diff options
Diffstat (limited to 'src/libxineadec/adpcm.c')
| -rw-r--r-- | src/libxineadec/adpcm.c | 244 |
1 files changed, 240 insertions, 4 deletions
diff --git a/src/libxineadec/adpcm.c b/src/libxineadec/adpcm.c index 12ac4c6da..87b03b4dd 100644 --- a/src/libxineadec/adpcm.c +++ b/src/libxineadec/adpcm.c @@ -23,8 +23,15 @@ * formats that various entities have created. Details about the data * formats can be found here: * http://www.pcisys.net/~melanson/codecs/ + * CD-ROM/XA ADPCM decoder by Stuart Caie (kyzer@4u.net) + * - based on information in the USENET post by Jac Goudsmit (jac@codim.nl) + * <01bbc34c$dbf64020$f9c8a8c0@cray.codim.nl> + * - tested for correctness using Jon Atkins's CDXA software: + * http://jonatkins.org/cdxa/ + * this is also useful for extracting streams from Playstation discs * - * $Id: adpcm.c,v 1.28 2003/02/14 00:55:52 miguelfreitas Exp $ + * + * $Id: adpcm.c,v 1.29 2003/02/14 04:32:28 tmmm Exp $ */ #include <stdio.h> @@ -83,6 +90,10 @@ static int ea_adpcm_table[] = { 3, 4, 7, 8, 10, 11, 0, -1, -3, -4 }; +static int xa_adpcm_table[] = { + 0, 240, 460, 392, 0, 0, -208, -220 +}; + #define QT_IMA_ADPCM_PREAMBLE_SIZE 2 #define QT_IMA_ADPCM_BLOCK_SIZE 0x22 #define QT_IMA_ADPCM_SAMPLES_PER_BLOCK \ @@ -134,6 +145,12 @@ typedef struct adpcm_decoder_s { unsigned int in_block_size; unsigned int out_block_size; /* size in samples (2 bytes/sample) */ + int xa_mode; /* 1 for mode A, 0 for mode B or mode C */ + int xa_p_l; /* previous sample, left/mono channel */ + int xa_p_r; /* previous sample, right channel */ + int xa_pp_l; /* 2nd-previous sample, left/mono channel */ + int xa_pp_r; /* 2nd-previous sample, right channel */ + } adpcm_decoder_t; /* @@ -1147,6 +1164,202 @@ static void dialogic_ima_decode_block(adpcm_decoder_t *this, buf_element_t *buf) this->size = 0; } + +static void xa_adpcm_decode_block(adpcm_decoder_t *this, buf_element_t *buf) { + int32_t p_l, pp_l, coeff_p_l, coeff_pp_l, range_l; + int32_t p_r, pp_r, coeff_p_r, coeff_pp_r, range_r; + int32_t snd_group, snd_unit, snd_data, samp, i, j; + uint8_t *inp; + + /* restore decoding history */ + p_l = this->xa_p_l; pp_l = this->xa_pp_l; + p_r = this->xa_p_r; pp_r = this->xa_pp_r; + + inp = &this->buf[0]; + j = 0; + + if (this->xa_mode) { + if (this->channels == 2) { + /* mode A (8 bits per sample / 4 sound units) stereo + * - sound units 0,2 are left channel, 1,3 are right channel + * - sound data (8 bits) is shifted left to 16-bit border, then + * shifted right by the range parameter, therefore it's shifted + * (8-range) bits left. + * - two coefficients tables (4 entries each) are merged into one + * - coefficients are multiples of 1/256, so '>> 8' is applied + * after multiplication to get correct answer. + */ + for (snd_group = 0; snd_group < 18; snd_group++, inp += 128) { + for (snd_unit = 0; snd_unit < 4; snd_unit += 2) { + /* get left channel coeffs and range */ + coeff_p_l = xa_adpcm_table[((inp[snd_unit] >> 4) & 0x3)]; + coeff_pp_l = xa_adpcm_table[((inp[snd_unit] >> 4) & 0x3) + 4]; + range_l = 8 - (inp[snd_unit] & 0xF); + + /* get right channel coeffs and range */ + coeff_p_r = xa_adpcm_table[((inp[snd_unit+1] >> 4) & 0x3)]; + coeff_pp_r = xa_adpcm_table[((inp[snd_unit+1] >> 4) & 0x3) + 4]; + range_r = 8 - (inp[snd_unit+1] & 0xF); + + for (snd_data = 0; snd_data < 28; snd_data++) { + /* left channel */ + samp = ((signed char *)inp)[16 + (snd_data << 2) + snd_unit]; + samp <<= range_l; + samp += (coeff_p_l * p_l + coeff_pp_l * pp_l) >> 8; + CLAMP_S16(samp); + pp_l = p_l; + p_l = samp; + this->decode_buffer[j++] = (unsigned short) samp; + + /* right channel */ + samp = ((signed char *)inp)[16 + (snd_data << 2) + snd_unit+1]; + samp <<= range_r; + samp += (coeff_p_r * p_r + coeff_pp_r * pp_r) >> 8; + CLAMP_S16(samp); + pp_r = p_r; + p_r = samp; + this->decode_buffer[j++] = (unsigned short) samp; + } + } + } + } + else { + /* mode A (8 bits per sample / 4 sound units) mono + * - other details as before + */ + for (snd_group = 0; snd_group < 18; snd_group++, inp += 128) { + for (snd_unit = 0; snd_unit < 4; snd_unit++) { + /* get coeffs and range */ + coeff_p_l = xa_adpcm_table[((inp[snd_unit] >> 4) & 0x3)]; + coeff_pp_l = xa_adpcm_table[((inp[snd_unit] >> 4) & 0x3) + 4]; + range_l = 8 - (inp[snd_unit] & 0xF); + + for (snd_data = 0; snd_data < 28; snd_data++) { + samp = ((signed char *)inp)[16 + (snd_data << 2) + snd_unit]; + samp <<= range_l; + samp += (coeff_p_l * p_l + coeff_pp_l * pp_l) >> 8; + CLAMP_S16(samp); + pp_l = p_l; p_l = samp; + this->decode_buffer[j++] = (unsigned short) samp; + } + } + } + } + } + else { + if (this->channels == 2) { + /* mode B/C (4 bits per sample / 8 sound units) stereo + * - sound units 0,2,4,6 are left channel, 1,3,5,7 are right channel + * - sound parameters 0-7 are stored as 16 bytes in the order + * "0123012345674567", so inp[x+4] gives sound parameter x while + * inp[x] doesn't. + * - sound data (4 bits) is shifted left to 16-bit border, then + * shifted right by the range parameter, therefore it's shifted + * (12-range) bits left. + * - other details as before + */ + for (snd_group = 0; snd_group < 18; snd_group++, inp += 128) { + for (snd_unit = 0; snd_unit < 8; snd_unit += 2) { + /* get left channel coeffs and range */ + coeff_p_l = xa_adpcm_table[((inp[snd_unit+4] >> 4) & 0x3)]; + coeff_pp_l = xa_adpcm_table[((inp[snd_unit+4] >> 4) & 0x3) + 4]; + range_l = 12 - (inp[snd_unit+4] & 0xF); + + /* get right channel coeffs and range */ + coeff_p_r = xa_adpcm_table[((inp[snd_unit+5] >> 4) & 0x3)]; + coeff_pp_r = xa_adpcm_table[((inp[snd_unit+5] >> 4) & 0x3) + 4]; + range_r = 12 - (inp[snd_unit+5] & 0xF); + + for (snd_data = 0; snd_data < 28; snd_data++) { + /* left channel */ + samp = (inp[16 + (snd_data << 2) + (snd_unit >> 1)]) & 0xF; + SE_4BIT(samp); + samp <<= range_l; + samp += (coeff_p_l * p_l + coeff_pp_l * pp_l) >> 8; + CLAMP_S16(samp); + pp_l = p_l; + p_l = samp; + this->decode_buffer[j++] = (unsigned short) samp; + + /* right channel */ + samp = (inp[16 + (snd_data << 2) + (snd_unit >> 1)] >> 4) & 0xF; + SE_4BIT(samp); + samp <<= range_r; + samp += (coeff_p_r * p_r + coeff_pp_r * pp_r) >> 8; + CLAMP_S16(samp); + pp_r = p_r; + p_r = samp; + this->decode_buffer[j++] = (unsigned short) samp; + } + } + } + } + else { + /* mode B or C (4 bits per sample / 8 sound units) mono + * - other details as before + */ + for (snd_group = 0; snd_group < 18; snd_group++, inp += 128) { + for (snd_unit = 0; snd_unit < 8; snd_unit++) { + /* get coeffs and range */ + coeff_p_l = xa_adpcm_table[((inp[snd_unit+4] >> 4) & 0x3)]; + coeff_pp_l = xa_adpcm_table[((inp[snd_unit+4] >> 4) & 0x3) + 4]; + range_l = 12 - (inp[snd_unit+4] & 0xF); + + for (snd_data = 0; snd_data < 28; snd_data++) { + samp = inp[16 + (snd_data << 2) + (snd_unit >> 1)]; + if (snd_unit & 1) samp >>= 4; samp &= 0xF; + SE_4BIT(samp); + samp <<= range_l; + samp += (coeff_p_l * p_l + coeff_pp_l * pp_l) >> 