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author | Michael Niedermayer <michaelni@gmx.at> | 2011-10-01 02:54:46 +0200 |
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committer | Michael Niedermayer <michaelni@gmx.at> | 2011-10-01 02:54:46 +0200 |
commit | ef74ab20c255abf49b856c15f812cc9ea3fec061 (patch) | |
tree | 8d80c8ff7272908dede2ef2d90b4bac460f3748d /libavcodec/dpcm.c | |
parent | 5ca5d432e028ffdd4067b87aed6702168c3207b6 (diff) | |
parent | 08bd22a61b820160bff5f98cd51d2e0135d02e00 (diff) | |
download | ffmpeg-ef74ab20c255abf49b856c15f812cc9ea3fec061.tar.gz |
Merge remote-tracking branch 'qatar/master'
* qatar/master: (34 commits)
dpcm: return error if packet is too small
dpcm: use smaller data types for static tables
dpcm: use sol_table_16 directly instead of through the DPCMContext.
dpcm: replace short with int16_t
dpcm: check to make sure channels is 1 or 2.
dpcm: misc pretty-printing
dpcm: remove unnecessary variable by using bytestream functions.
dpcm: move codec-specific variable declarations to their corresponding decoding blocks.
dpcm: consistently use the variable name 'n' for the next input byte.
dpcm: output AV_SAMPLE_FMT_U8 for Sol DPCM subcodecs 1 and 2.
dpcm: calculate and check actual output data size prior to decoding.
dpcm: factor out the stereo flag calculation
dpcm: cosmetics: rename channel_number to ch
avserver: Fix a bug where the socket is IPv4, but IPv6 is autoselected for the loopback address.
lavf: Avoid using av_malloc(0) in av_dump_format
dxva2_h264: pass the correct 8x8 scaling lists
dca: NEON optimised high freq VQ decoding
avcodec: reject audio packets with NULL data and non-zero size
dxva: Add ability to enable workaround for older ATI cards
latmenc: Set latmBufferFullness to largest value to indicate it is not used
...
Conflicts:
libavcodec/dxva2_h264.c
Merged-by: Michael Niedermayer <michaelni@gmx.at>
Diffstat (limited to 'libavcodec/dpcm.c')
-rw-r--r-- | libavcodec/dpcm.c | 240 |
1 files changed, 128 insertions, 112 deletions
diff --git a/libavcodec/dpcm.c b/libavcodec/dpcm.c index d9c15246e9..8f6cd8e115 100644 --- a/libavcodec/dpcm.c +++ b/libavcodec/dpcm.c @@ -39,17 +39,16 @@ #include "libavutil/intreadwrite.h" #include "avcodec.h" +#include "bytestream.h" typedef struct DPCMContext { int channels; - short roq_square_array[256]; - long sample[2];//for SOL_DPCM - const int *sol_table;//for SOL_DPCM + int16_t roq_square_array[256]; + int sample[2]; ///< previous sample (for SOL_DPCM) + const int8_t *sol_table; ///< delta table for SOL_DPCM } DPCMContext; -#define SE_16BIT(x) if (x & 0x8000) x -= 0x10000; - -static const int interplay_delta_table[] = { +static const int16_t interplay_delta_table[] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17, 18, 19, 20, 21, 22, 23, @@ -85,15 +84,17 @@ static const int interplay_delta_table[] = { }; -static const int sol_table_old[16] = - { 0x0, 0x1, 0x2 , 0x3, 0x6, 0xA, 0xF, 0x15, - -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0}; +static const int8_t sol_table_old[16] = { + 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15, + -0x15, -0xF, -0xA, -0x6, -0x3, -0x2, -0x1, 0x0 +}; -static const int sol_table_new[16] = - { 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15, - 0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15}; +static const int8_t sol_table_new[16] = { + 0x0, 0x1, 0x2, 0x3, 0x6, 0xA, 0xF, 0x15, + 0x0, -0x1, -0x2, -0x3, -0x6, -0xA, -0xF, -0x15 +}; -static const int sol_table_16[128] = { +static const int16_t sol_table_16[128] = { 0x000, 0x008, 0x010, 0x020, 0x030, 0x040, 0x050, 0x060, 0x070, 0x080, 0x090, 0x0A0, 0x0B0, 0x0C0, 0x0D0, 0x0E0, 0x0F0, 0x100, 0x110, 0x120, 0x130, 0x140, 0x150, 0x160, 0x170, 0x180, 0x190, 0x1A0, 0x1B0, 0x1C0, @@ -110,12 +111,15 @@ static const int sol_table_16[128] = { }; - static av_cold int dpcm_decode_init(AVCodecContext *avctx) { DPCMContext *s = avctx->priv_data; int i; - short square; + + if (avctx->channels < 1 || avctx->channels > 