS:
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n = buf_size - 4 * avctx->channels;
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if (n < 0) {
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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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for (channel = 0; channel < avctx->channels; channel++) {
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cs = &c->status[channel];
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cs->predictor = (int16_t)bytestream_get_le16(&src);
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@ -599,9 +749,9 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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src++;
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}
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while (n-- > 0) {
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for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
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uint8_t v1, v2;
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uint8_t v = *src++;
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uint8_t v = *src;
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/* nibbles are swapped for mono */
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if (st) {
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v1 = v >> 4;
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@ -630,28 +780,21 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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buf_size -= 128;
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}
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break;
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case CODEC_ID_ADPCM_IMA_EA_EACS: {
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unsigned header_size = 4 + (8<<st);
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samples_in_chunk = bytestream_get_le32(&src) >> (1-st);
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if (buf_size < header_size || samples_in_chunk > buf_size - header_size) {
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src += buf_size - 4;
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break;
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}
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case CODEC_ID_ADPCM_IMA_EA_EACS:
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src += 4; // skip sample count (already read)
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for (i=0; i<=st; i++)
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c->status[i].step_index = bytestream_get_le32(&src);
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for (i=0; i<=st; i++)
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c->status[i].predictor = bytestream_get_le32(&src);
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for (; samples_in_chunk; samples_in_chunk--, src++) {
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for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
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*samples++ = adpcm_ima_expand_nibble(&c->status[0], *src>>4, 3);
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*samples++ = adpcm_ima_expand_nibble(&c->status[st], *src&0x0F, 3);
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}
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break;
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}
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case CODEC_ID_ADPCM_IMA_EA_SEAD:
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for (; src < buf+buf_size; src++) {
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for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
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*samples++ = adpcm_ima_expand_nibble(&c->status[0], src[0] >> 4, 6);
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*samples++ = adpcm_ima_expand_nibble(&c->status[st],src[0]&0x0F, 6);
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}
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@ -666,22 +809,15 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
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/* Each EA ADPCM frame has a 12-byte header followed by 30-byte pieces,
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each coding 28 stereo samples. */
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if (buf_size < 12) {
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av_log(avctx, AV_LOG_ERROR, "frame too small\n");
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return AVERROR(EINVAL);
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}