8; + CLAMP_S16(samp); + pp_l = p_l; + p_l = samp; + this->decode_buffer[j++] = (unsigned short) samp; + } + } + } + } + } + + /* store decoding history */ + this->xa_p_l = p_l; this->xa_pp_l = pp_l; + this->xa_p_r = p_r; this->xa_pp_r = pp_r; + + /* despatch the decoded audio */ + i = 0; + while (i < j) { + audio_buffer_t *audio_buffer; + int bytes_to_send; + + audio_buffer= this->stream->audio_out->get_buffer(this->stream->audio_out); + if (audio_buffer->mem_size == 0) { + printf ("adpcm: Help! Allocated audio buffer with nothing in it!\n"); + return; + } + + if (((j - i) * 2) > audio_buffer->mem_size) { + bytes_to_send = audio_buffer->mem_size; + } + else { + bytes_to_send = (j - i) * 2; + } + + xine_fast_memcpy(audio_buffer->mem, &this->decode_buffer[i], + bytes_to_send); + + audio_buffer->num_frames = bytes_to_send / (2 * this->channels); + audio_buffer->vpts = buf->pts; + buf->pts = 0; + this->stream->audio_out->put_buffer(this->stream->audio_out, + audio_buffer, this->stream); + + i += bytes_to_send / 2; + } + + /* reset input buffer */ + this->size = 0; +} + static void adpcm_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { adpcm_decoder_t *this = (adpcm_decoder_t *) this_gen; @@ -1163,7 +1376,7 @@ static void adpcm_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { this->size = 0; /* load the stream information */ - switch (buf->type) { + switch (buf->type & 0xFFFF0000) { case BUF_AUDIO_MSADPCM: this->stream->meta_info[XINE_META_INFO_AUDIOCODEC] = @@ -1210,6 +1423,11 @@ static void adpcm_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { strdup("Dialogic IMA ADPCM"); break; + case BUF_AUDIO_XA_ADPCM: + this->stream->meta_info[XINE_META_INFO_AUDIOCODEC] = + strdup("CD-ROM/XA ADPCM"); + break; + } /* if the data was transported in an MS-type file (packet size will be @@ -1274,6 +1492,20 @@ static void adpcm_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { this->decode_buffer = xine_xmalloc(this->out_block_size * 2); } + /* XA blocks are always 2304 bytes of input data. For output, there + * are 18 sound groups. These sound groups have 4 sound units (mode A) + * or 8 sound units (mode B or mode C). The sound units have 28 sound + * data samples. So, either 18*4*28=2016 or 18*8*28=4032 samples per + * sector. 2 bytes per sample means 4032 or 8064 bytes per sector. + */ + if ((buf->type & 0xFFFF0000) == BUF_AUDIO_XA_ADPCM) { + /* initialise decoder state */ + this->xa_mode = buf->decoder_info[2]; + this->xa_p_l = this->xa_pp_l = this->xa_p_r = this->xa_pp_r = 0; + /* allocate 2 bytes per sample */ + this->decode_buffer = xine_xmalloc((this->xa_mode) ? 4032 : 8064); + } + return; } @@ -1302,7 +1534,7 @@ static void adpcm_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { /* time to decode a frame */ if (buf->decoder_flags & BUF_FLAG_FRAME_END) { - switch(buf->type) { + switch(buf->type & 0xFFFF0000) { case BUF_AUDIO_MSADPCM: ms_adpcm_decode_block(this, buf); @@ -1339,6 +1571,10 @@ static void adpcm_decode_data (audio_decoder_t *this_gen, buf_element_t *buf) { case BUF_AUDIO_DIALOGIC_IMA: dialogic_ima_decode_block(this, buf); break; + + case BUF_AUDIO_XA_ADPCM: + xa_adpcm_decode_block(this, buf); + break; } } } @@ -1428,7 +1664,7 @@ static uint32_t audio_types[] = { BUF_AUDIO_QTIMAADPCM, BUF_AUDIO_DK3ADPCM, BUF_AUDIO_DK4ADPCM, BUF_AUDIO_SMJPEG_IMA, BUF_AUDIO_VQA_IMA, BUF_AUDIO_EA_ADPCM, - BUF_AUDIO_DIALOGIC_IMA, + BUF_AUDIO_DIALOGIC_IMA, BUF_AUDIO_XA_ADPCM, 0 }; |