2) { + av_log(avctx, AV_LOG_INFO, "invalid number of channels\n"); + return AVERROR(EINVAL); + } s->channels = avctx->channels; s->sample[0] = s->sample[1] = 0; @@ -125,25 +129,23 @@ static av_cold int dpcm_decode_init(AVCodecContext *avctx) case CODEC_ID_ROQ_DPCM: /* initialize square table */ for (i = 0; i < 128; i++) { - square = i * i; - s->roq_square_array[i] = square; + int16_t square = i * i; + s->roq_square_array[i ] = square; s->roq_square_array[i + 128] = -square; } break; - case CODEC_ID_SOL_DPCM: switch(avctx->codec_tag){ case 1: - s->sol_table=sol_table_old; + s->sol_table = sol_table_old; s->sample[0] = s->sample[1] = 0x80; break; case 2: - s->sol_table=sol_table_new; + s->sol_table = sol_table_new; s->sample[0] = s->sample[1] = 0x80; break; case 3: - s->sol_table=sol_table_16; break; default: av_log(avctx, AV_LOG_ERROR, "Unknown SOL subcodec\n"); @@ -155,146 +157,160 @@ static av_cold int dpcm_decode_init(AVCodecContext *avctx) break; } - avctx->sample_fmt = AV_SAMPLE_FMT_S16; + if (avctx->codec->id == CODEC_ID_SOL_DPCM && avctx->codec_tag != 3) + avctx->sample_fmt = AV_SAMPLE_FMT_U8; + else + avctx->sample_fmt = AV_SAMPLE_FMT_S16; + return 0; } -static int dpcm_decode_frame(AVCodecContext *avctx, - void *data, int *data_size, + +static int dpcm_decode_frame(AVCodecContext *avctx, void *data, int *data_size, AVPacket *avpkt) { const uint8_t *buf = avpkt->data; int buf_size = avpkt->size; + const uint8_t *buf_end = buf + buf_size; DPCMContext *s = avctx->priv_data; - int in, out = 0; + int out = 0; int predictor[2]; - int channel_number = 0; - short *output_samples = data; - int shift[2]; - unsigned char byte; - short diff; + int ch = 0; + int stereo = s->channels - 1; + int16_t *output_samples = data; if (!buf_size) return 0; - // almost every DPCM variant expands one byte of data into two - if(*data_size/2 < buf_size) - return -1; + /* calculate output size */ + switch(avctx->codec->id) { + case CODEC_ID_ROQ_DPCM: + out = buf_size - 8; + break; + case CODEC_ID_INTERPLAY_DPCM: + out = buf_size - 6 - s->channels; + break; + case CODEC_ID_XAN_DPCM: + out = buf_size - 2 * s->channels; + break; + case CODEC_ID_SOL_DPCM: + if (avctx->codec_tag != 3) + out = buf_size * 2; + else + out = buf_size; + break; + } + out *= av_get_bytes_per_sample(avctx->sample_fmt); + if (out < 0) { + av_log(avctx, AV_LOG_ERROR, "packet is too small\n"); + return AVERROR(EINVAL); + } + if (*data_size < out) { + av_log(avctx, AV_LOG_ERROR, "output buffer is too small\n"); + return AVERROR(EINVAL); + } switch(avctx->codec->id) { case CODEC_ID_ROQ_DPCM: - if (s->channels == 1) - predictor[0] = AV_RL16(&buf[6]); - else { - predictor[0] = buf[7] << 8; - predictor[1] = buf[6] << 8; + buf += 6; + + if (stereo) { + predictor[1] = (int16_t)(bytestream_get_byte(&buf) << 8); + predictor[0] = (int16_t)(bytestream_get_byte(&buf) << 8); + } else { + predictor[0] = (int16_t)bytestream_get_le16(&buf); } - SE_16BIT(predictor[0]); - SE_16BIT(predictor[1]); /* decode the samples */ - for (in = 8, out = 0; in < buf_size; in++, out++) { - predictor[channel_number] += s->roq_square_array[buf[in]]; - predictor[channel_number] = av_clip_int16(predictor[channel_number]); - output_samples[out] = predictor[channel_number]; + while (buf < buf_end) { + predictor[ch] += s->roq_square_array[*buf++]; + predictor[ch] = av_clip_int16(predictor[ch]); + *output_samples++ = predictor[ch]; /* toggle channel */ - channel_number ^= s->channels - 1; + ch ^= stereo; } break; case CODEC_ID_INTERPLAY_DPCM: - in = 6; /* skip over the stream mask and stream length */ - predictor[0] = AV_RL16(&buf[in]); - in += 2; - SE_16BIT(predictor[0]) - output_samples[out++] = predictor[0]; - if (s->channels == 2) { - predictor[1] = AV_RL16(&buf[in]); - in += 2; - SE_16BIT(predictor[1]) - output_samples[out++] = predictor[1]; + buf += 6; /* skip over the stream mask and stream length */ + + for (ch = 0; ch < s->channels; ch++) { + predictor[ch] = (int16_t)bytestream_get_le16(&buf); + *output_samples++ = predictor[ch]; } - while (in < buf_size) { - predictor[channel_number] += interplay_delta_table[buf[in++]]; - predictor[channel_number] = av_clip_int16(predictor[channel_number]); - output_samples[out++] = predictor[channel_number]; + ch = 0; + while (buf < buf_end) { + predictor[ch] += interplay_delta_table[*buf++]; + predictor[ch] = av_clip_int16(predictor[ch]); + *output_samples++ = predictor[ch]; /* toggle channel */ - channel_number ^= s->channels - 1; + ch ^= stereo; } - break; case CODEC_ID_XAN_DPCM: - in = 0; - shift[0] = shift[1] = 4; - predictor[0] = AV_RL16(&buf[in]); - in += 2; - SE_16BIT(predictor[0]); - if (s->channels == 2) { - predictor[1] = AV_RL16(&buf[in]); - in += 2; - SE_16BIT(predictor[1]); - } - - while (in < buf_size) { - byte = buf[in++]; - diff = (byte & 0xFC) << 8; - if ((byte & 0x03) == 3) - shift[channel_number]++; + { + int shift[2] = { 4, 4 }; + + for (ch = 0; ch < s->channels; ch++) + predictor[ch] = (int16_t)bytestream_get_le16(&buf); + + ch = 0; + while (buf < buf_end) { + uint8_t n = *buf++; + int16_t diff = (n & 0xFC) << 8; + if ((n & 0x03) == 3) + shift[ch]++; else - shift[channel_number] -= (2 * (byte & 3)); + shift[ch] -= (2 * (n & 3)); /* saturate the shifter to a lower limit of 0 */ - if (shift[channel_number] < 0) - shift[channel_number] = 0; + if (shift[ch] < 0) + shift[ch] = 0; - diff >>= shift[channel_number]; - predictor[channel_number] += diff; + diff >>= shift[ch]; + predictor[ch] += diff; - predictor[channel_number] = av_clip_int16(predictor[channel_number]); - output_samples[out++] = predictor[channel_number]; + predictor[ch] = av_clip_int16(predictor[ch]); + *output_samples++ = predictor[ch]; /* toggle channel */ - channel_number ^= s->channels - 1; + ch ^= stereo; } break; + } case CODEC_ID_SOL_DPCM: - in = 0; if (avctx->codec_tag != 3) { - if(*data_size/4 < buf_size) - return -1; - while (in < buf_size) { - int n1, n2; - n1 = (buf[in] >> 4) & 0xF; - n2 = buf[in++] & 0xF; - s->sample[0] += s->sol_table[n1]; - if (s->sample[0] < 0) s->sample[0] = 0; - if (s->sample[0] > 255) s->sample[0] = 255; - output_samples[out++] = (s->sample[0] - 128) << 8; - s->sample[s->channels - 1] += s->sol_table[n2]; - if (s->sample[s->channels - 1] < 0) s->sample[s->channels - 1] = 0; - if (s->sample[s->channels - 1] > 255) s->sample[s->channels - 1] = 255; - output_samples[out++] = (s->sample[s->channels - 1] - 128) << 8; + uint8_t *output_samples_u8 = data; + while (buf < buf_end) { + uint8_t n = *buf++; + + s->sample[0] += s->sol_table[n >> 4]; + s->sample[0] = av_clip_uint8(s->sample[0]); + *output_samples_u8++ = s->sample[0]; + + s->sample[stereo] += s->sol_table[n & 0x0F]; + s->sample[stereo] = av_clip_uint8(s->sample[stereo]); + *output_samples_u8++ = s->sample[stereo]; } } else { - while (in < buf_size) { - int n; - n = buf[in++]; - if (n & 0x80) s->sample[channel_number] -= s->sol_table[n & 0x7F]; - else s->sample[channel_number] += s->sol_table[n & 0x7F]; - s->sample[channel_number] = av_clip_int16(s->sample[channel_number]); - output_samples[out++] = s->sample[channel_number]; + while (buf < buf_end) { + uint8_t n = *buf++; + if (n & 0x80) s->sample[ch] -= sol_table_16[n & 0x7F]; + else s->sample[ch] += sol_table_16[n & 0x7F]; + s->sample[ch] = av_clip_int16(s->sample[ch]); + *output_samples++ = s->sample[ch]; /* toggle channel */ - channel_number ^= s->channels - 1; + ch ^= stereo; } } break; } - *data_size = out * sizeof(short); + *data_size = out; return buf_size; } @@ -310,6 +326,6 @@ AVCodec ff_ ## name_ ## _decoder = { \ } DPCM_DECODER(CODEC_ID_INTERPLAY_DPCM, interplay_dpcm, "DPCM Interplay"); -DPCM_DECODER(CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ"); -DPCM_DECODER(CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol"); -DPCM_DECODER(CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan"); +DPCM_DECODER(CODEC_ID_ROQ_DPCM, roq_dpcm, "DPCM id RoQ"); +DPCM_DECODER(CODEC_ID_SOL_DPCM, sol_dpcm, "DPCM Sol"); +DPCM_DECODER(CODEC_ID_XAN_DPCM, xan_dpcm, "DPCM Xan"); |