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samples_in_chunk = AV_RL32(src);
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if (samples_in_chunk / 28 > (buf_size - 12) / 30) {
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av_log(avctx, AV_LOG_ERROR, "invalid frame\n");
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return AVERROR(EINVAL);
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}
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src += 4;
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src += 4; // skip sample count (already read)
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current_left_sample = (int16_t)bytestream_get_le16(&src);
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previous_left_sample = (int16_t)bytestream_get_le16(&src);
|
||||
current_right_sample = (int16_t)bytestream_get_le16(&src);
|
||||
previous_right_sample = (int16_t)bytestream_get_le16(&src);
|
||||
|
||||
for (count1 = 0; count1 < samples_in_chunk/28;count1++) {
|
||||
for (count1 = 0; count1 < nb_samples / 28; count1++) {
|
||||
coeff1l = ea_adpcm_table[ *src >> 4 ];
|
||||
coeff2l = ea_adpcm_table[(*src >> 4 ) + 4];
|
||||
coeff1r = ea_adpcm_table[*src & 0x0F];
|
||||
@ -728,7 +864,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
shift[channel] = (*src & 0x0F) + 8;
|
||||
src++;
|
||||
}
|
||||
for (count1 = 0; count1 < (buf_size - avctx->channels) / avctx->channels; count1++) {
|
||||
for (count1 = 0; count1 < nb_samples / 2; count1++) {
|
||||
for(i = 4; i >= 0; i-=4) { /* Pairwise samples LL RR (st) or LL LL (mono) */
|
||||
for(channel = 0; channel < avctx->channels; channel++) {
|
||||
int32_t sample = (int32_t)(((*(src+channel) >> i) & 0x0F) << 0x1C) >> shift[channel];
|
||||
@ -742,6 +878,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
}
|
||||
src+=avctx->channels;
|
||||
}
|
||||
/* consume whole packet */
|
||||
src = buf + buf_size;
|
||||
break;
|
||||
}
|
||||
case CODEC_ID_ADPCM_EA_R1:
|
||||
@ -759,14 +897,9 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
uint16_t *samplesC;
|
||||
const uint8_t *srcC;
|
||||
const uint8_t *src_end = buf + buf_size;
|
||||
int count = 0;
|
||||
|
||||
samples_in_chunk = (big_endian ? bytestream_get_be32(&src)
|
||||
: bytestream_get_le32(&src)) / 28;
|
||||
if (samples_in_chunk > UINT32_MAX/(28*avctx->channels) ||
|
||||
28*samples_in_chunk*avctx->channels > samples_end-samples) {
|
||||
src += buf_size - 4;
|
||||
break;
|
||||
}
|
||||
src += 4; // skip sample count (already read)
|
||||
|
||||
for (channel=0; channel<avctx->channels; channel++) {
|
||||
int32_t offset = (big_endian ? bytestream_get_be32(&src)
|
||||
@ -785,7 +918,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
previous_sample = c->status[channel].prev_sample;
|
||||
}
|
||||
|
||||
for (count1=0; count1<samples_in_chunk; count1++) {
|
||||
for (count1 = 0; count1 < nb_samples / 28; count1++) {
|
||||
if (*srcC == 0xEE) { /* only seen in R2 and R3 */
|
||||
srcC++;
|
||||
if (srcC > src_end - 30*2) break;
|
||||
@ -819,6 +952,12 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
}
|
||||
}
|
||||
}
|
||||
if (!count) {
|
||||
count = count1;
|
||||
} else if (count != count1) {
|
||||
av_log(avctx, AV_LOG_WARNING, "per-channel sample count mismatch\n");
|
||||
count = FFMAX(count, count1);
|
||||
}
|
||||
|
||||
if (avctx->codec->id != CODEC_ID_ADPCM_EA_R1) {
|
||||
c->status[channel].predictor = current_sample;
|
||||
@ -826,16 +965,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
}
|
||||
}
|
||||
|
||||
src = src + buf_size - (4 + 4*avctx->channels);
|
||||
samples += 28 * samples_in_chunk * avctx->channels;
|
||||
out_size = count * 28 * avctx->channels * out_bps;
|
||||
src = src_end;
|
||||
break;
|
||||
}
|
||||
case CODEC_ID_ADPCM_EA_XAS:
|
||||
if (samples_end-samples < 32*4*avctx->channels
|
||||
|| buf_size < (4+15)*4*avctx->channels) {
|
||||
src += buf_size;
|
||||
break;
|
||||
}
|
||||
for (channel=0; channel<avctx->channels; channel++) {
|
||||
int coeff[2][4], shift[4];
|
||||
short *s2, *s = &samples[channel];
|
||||
@ -859,7 +993,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
}
|
||||
}
|
||||
}
|
||||
samples += 32*4*avctx->channels;
|
||||
break;
|
||||
case CODEC_ID_ADPCM_IMA_AMV:
|
||||
case CODEC_ID_ADPCM_IMA_SMJPEG:
|
||||
@ -869,7 +1002,7 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
if (avctx->codec->id == CODEC_ID_ADPCM_IMA_AMV)
|
||||
src+=4;
|
||||
|
||||
while (src < buf + buf_size) {
|
||||
for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
|
||||
char hi, lo;
|
||||
lo = *src & 0x0F;
|
||||
hi = *src >> 4;
|
||||
@ -881,12 +1014,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
lo, 3);
|
||||
*samples++ = adpcm_ima_expand_nibble(&c->status[0],
|
||||
hi, 3);
|
||||
src++;
|
||||
}
|
||||
break;
|
||||
case CODEC_ID_ADPCM_CT:
|
||||
while (src < buf + buf_size) {
|
||||
uint8_t v = *src++;
|
||||
for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
|
||||
uint8_t v = *src;
|
||||
*samples++ = adpcm_ct_expand_nibble(&c->status[0 ], v >> 4 );
|
||||
*samples++ = adpcm_ct_expand_nibble(&c->status[st], v & 0x0F);
|
||||
}
|
||||
@ -900,27 +1032,26 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
if (st)
|
||||
*samples++ = 128 * (*src++ - 0x80);
|
||||
c->status[0].step_index = 1;
|
||||
nb_samples--;
|
||||
}
|
||||
if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_4) {
|
||||
while (src < buf + buf_size) {
|
||||
for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
|
||||
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
|
||||
src[0] >> 4, 4, 0);
|
||||
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
|
||||
src[0] & 0x0F, 4, 0);
|
||||
src++;
|
||||
}
|
||||
} else if (avctx->codec->id == CODEC_ID_ADPCM_SBPRO_3) {
|
||||
while (src < buf + buf_size && samples + 2 < samples_end) {
|
||||
for (n = nb_samples / 3; n > 0; n--, src++) {
|
||||
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
|
||||
src[0] >> 5 , 3, 0);
|
||||
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
|
||||
(src[0] >> 2) & 0x07, 3, 0);
|
||||
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
|
||||
src[0] & 0x03, 2, 0);
|
||||
src++;
|
||||
}
|
||||
} else {
|
||||
while (src < buf + buf_size && samples + 3 < samples_end) {
|
||||
for (n = nb_samples >> (2 - st); n > 0; n--, src++) {
|
||||
*samples++ = adpcm_sbpro_expand_nibble(&c->status[0],
|
||||
src[0] >> 6 , 2, 2);
|
||||
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
|
||||
@ -929,7 +1060,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
(src[0] >> 2) & 0x03, 2, 2);
|
||||
*samples++ = adpcm_sbpro_expand_nibble(&c->status[st],
|
||||
src[0] & 0x03, 2, 2);
|
||||
src++;
|
||||
}
|
||||
}
|
||||
break;
|
||||
@ -984,10 +1114,6 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
c->status[i].predictor = av_clip_int16(c->status[i].predictor);
|
||||
|
||||
*samples++ = c->status[i].predictor;
|
||||
if (samples >= samples_end) {
|
||||
av_log(avctx, AV_LOG_ERROR, "allocated output buffer is too small\n");
|
||||
return -1;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
@ -995,8 +1121,8 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
break;
|
||||
}
|
||||
case CODEC_ID_ADPCM_YAMAHA:
|
||||
while (src < buf + buf_size) {
|
||||
uint8_t v = *src++;
|
||||
for (n = nb_samples >> (1 - st); n > 0; n--, src++) {
|
||||
uint8_t v = *src;
|
||||
*samples++ = adpcm_yamaha_expand_nibble(&c->status[0 ], v & 0x0F);
|
||||
*samples++ = adpcm_yamaha_expand_nibble(&c->status[st], v >> 4 );
|
||||
}
|
||||
@ -1004,17 +1130,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
case CODEC_ID_ADPCM_THP:
|
||||
{
|
||||
int table[2][16];
|
||||
unsigned int samplecnt;
|
||||
int prev[2][2];
|
||||
int ch;
|
||||
|
||||
if (buf_size < 80) {
|
||||
av_log(avctx, AV_LOG_ERROR, "frame too small\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
src+=4;
|
||||
samplecnt = bytestream_get_be32(&src);
|
||||
src += 4; // skip channel size
|
||||
src += 4; // skip number of samples (already read)
|
||||
|
||||
for (i = 0; i < 32; i++)
|
||||
table[0][i] = (int16_t)bytestream_get_be16(&src);
|
||||
@ -1023,16 +1143,11 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
for (i = 0; i < 4; i++)
|
||||
prev[0][i] = (int16_t)bytestream_get_be16(&src);
|
||||
|
||||
if (samplecnt >= (samples_end - samples) / (st + 1)) {
|
||||
av_log(avctx, AV_LOG_ERROR, "allocated output buffer is too small\n");
|
||||
return -1;
|
||||
}
|
||||
|
||||
for (ch = 0; ch <= st; ch++) {
|
||||
samples = (unsigned short *) data + ch;
|
||||
|
||||
/* Read in every sample for this channel. */
|
||||
for (i = 0; i < samplecnt / 14; i++) {
|
||||
for (i = 0; i < nb_samples / 14; i++) {
|
||||
int index = (*src >> 4) & 7;
|
||||
unsigned int exp = 28 - (*src++ & 15);
|
||||
int factor1 = table[ch][index * 2];
|
||||
@ -1056,17 +1171,13 @@ static int adpcm_decode_frame(AVCodecContext *avctx,
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
/* In the previous loop, in case stereo is used, samples is
|
||||
increased exactly one time too often. */
|
||||
samples -= st;
|
||||
break;
|
||||
}
|
||||
|
||||
default:
|
||||
return -1;
|
||||
}
|
||||
*data_size = (uint8_t *)samples - (uint8_t *)data;
|
||||
*data_size = out_size;
|
||||
return src - buf;
|
||||
}
|
||||
|
||||
|
Loading…
Reference in New Issue
Block a user